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 362d022..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->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 1ddf775..756e76a 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -513,8 +513,6 @@
                     }
                 }
             } 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 5a7a785..6df2ddd 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.
+
+      "License" shall mean the terms and conditions for use, reproduction,
+      and distribution as defined by Sections 1 through 9 of this document.
+
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+   END OF TERMS AND CONDITIONS
+
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 bd9421c..4416b52 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -19,11 +19,9 @@
     AudioResampler.cpp.arm      \
     AudioPolicyService.cpp      \
     ServiceUtilities.cpp        \
+	AudioResamplerCubic.cpp.arm \
     AudioResamplerSinc.cpp.arm
 
-# uncomment to enable AudioResampler::MED_QUALITY
-# LOCAL_SRC_FILES += AudioResamplerCubic.cpp.arm
-
 LOCAL_SRC_FILES += StateQueue.cpp
 
 # uncomment for debugging timing problems related to StateQueue::push()
@@ -33,7 +31,6 @@
     $(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 \
@@ -41,7 +38,6 @@
     libutils \
     libbinder \
     libmedia \
-    libmedia_native \
     libnbaio \
     libhardware \
     libhardware_legacy \
@@ -74,10 +70,37 @@
 # 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
 
 include $(BUILD_SHARED_LIBRARY)
 
+#
+# build audio resampler test tool
+#
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:=               \
+	test-resample.cpp 			\
+    AudioResampler.cpp.arm      \
+	AudioResamplerCubic.cpp.arm \
+    AudioResamplerSinc.cpp.arm
+
+LOCAL_SHARED_LIBRARIES := \
+	libdl \
+    libcutils \
+    libutils
+
+LOCAL_MODULE:= test-resample
+
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_EXECUTABLE)
+
+
 include $(call all-makefiles-under,$(LOCAL_PATH))
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 1913b6f..0c1ab3c 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -417,7 +417,15 @@
             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 +452,8 @@
         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 +474,14 @@
         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 +872,8 @@
 
 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;
@@ -941,8 +957,8 @@
 
 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 +1044,7 @@
     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 +1128,8 @@
 // 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);
 }
 
@@ -1215,7 +1232,8 @@
 {
     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);
+    ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event,
+            param);
     mWaitWorkCV.signal();
 }
 
@@ -1244,7 +1262,8 @@
                 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",
+                    ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; "
+                          "error %d",
                           prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err);
                 }
             } break;
@@ -1282,7 +1301,7 @@
     result.append(buffer);
     snprintf(buffer, SIZE, "standby: %d\n", mStandby);
     result.append(buffer);
-    snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
+    snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate);
     result.append(buffer);
     snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount);
     result.append(buffer);
@@ -1661,7 +1680,8 @@
 
     snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
     result.append(buffer);
-    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
+    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);
@@ -1712,20 +1732,20 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        int frameCount,
+        size_t frameCount,
         const sp<IMemory>& sharedBuffer,
         int sessionId,
-        IAudioFlinger::track_flags_t flags,
+        IAudioFlinger::track_flags_t *flags,
         pid_t tid,
         status_t *status)
 {
     sp<Track> track;
     status_t lStatus;
 
-    bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0;
+    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 (*flags & IAudioFlinger::TRACK_FAST) {
       if (
             // not timed
             (!isTimed) &&
@@ -1739,7 +1759,7 @@
               (
                 (tid != -1) &&
                 ((frameCount == 0) ||
-                (frameCount >= (int) (mFrameCount * kFastTrackMultiplier)))
+                (frameCount >= (mFrameCount * kFastTrackMultiplier)))
               )
             ) &&
             // PCM data
@@ -1766,12 +1786,12 @@
                 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 "
+                "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;
+        *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.
@@ -1781,7 +1801,7 @@
         if (minBufCount < 2) {
             minBufCount = 2;
         }
-        int minFrameCount = mNormalFrameCount * minBufCount;
+        size_t minFrameCount = mNormalFrameCount * minBufCount;
         if (frameCount < minFrameCount) {
             frameCount = minFrameCount;
         }
@@ -1791,7 +1811,7 @@
     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 \""
+                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;
@@ -1801,7 +1821,7 @@
     } 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);
+            ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate);
             lStatus = BAD_VALUE;
             goto Exit;
         }
@@ -1835,7 +1855,7 @@
 
         if (!isTimed) {
             track = new Track(this, client, streamType, sampleRate, format,
-                    channelMask, frameCount, sharedBuffer, sessionId, flags);
+                    channelMask, frameCount, sharedBuffer, sessionId, *flags);
         } else {
             track = TimedTrack::create(this, client, streamType, sampleRate, format,
                     channelMask, frameCount, sharedBuffer, sessionId);
@@ -1854,7 +1874,7 @@
             chain->incTrackCnt();
         }
 
-        if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) {
+        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
@@ -1871,7 +1891,7 @@
     return track;
 }
 
-uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const
+uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const
 {
     if (mFastMixer != NULL) {
         MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
@@ -1880,7 +1900,7 @@
     return latency;
 }
 
-uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const
+uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const
 {
     return latency;
 }
@@ -1893,7 +1913,7 @@
 uint32_t AudioFlinger::PlaybackThread::latency_l() const
 {
     if (initCheck() == NO_ERROR) {
-        return correctLatency(mOutput->stream->get_latency(mOutput->stream));
+        return correctLatency_l(mOutput->stream->get_latency(mOutput->stream));
     } else {
         return 0;
     }
@@ -1959,7 +1979,8 @@
         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());
+                ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(),
+                        track->sessionId());
                 chain->incActiveTrackCnt();
             }
         }
@@ -2025,7 +2046,8 @@
     AudioSystem::OutputDescriptor desc;
     void *param2 = NULL;
 
-    ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
+    ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event,
+            param);
 
     switch (event) {
     case AudioSystem::OUTPUT_OPENED:
@@ -2033,7 +2055,8 @@
         desc.channels = mChannelMask;
         desc.samplingRate = mSampleRate;
         desc.format = mFormat;
-        desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t)
+        desc.frameCount = mNormalFrameCount; // FIXME see
+                                             // AudioFlinger::frameCount(audio_io_handle_t)
         desc.latency = latency();
         param2 = &desc;
         break;
@@ -2062,7 +2085,8 @@
 
     // 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)) {
+    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
@@ -2081,9 +2105,10 @@
                 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)
+            // 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)) {
@@ -2097,7 +2122,8 @@
     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);
+    ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount,
+            mNormalFrameCount);
 
     delete[] mMixBuffer;
     mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount];
@@ -2116,7 +2142,7 @@
 }
 
 
-status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
+status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames)
 {
     if (halFrames == NULL || dspFrames == NULL) {
         return BAD_VALUE;
@@ -2125,15 +2151,13 @@
     if (initCheck() != NO_ERROR) {
         return INVALID_OPERATION;
     }
-    *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
+    size_t framesWritten = mBytesWritten / mFrameSize;
+    *halFrames = framesWritten;
 
     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;
+        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);
@@ -2151,7 +2175,7 @@
     for (size_t i = 0; i < mTracks.size(); ++i) {
         sp<Track> track = mTracks[i];
         if (sessionId == track->sessionId() &&
-                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
+                !(track->mCblk->flags & CBLK_INVALID)) {
             result |= TRACK_SESSION;
             break;
         }
@@ -2170,7 +2194,7 @@
     for (size_t i = 0; i < mTracks.size(); i++) {
         sp<Track> track = mTracks[i];
         if (sessionId == track->sessionId() &&
-                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
+                !(track->mCblk->flags & CBLK_INVALID)) {
             return AudioSystem::getStrategyForStream(track->streamType());
         }
     }
@@ -2235,7 +2259,8 @@
     return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE;
 }
 
-void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove)
+void AudioFlinger::PlaybackThread::threadLoop_removeTracks(
+        const Vector< sp<Track> >& tracksToRemove)
 {
     size_t count = tracksToRemove.size();
     if (CC_UNLIKELY(count)) {
@@ -2263,7 +2288,7 @@
         // mNormalSink below
 {
     ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
-    ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, "
+    ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, "
             "mFrameCount=%d, mNormalFrameCount=%d",
             mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
             mNormalFrameCount);
@@ -2604,7 +2629,9 @@
 
                     clearOutputTracks();
 
-                    if (exitPending()) break;
+                    if (exitPending()) {
+                        break;
+                    }
 
                     releaseWakeLock_l();
                     // wait until we have something to do...
@@ -2797,7 +2824,9 @@
         bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
     }
 
-    if (bytesWritten > 0) mBytesWritten += mixBufferSize;
+    if (bytesWritten > 0) {
+        mBytesWritten += mixBufferSize;
+    }
     mNumWrites++;
     mInWrite = false;
 }
@@ -2891,7 +2920,8 @@
     } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) {
         memset (mMixBuffer, 0, mixBufferSize);
         sleepTime = 0;
-        ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start");
+        ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED),
+                "anticipated start");
     }
     // TODO add standby time extension fct of effect tail
 }
@@ -2937,7 +2967,9 @@
 
     for (size_t i=0 ; i<count ; i++) {
         sp<Track> t = mActiveTracks[i].promote();
-        if (t == 0) continue;
+        if (t == 0) {
+            continue;
+        }
 
         // this const just means the local variable doesn't change
         Track* const track = t.get();
@@ -3014,7 +3046,7 @@
                     }
                     // 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);
+                    android_atomic_or(CBLK_DISABLED, &track->mCblk->flags);
                     // remove from active list, but state remains ACTIVE [confusing but true]
                     isActive = false;
                     break;
@@ -3031,8 +3063,7 @@
                 {
                     size_t audioHALFrames =
                             (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
-                    size_t framesWritten =
-                            mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
+                    size_t framesWritten = mBytesWritten / mFrameSize;
                     if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) {
                         // track stays in active list until presentation is complete
                         break;
@@ -3108,7 +3139,7 @@
         uint32_t minFrames = 1;
         if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
                 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
-            if (t->sampleRate() == (int)mSampleRate) {
+            if (t->sampleRate() == mSampleRate) {
                 minFrames = mNormalFrameCount;
             } else {
                 // +1 for rounding and +1 for additional sample needed for interpolation
@@ -3125,7 +3156,8 @@
         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);
+            ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server,
+                    this);
 
             mixedTracks++;
 
@@ -3138,7 +3170,8 @@
                 if (chain != 0) {
                     tracksWithEffect++;
                 } else {
-                    ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d",
+                    ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on "
+                            "session %d",
                             name, track->sessionId());
                 }
             }
@@ -3215,11 +3248,17 @@
             // 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;
+            if (vl > MAX_GAIN_INT) {
+                vl = MAX_GAIN_INT;
+            }
             vr = (vr + (1 << 11)) >> 12;
-            if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT;
+            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 -
+            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);
@@ -3268,7 +3307,8 @@
                 chain->clearInputBuffer();
             }
 
-            //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this);
+            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.
@@ -3276,8 +3316,7 @@
                 // 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);
+                size_t framesWritten = mBytesWritten / mFrameSize;
                 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
                     if (track->isStopped()) {
                         track->reset();
@@ -3293,7 +3332,7 @@
                     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);
+                    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
@@ -3347,7 +3386,9 @@
         ALOG_ASSERT(i < count);
         resetMask &= ~(1 << i);
         sp<Track> t = mActiveTracks[i].promote();
-        if (t == 0) continue;
+        if (t == 0) {
+            continue;
+        }
         Track* track = t.get();
         ALOG_ASSERT(track->isFastTrack() && track->isStopped());
         track->reset();
@@ -3362,7 +3403,8 @@
             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());
+                    ALOGV("stopping track on chain %p for session Id: %d", chain.get(),
+                            track->sessionId());
                     chain->decActiveTrackCnt();
                 }
             }
@@ -3375,7 +3417,8 @@
     // 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)) {
+    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));
     }
@@ -3424,7 +3467,7 @@
     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);
+            android_atomic_or(CBLK_INVALID, &t->mCblk->flags);
             t->mCblk->cv.signal();
         }
     }
@@ -3547,7 +3590,9 @@
                 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;
+                    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()) {
@@ -3580,39 +3625,18 @@
     return reconfig;
 }
 
-void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
+void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
 {
-    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();
+    NBAIO_Source *teeSource = source.get();
     if (teeSource != NULL) {
-        char teePath[64];
+        char teeTime[16];
         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);
+        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];
@@ -3622,7 +3646,7 @@
             NBAIO_Format format = teeSource->format();
             unsigned channelCount = Format_channelCount(format);
             ALOG_ASSERT(channelCount <= FCC_2);
-            unsigned sampleRate = Format_sampleRate(format);
+            uint32_t sampleRate = Format_sampleRate(format);
             wavHeader[22] = channelCount;       // number of channels
             wavHeader[24] = sampleRate;         // sample rate
             wavHeader[25] = sampleRate >> 8;
@@ -3660,6 +3684,34 @@
             fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno));
         }
     }
+}
+
+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) {
@@ -3715,7 +3767,9 @@
     if (mActiveTracks.size() != 0) {
         sp<Track> t = mActiveTracks[0].promote();
         // The track died recently
-        if (t == 0) return MIXER_IDLE;
+        if (t == 0) {
+            return MIXER_IDLE;
+        }
 
         Track* const track = t.get();
         audio_track_cblk_t* cblk = track->cblk();
@@ -3731,7 +3785,7 @@
         if ((track->framesReady() >= minFrames) && track->isReady() &&
                 !track->isPaused() && !track->isTerminated())
         {
-            //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
+            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;
@@ -3754,10 +3808,14 @@
                 float v = mMasterVolume * typeVolume;
                 uint32_t vlr = cblk->getVolumeLR();
                 float v_clamped = v * (vlr & 0xFFFF);
-                if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+                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;
+                if (v_clamped > MAX_GAIN) {
+                    v_clamped = MAX_GAIN;
+                }
                 right = v_clamped/MAX_GAIN;
             }
 
@@ -3792,15 +3850,14 @@
                 mEffectChains[0]->clearInputBuffer();
             }
 
-            //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
+            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 / audio_stream_frame_size(&mOutput->stream->common);
+                size_t framesWritten = mBytesWritten / mFrameSize;
                 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
                     if (track->isStopped()) {
                         track->reset();
@@ -3982,7 +4039,8 @@
 
 AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
         AudioFlinger::MixerThread* mainThread, audio_io_handle_t id)
-    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), DUPLICATING),
+    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(),
+                DUPLICATING),
         mWaitTimeMs(UINT_MAX)
 {
     addOutputTrack(mainThread);
@@ -4058,7 +4116,7 @@
 {
     Mutex::Autolock _l(mLock);
     // FIXME explain this formula
-    int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
+    size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
     OutputTrack *outputTrack = new OutputTrack(thread,
                                             this,
                                             mSampleRate,
@@ -4103,18 +4161,21 @@
 }
 
 
-bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks)
+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());
+            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());
+            ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(),
+                    thread.get());
             return false;
         }
     }
@@ -4143,7 +4204,7 @@
             uint32_t sampleRate,
             audio_format_t format,
             audio_channel_mask_t channelMask,
-            int frameCount,
+            size_t frameCount,
             const sp<IMemory>& sharedBuffer,
             int sessionId)
     :   RefBase(),
@@ -4152,52 +4213,36 @@
         mCblk(NULL),
         // mBuffer
         // mBufferEnd
-        mFrameCount(0),
+        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)
-        // mChannelCount
-        // mChannelMask
 {
-    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
+    // 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);
-    uint8_t channelCount = popcount(channelMask);
-    size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
+    size_t bufferSize = frameCount * mFrameSize;
     if (sharedBuffer == 0) {
         size += bufferSize;
     }
 
-    if (client != NULL) {
+    if (client != 0) {
         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;
-            }
+            // can't assume mCblk != NULL
         } else {
             ALOGE("not enough memory for AudioTrack size=%u", size);
             client->heap()->dump("AudioTrack");
@@ -4205,23 +4250,29 @@
         }
     } else {
         mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
-        // construct the shared structure in-place.
+        // 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->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;
+//      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;
     }
 }
@@ -4252,7 +4303,7 @@
 void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
 {
     buffer->raw = NULL;
-    mFrameCount = buffer->frameCount;
+    mStepCount = buffer->frameCount;
     // FIXME See note at getNextBuffer()
     (void) step();      // ignore return value of step()
     buffer->frameCount = 0;
@@ -4262,7 +4313,7 @@
     bool result;
     audio_track_cblk_t* cblk = this->cblk();
 
-    result = cblk->stepServer(mFrameCount);
+    result = cblk->stepServer(mStepCount, mFrameCount, isOut());
     if (!result) {
         ALOGV("stepServer failed acquiring cblk mutex");
         mStepServerFailed = true;
@@ -4281,23 +4332,22 @@
     ALOGV("TrackBase::reset");
 }
 
-int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
-    return (int)mCblk->sampleRate;
+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();
-    size_t frameSize = cblk->frameSize;
-    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize;
-    int8_t *bufferEnd = bufferStart + frames * frameSize;
+    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 %d",
+                "    server %u, serverBase %u, user %u, userBase %u, frameSize %u",
                 bufferStart, bufferEnd, mBuffer, mBufferEnd,
-                cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize);
+                cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize);
 
     return bufferStart;
 }
@@ -4318,11 +4368,12 @@
             uint32_t sampleRate,
             audio_format_t format,
             audio_channel_mask_t channelMask,
-            int frameCount,
+            size_t frameCount,
             const sp<IMemory>& sharedBuffer,
             int sessionId,
             IAudioFlinger::track_flags_t flags)
-    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId),
+    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer,
+            sessionId),
     mMute(false),
     mFillingUpStatus(FS_INVALID),
     // mRetryCount initialized later when needed
@@ -4339,9 +4390,6 @@
     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;
@@ -4351,7 +4399,6 @@
         }
         // 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);
@@ -4366,7 +4413,8 @@
             thread->mFastTrackAvailMask &= ~(1 << i);
         }
     }
-    ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid());
+    ALOGV("Track constructor name %d, calling pid %d", mName,
+            IPCThreadState::self()->getCallingPid());
 }
 
 AudioFlinger::PlaybackThread::Track::~Track()
@@ -4408,8 +4456,8 @@
 
 /*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");
+    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)
@@ -4479,8 +4527,8 @@
             mFormat,
             mChannelMask,
             mSessionId,
+            mStepCount,
             mFrameCount,
-            mCblk->frameCount,
             stateChar,
             mMute,
             mFillingUpStatus,
@@ -4519,11 +4567,11 @@
     }
 
     // FIXME Same as above
-    framesReady = cblk->framesReady();
+    framesReady = cblk->framesReadyOut();
 
     if (CC_LIKELY(framesReady)) {
         uint32_t s = cblk->server;
-        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
+        uint32_t bufferEnd = cblk->serverBase + mFrameCount;
 
         bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
         if (framesReq > framesReady) {
@@ -4554,17 +4602,19 @@
 // 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();
+    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 (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
+        return true;
+    }
 
-    if (framesReady() >= mCblk->frameCount ||
-            (mCblk->flags & CBLK_FORCEREADY_MSK)) {
+    if (framesReady() >= mFrameCount ||
+            (mCblk->flags & CBLK_FORCEREADY)) {
         mFillingUpStatus = FS_FILLED;
-        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
+        android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags);
         return true;
     }
     return false;
@@ -4636,7 +4686,8 @@
                 // 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);
+            ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName,
+                    playbackThread);
         }
         if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
             thread->mLock.unlock();
@@ -4706,8 +4757,8 @@
         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);
+        android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags);
+        android_atomic_or(CBLK_UNDERRUN, &mCblk->flags);
         mFillingUpStatus = FS_FILLING;
         mResetDone = true;
         if (mState == FLUSHED) {
@@ -4848,6 +4899,11 @@
     return NO_ERROR;
 }
 
+bool AudioFlinger::PlaybackThread::Track::isOut() const
+{
+    return true;
+}
+
 // timed audio tracks
 
 sp<AudioFlinger::PlaybackThread::TimedTrack>
@@ -4858,7 +4914,7 @@
             uint32_t sampleRate,
             audio_format_t format,
             audio_channel_mask_t channelMask,
-            int frameCount,
+            size_t frameCount,
             const sp<IMemory>& sharedBuffer,
             int sessionId) {
     if (!client->reserveTimedTrack())
@@ -4876,7 +4932,7 @@
             uint32_t sampleRate,
             audio_format_t format,
             audio_channel_mask_t channelMask,
-            int frameCount,
+            size_t frameCount,
             const sp<IMemory>& sharedBuffer,
             int sessionId)
     : Track(thread, client, streamType, sampleRate, format, channelMask,
@@ -4982,7 +5038,7 @@
             // this frame in media time units and adding it to the PTS of the
             // buffer.
             int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
-                               / mCblk->frameSize;
+                               / mFrameSize;
 
             if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
                                                                 &bufEnd)) {
@@ -5042,7 +5098,7 @@
                 " bytes.  (update reason: \"%s\")",
                 bufBytes, consumedAlready, logTag);
 
-    uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize;
+    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"
@@ -5063,7 +5119,7 @@
 
     Mutex::Autolock _l(mTimedBufferQueueLock);
 
-    uint32_t bufFrames = buffer->size() / mCblk->frameSize;
+    uint32_t bufFrames = buffer->size() / mFrameSize;
     mFramesPendingInQueue += bufFrames;
     mTimedBufferQueue.add(TimedBuffer(buffer, pts));
 
@@ -5160,7 +5216,7 @@
         // 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());
+                ((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.
@@ -5225,7 +5281,7 @@
             // the next input sample is late
             uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
             size_t onTimeSamplePosition =
-                    head.position() + lateFrames * mCblk->frameSize;
+                    head.position() + lateFrames * mFrameSize;
 
             if (onTimeSamplePosition > head.buffer()->size()) {
                 // all the remaining samples in the head are too late, so
@@ -5260,7 +5316,7 @@
                    head.position());
 
     uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
-                                 mCblk->frameSize);
+                                 mFrameSize);
     size_t framesRequested = buffer->frameCount;
     buffer->frameCount = min(framesLeftInHead, framesRequested);
 
@@ -5275,9 +5331,9 @@
     uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
 
     // lazily allocate a buffer filled with silence
-    if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) {
+    if (mTimedSilenceBufferSize < numFrames * mFrameSize) {
         delete [] mTimedSilenceBuffer;
-        mTimedSilenceBufferSize = numFrames * mCblk->frameSize;
+        mTimedSilenceBufferSize = numFrames * mFrameSize;
         mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
         memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
     }
@@ -5325,7 +5381,7 @@
                     start, end, buffer->raw);
 
         head.setPosition(head.position() +
-                (buffer->frameCount * mCblk->frameSize));
+                (buffer->frameCount * mFrameSize));
         mQueueHeadInFlight = false;
 
         ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
@@ -5371,22 +5427,13 @@
             uint32_t sampleRate,
             audio_format_t format,
             audio_channel_mask_t channelMask,
-            int frameCount,
+            size_t 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);
-        }
-    }
+    ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
 }
 
 AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
@@ -5395,7 +5442,8 @@
 }
 
 // AudioBufferProvider interface
-status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
+status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
+        int64_t pts)
 {
     audio_track_cblk_t* cblk = this->cblk();
     uint32_t framesAvail;
@@ -5403,16 +5451,19 @@
 
     // Check if last stepServer failed, try to step now
     if (mStepServerFailed) {
-        if (!step()) goto getNextBuffer_exit;
+        if (!step()) {
+            goto getNextBuffer_exit;
+        }
         ALOGV("stepServer recovered");
         mStepServerFailed = false;
     }
 
-    framesAvail = cblk->framesAvailable_l();
+    // 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 + cblk->frameCount;
+        uint32_t bufferEnd = cblk->serverBase + mFrameCount;
 
         if (framesReq > framesAvail) {
             framesReq = framesAvail;
@@ -5455,7 +5506,7 @@
             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);
+            android_atomic_or(CBLK_UNDERRUN, &mCblk->flags);
         }
         recordThread->mLock.unlock();
         if (doStop) {
@@ -5466,7 +5517,7 @@
 
 /*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
 {
-    result.append("   Clien Fmt Chn mask   Session Buf  S SRate  Serv     User   FrameCount\n");
+    result.append("   Clien Fmt Chn mask   Session Step S SRate  Serv     User   FrameCount\n");
 }
 
 void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
@@ -5476,14 +5527,18 @@
             mFormat,
             mChannelMask,
             mSessionId,
-            mFrameCount,
+            mStepCount,
             mState,
             mCblk->sampleRate,
             mCblk->server,
             mCblk->user,
-            mCblk->frameCount);
+            mFrameCount);
 }
 
+bool AudioFlinger::RecordThread::RecordTrack::isOut() const
+{
+    return false;
+}
 
 // ----------------------------------------------------------------------------
 
@@ -5493,20 +5548,19 @@
             uint32_t sampleRate,
             audio_format_t format,
             audio_channel_mask_t channelMask,
-            int frameCount)
+            size_t frameCount)
     :   Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
                 NULL, 0, IAudioFlinger::TRACK_DEFAULT),
-    mActive(false), mSourceThread(sourceThread)
+    mActive(false), mSourceThread(sourceThread), mBuffers(NULL)
 {
 
     if (mCblk != NULL) {
-        mCblk->flags |= CBLK_DIRECTION_OUT;
-        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
+        mBuffers = (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,
+        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);
@@ -5555,9 +5609,9 @@
         sp<ThreadBase> thread = mThread.promote();
         if (thread != 0) {
             MixerThread *mixerThread = (MixerThread *)thread.get();
-            if (mCblk->frameCount > frames){
+            if (mFrameCount > frames){
                 if (mBufferQueue.size() < kMaxOverFlowBuffers) {
-                    uint32_t startFrames = (mCblk->frameCount - frames);
+                    uint32_t startFrames = (mFrameCount - frames);
                     pInBuffer = new Buffer;
                     pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
                     pInBuffer->frameCount = startFrames;
@@ -5587,7 +5641,8 @@
             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());
+                ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this,
+                        mThread.unsafe_get());
                 outputBufferFull = true;
                 break;
             }
@@ -5599,9 +5654,10 @@
             }
         }
 
-        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
+        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
+                pInBuffer->frameCount;
         memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
-        mCblk->stepUser(outFrames);
+        mCblk->stepUserOut(outFrames, mFrameCount);
         pInBuffer->frameCount -= outFrames;
         pInBuffer->i16 += outFrames * channelCount;
         mOutBuffer.frameCount -= outFrames;
@@ -5612,7 +5668,8 @@
                 mBufferQueue.removeAt(0);
                 delete [] pInBuffer->mBuffer;
                 delete pInBuffer;
-                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
+                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
+                        mThread.unsafe_get(), mBufferQueue.size());
             } else {
                 break;
             }
@@ -5628,11 +5685,14 @@
                 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));
+                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());
+                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);
+                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
+                        mThread.unsafe_get(), this);
             }
         }
     }
@@ -5641,8 +5701,8 @@
     // 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;
+        if (mCblk->user < mFrameCount) {
+            frames = mFrameCount - mCblk->user;
             pInBuffer = new Buffer;
             pInBuffer->mBuffer = new int16_t[frames * channelCount];
             pInBuffer->frameCount = frames;
@@ -5657,17 +5717,18 @@
     return outputBufferFull;
 }
 
-status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
+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);
+    ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
     buffer->frameCount  = 0;
 
-    uint32_t framesAvail = cblk->framesAvailable();
+    uint32_t framesAvail = cblk->framesAvailableOut(mFrameCount);
 
 
     if (framesAvail == 0) {
@@ -5685,7 +5746,7 @@
             }
             // read the server count again
         start_loop_here:
-            framesAvail = cblk->framesAvailable_l();
+            framesAvail = cblk->framesAvailableOut_l(mFrameCount);
         }
     }
 
@@ -5698,14 +5759,14 @@
     }
 
     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;
     }
 
     buffer->frameCount  = framesReq;
-    buffer->raw         = (void *)cblk->buffer(u);
+    buffer->raw         = cblk->buffer(mBuffers, mFrameSize, u);
     return NO_ERROR;
 }
 
@@ -5881,7 +5942,7 @@
         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 +5982,14 @@
                 *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;
@@ -5946,7 +6008,8 @@
 
 // ----------------------------------------------------------------------------
 
-AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
+AudioFlinger::RecordHandle::RecordHandle(
+        const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
     : BnAudioRecord(),
     mRecordTrack(recordTrack)
 {
@@ -5961,7 +6024,8 @@
     return mRecordTrack->getCblk();
 }
 
-status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, int triggerSession) {
+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);
 }
@@ -5988,18 +6052,21 @@
                                          uint32_t sampleRate,
                                          audio_channel_mask_t channelMask,
                                          audio_io_handle_t id,
-                                         audio_devices_t device) :
+                                         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)
+    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();
+
 }
 
 
@@ -6047,7 +6114,9 @@
             if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
                 standby();
 
-                if (exitPending()) break;
+                if (exitPending()) {
+                    break;
+                }
 
                 releaseWakeLock_l();
                 ALOGV("RecordThread: loop stopping");
@@ -6106,12 +6175,13 @@
                         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;
+                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) *
+                                    mActiveTrack->mFrameSize;
                             if (framesIn > framesOut)
                                 framesIn = framesOut;
                             mRsmpInIndex += framesIn;
                             framesOut -= framesIn;
-                            if ((int)mChannelCount == mReqChannelCount ||
+                            if (mChannelCount == mReqChannelCount ||
                                 mFormat != AUDIO_FORMAT_PCM_16_BIT) {
                                 memcpy(dst, src, framesIn * mFrameSize);
                             } else {
@@ -6125,14 +6195,17 @@
                             }
                         }
                         if (framesOut && mFrameCount == mRsmpInIndex) {
+                            void *readInto;
                             if (framesOut == mFrameCount &&
-                                ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
-                                mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes);
+                                (mChannelCount == mReqChannelCount ||
+                                        mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
+                                readInto = buffer.raw;
                                 framesOut = 0;
                             } else {
-                                mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
+                                readInto = mRsmpInBuffer;
                                 mRsmpInIndex = 0;
                             }
+                            mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes);
                             if (mBytesRead <= 0) {
                                 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE))
                                 {
@@ -6145,6 +6218,9 @@
                                 mRsmpInIndex = mFrameCount;
                                 framesOut = 0;
                                 buffer.frameCount = 0;
+                            } else if (mTeeSink != 0) {
+                                (void) mTeeSink->write(readInto,
+                                        mBytesRead >> Format_frameBitShift(mTeeSink->format()));
                             }
                         }
                     }
@@ -6156,12 +6232,14 @@
                     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.
+                    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
+                        // 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 {
@@ -6243,7 +6321,7 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        int frameCount,
+        size_t frameCount,
         int sessionId,
         IAudioFlinger::track_flags_t flags,
         pid_t tid,
@@ -6498,9 +6576,9 @@
         result.append(buffer);
         snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
         result.append(buffer);
-        snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
+        snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount);
         result.append(buffer);
-        snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
+        snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate);
         result.append(buffer);
     } else {
         result.append("No active record client\n");
@@ -6595,8 +6673,8 @@
         AudioParameter param = AudioParameter(keyValuePair);
         int value;
         audio_format_t reqFormat = mFormat;
-        int reqSamplingRate = mReqSampleRate;
-        int reqChannelCount = mReqChannelCount;
+        uint32_t reqSamplingRate = mReqSampleRate;
+        uint32_t reqChannelCount = mReqChannelCount;
 
         if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
             reqSamplingRate = value;
@@ -6635,7 +6713,8 @@
                 status = BAD_VALUE;
             } else {
                 mInDevice = value;
-                // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
+                // 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();
@@ -6657,7 +6736,8 @@
             mAudioSource = (audio_source_t)value;
         }
         if (status == NO_ERROR) {
-            status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
+            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,
@@ -6667,8 +6747,10 @@
                 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 &&
+                    ((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;
                 }
@@ -6762,7 +6844,8 @@
         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
+        // optmization: if mono to mono, alter input frame count as if we were inputing
+        // stereo samples
         if (mChannelCount == 1 && mReqChannelCount == 1) {
             mFrameCount >>= 1;
         }
@@ -6924,14 +7007,14 @@
 
 // ----------------------------------------------------------------------------
 
-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 +7072,8 @@
                                           &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 +7126,8 @@
     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 +7162,8 @@
         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 +7250,17 @@
 
     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 +7271,66 @@
     }
 
     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;
@@ -8003,7 +8138,8 @@
         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);
+                ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(),
+                        buffer);
                 track->setMainBuffer(buffer);
                 chain->incTrackCnt();
             }
@@ -8012,7 +8148,9 @@
         // 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 (track == 0) {
+                continue;
+            }
             if (session == track->sessionId()) {
                 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
                 chain->incActiveTrackCnt();
@@ -8035,7 +8173,9 @@
     size_t size = mEffectChains.size();
     size_t i = 0;
     for (i = 0; i < size; i++) {
-        if (mEffectChains[i]->sessionId() < session) break;
+        if (mEffectChains[i]->sessionId() < session) {
+            break;
+        }
     }
     mEffectChains.insertAt(chain, i);
     checkSuspendOnAddEffectChain_l(chain);
@@ -8055,7 +8195,9 @@
             // 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 (track == 0) {
+                    continue;
+                }
                 if (session == track->sessionId()) {
                     ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
                             chain.get(), session);
@@ -8222,11 +8364,15 @@
     size_t i;
     for (i = 0; i < size; i++) {
         EffectHandle *h = mHandles[i];
-        if (h == NULL || h->destroyed_l()) continue;
+        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 (h->priority() <= priority) {
+            break;
+        }
     }
     // if inserted in first place, move effect control from previous owner to this handle
     if (i == 0) {
@@ -8251,7 +8397,9 @@
     size_t size = mHandles.size();
     size_t i;
     for (i = 0; i < size; i++) {
-        if (mHandles[i] == handle) break;
+        if (mHandles[i] == handle) {
+            break;
+        }
     }
     if (i == size) {
         return size;
@@ -8594,7 +8742,7 @@
                                              void *pReplyData)
 {
     Mutex::Autolock _l(mLock);
-//    ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
+    ALOGVV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
 
     if (mState == DESTROYED || mEffectInterface == NULL) {
         return NO_INIT;
@@ -8845,12 +8993,15 @@
     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.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.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",
@@ -8934,7 +9085,8 @@
             mBuffer = (uint8_t *)mCblk + bufOffset;
         }
     } else {
-        ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
+        ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE +
+                sizeof(effect_param_cblk_t));
         return;
     }
 }
@@ -8956,8 +9108,12 @@
 status_t AudioFlinger::EffectHandle::enable()
 {
     ALOGV("enable %p", this);
-    if (!mHasControl) return INVALID_OPERATION;
-    if (mEffect == 0) return DEAD_OBJECT;
+    if (!mHasControl) {
+        return INVALID_OPERATION;
+    }
+    if (mEffect == 0) {
+        return DEAD_OBJECT;
+    }
 
     if (mEnabled) {
         return NO_ERROR;
@@ -8988,8 +9144,12 @@
 status_t AudioFlinger::EffectHandle::disable()
 {
     ALOGV("disable %p", this);
-    if (!mHasControl) return INVALID_OPERATION;
-    if (mEffect == 0) return DEAD_OBJECT;
+    if (!mHasControl) {
+        return INVALID_OPERATION;
+    }
+    if (mEffect == 0) {
+        return DEAD_OBJECT;
+    }
 
     if (!mEnabled) {
         return NO_ERROR;
@@ -9049,20 +9209,25 @@
                                              uint32_t *replySize,
                                              void *pReplyData)
 {
-//    ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
-//              cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
+    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;
+    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
+        // 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) {
@@ -9202,7 +9367,8 @@
 }
 
 // getEffectFromDesc_l() must be called with ThreadBase::mLock held
-sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
+sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(
+        effect_descriptor_t *descriptor)
 {
     size_t size = mEffects.size();
 
@@ -9361,7 +9527,8 @@
                 // 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);
+                    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
@@ -9412,7 +9579,8 @@
         }
         mEffects.insertAt(effect, idx_insert);
 
-        ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
+        ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this,
+                idx_insert);
     }
     effect->configure();
     return NO_ERROR;
@@ -9443,7 +9611,8 @@
                 }
             }
             mEffects.removeAt(i);
-            ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
+            ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(),
+                    this, i);
             break;
         }
     }
@@ -9522,7 +9691,9 @@
     uint32_t rVol = newRight;
 
     for (size_t i = 0; i < size; i++) {
-        if ((int)i == ctrlIdx) continue;
+        if ((int)i == ctrlIdx) {
+            continue;
+        }
         // this also works for ctrlIdx == -1 when there is no volume controller
         if ((int)i > ctrlIdx) {
             lVol = *left;
@@ -9667,7 +9838,8 @@
             for (size_t i = 0; i < types.size(); i++) {
                 setEffectSuspended_l(types[i], false);
             }
-            ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
+            ALOGV("setEffectSuspendedAll_l() remove entry for %08x",
+                    mSuspendedEffects.keyAt(index));
             mSuspendedEffects.removeItem((int)kKeyForSuspendAll);
         }
     }
@@ -9693,7 +9865,8 @@
     return true;
 }
 
-void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects)
+void AudioFlinger::EffectChain::getSuspendEligibleEffects(
+        Vector< sp<AudioFlinger::EffectModule> > &effects)
 {
     effects.clear();
     for (size_t i = 0; i < mEffects.size(); i++) {
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 49e2b2c..2541b15 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -92,8 +92,8 @@
                                 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 +106,7 @@
                                 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 +174,7 @@
 
     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 +207,8 @@
 
     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,
@@ -269,12 +269,14 @@
     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
@@ -356,7 +358,7 @@
             RECORD              // Thread class is RecordThread
         };
 
-        ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+        ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
                     audio_devices_t outDevice, audio_devices_t inDevice, type_t type);
         virtual             ~ThreadBase();
 
@@ -388,7 +390,7 @@
                                         uint32_t sampleRate,
                                         audio_format_t format,
                                         audio_channel_mask_t channelMask,
-                                        int frameCount,
+                                        size_t frameCount,
                                         const sp<IMemory>& sharedBuffer,
                                         int sessionId);
             virtual             ~TrackBase();
@@ -417,11 +419,11 @@
                 return mFormat;
             }
 
-            int channelCount() const { return mChannelCount; }
+            uint32_t channelCount() const { return mChannelCount; }
 
             audio_channel_mask_t channelMask() const { return mChannelMask; }
 
-            int sampleRate() const; // FIXME inline after cblk sr moved
+            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
@@ -451,9 +453,12 @@
                 return mState == TERMINATED;
             }
 
-            bool step();
+            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;
@@ -461,16 +466,22 @@
             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;
+            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;
-            uint8_t             mChannelCount;
-            audio_channel_mask_t mChannelMask;
             Vector < sp<SyncEvent> >mSyncEvents;
         };
 
@@ -554,7 +565,7 @@
 
                     // dynamic externally-visible
                     uint32_t    sampleRate() const { return mSampleRate; }
-                    int         channelCount() const { return mChannelCount; }
+                    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,
@@ -746,7 +757,8 @@
                     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;
+                    KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > >
+                                            mSuspendedSessions;
     };
 
     struct  stream_type_t {
@@ -780,7 +792,7 @@
                                         uint32_t sampleRate,
                                         audio_format_t format,
                                         audio_channel_mask_t channelMask,
-                                        int frameCount,
+                                        size_t frameCount,
                                         const sp<IMemory>& sharedBuffer,
                                         int sessionId,
                                         IAudioFlinger::track_flags_t flags);
@@ -788,7 +800,8 @@
 
             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,
+            virtual status_t    start(AudioSystem::sync_event_t event =
+                                            AudioSystem::SYNC_EVENT_NONE,
                                      int triggerSession = 0);
             virtual void        stop();
                     void        pause();
@@ -810,6 +823,7 @@
 
         // implement FastMixerState::VolumeProvider interface
             virtual uint32_t    getVolumeLR();
+
             virtual status_t    setSyncEvent(const sp<SyncEvent>& event);
 
         protected:
@@ -822,7 +836,8 @@
                                 Track& operator = (const Track&);
 
             // AudioBufferProvider interface
-            virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS);
+            virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
+                                           int64_t pts = kInvalidPTS);
             // releaseBuffer() not overridden
 
             virtual size_t framesReady() const;
@@ -847,12 +862,16 @@
 
             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:
 
@@ -876,8 +895,9 @@
             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
+            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;
 
@@ -904,7 +924,7 @@
                                          uint32_t sampleRate,
                                          audio_format_t format,
                                          audio_channel_mask_t channelMask,
-                                         int frameCount,
+                                         size_t frameCount,
                                          const sp<IMemory>& sharedBuffer,
                                          int sessionId);
             virtual ~TimedTrack();
@@ -947,7 +967,7 @@
                        uint32_t sampleRate,
                        audio_format_t format,
                        audio_channel_mask_t channelMask,
-                       int frameCount,
+                       size_t frameCount,
                        const sp<IMemory>& sharedBuffer,
                        int sessionId);
 
@@ -982,11 +1002,11 @@
         };
 
 
-        // playback track
+        // playback track, used by DuplicatingThread
         class OutputTrack : public Track {
         public:
 
-            class Buffer: public AudioBufferProvider::Buffer {
+            class Buffer : public AudioBufferProvider::Buffer {
             public:
                 int16_t *mBuffer;
             };
@@ -996,10 +1016,11 @@
                                         uint32_t sampleRate,
                                         audio_format_t format,
                                         audio_channel_mask_t channelMask,
-                                        int frameCount);
+                                        size_t frameCount);
             virtual             ~OutputTrack();
 
-            virtual status_t    start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
+            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);
@@ -1013,7 +1034,8 @@
                 NO_MORE_BUFFERS = 0x80000001,   // same in AudioTrack.h, ok to be different value
             };
 
-            status_t            obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs);
+            status_t            obtainBuffer(AudioBufferProvider::Buffer* buffer,
+                                             uint32_t waitTimeMs);
             void                clearBufferQueue();
 
             // Maximum number of pending buffers allocated by OutputTrack::write()
@@ -1023,10 +1045,11 @@
             AudioBufferProvider::Buffer mOutBuffer;
             bool                        mActive;
             DuplicatingThread* const mSourceThread; // for waitTimeMs() in write()
+            void*                       mBuffers;   // starting address of buffers in plain memory
         };  // end of OutputTrack
 
-        PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
-                        audio_io_handle_t id, audio_devices_t device, type_t type);
+        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);
@@ -1078,10 +1101,10 @@
                                     uint32_t sampleRate,
                                     audio_format_t format,
                                     audio_channel_mask_t channelMask,
-                                    int frameCount,
+                                    size_t frameCount,
                                     const sp<IMemory>& sharedBuffer,
                                     int sessionId,
-                                    IAudioFlinger::track_flags_t flags,
+                                    IAudioFlinger::track_flags_t *flags,
                                     pid_t tid,
                                     status_t *status);
 
@@ -1104,7 +1127,7 @@
 
         virtual     String8     getParameters(const String8& keys);
         virtual     void        audioConfigChanged_l(int event, int param = 0);
-                    status_t    getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
+                    status_t    getRenderPosition(size_t *halFrames, size_t *dspFrames);
                     int16_t     *mixBuffer() const { return mMixBuffer; };
 
         virtual     void detachAuxEffect_l(int effectId);
@@ -1134,7 +1157,9 @@
         // 'volatile' means accessed via atomic operations and no lock.
         volatile int32_t                mSuspended;
 
-        int                             mBytesWritten;
+        // 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
@@ -1166,7 +1191,7 @@
         // Cache various calculated values, at threadLoop() entry and after a parameter change
         virtual     void        cacheParameters_l();
 
-        virtual     uint32_t    correctLatency(uint32_t latency) const;
+        virtual     uint32_t    correctLatency_l(uint32_t latency) const;
 
     private:
 
@@ -1185,7 +1210,8 @@
         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
+        // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by
+        // DuplicatingThread
         stream_type_t                   mStreamTypes[AUDIO_STREAM_CNT + 1];
         AudioStreamOut                  *mOutput;
 
@@ -1248,11 +1274,11 @@
 
     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);
+        MixerThread(const sp<AudioFlinger>& audioFlinger,
+                    AudioStreamOut* output,
+                    audio_io_handle_t id,
+                    audio_devices_t device,
+                    type_t type = MIXER);
         virtual             ~MixerThread();
 
         // Thread virtuals
@@ -1274,7 +1300,7 @@
         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;
+        virtual     uint32_t    correctLatency_l(uint32_t latency) const;
 
                     AudioMixer* mAudioMixer;    // normal mixer
     private:
@@ -1305,8 +1331,8 @@
     class DirectOutputThread : public PlaybackThread {
     public:
 
-        DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
-                            audio_io_handle_t id, audio_devices_t device);
+        DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+                           audio_io_handle_t id, audio_devices_t device);
         virtual                 ~DirectOutputThread();
 
         // Thread virtuals
@@ -1326,11 +1352,11 @@
         virtual     void        threadLoop_mix();
         virtual     void        threadLoop_sleepTime();
 
+    private:
         // 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:
@@ -1339,8 +1365,8 @@
 
     class DuplicatingThread : public MixerThread {
     public:
-        DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread,
-                           audio_io_handle_t id);
+        DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread,
+                          audio_io_handle_t id);
         virtual                 ~DuplicatingThread();
 
         // Thread virtuals
@@ -1441,7 +1467,7 @@
                                         uint32_t sampleRate,
                                         audio_format_t format,
                                         audio_channel_mask_t channelMask,
-                                        int frameCount,
+                                        size_t frameCount,
                                         int sessionId);
             virtual             ~RecordTrack();
 
@@ -1453,11 +1479,14 @@
                     // 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; }
+                    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
 
@@ -1465,7 +1494,8 @@
                                 RecordTrack& operator = (const RecordTrack&);
 
             // AudioBufferProvider interface
-            virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS);
+            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
@@ -1476,7 +1506,8 @@
                         uint32_t sampleRate,
                         audio_channel_mask_t channelMask,
                         audio_io_handle_t id,
-                        audio_devices_t device);
+                        audio_devices_t device,
+                        const sp<NBAIO_Sink>& teeSink);
                 virtual     ~RecordThread();
 
         // no addTrack_l ?
@@ -1499,7 +1530,7 @@
                         uint32_t sampleRate,
                         audio_format_t format,
                         audio_channel_mask_t channelMask,
-                        int frameCount,
+                        size_t frameCount,
                         int sessionId,
                         IAudioFlinger::track_flags_t flags,
                         pid_t tid,
@@ -1562,7 +1593,7 @@
                 int16_t                             *mRsmpInBuffer;
                 size_t                              mRsmpInIndex;
                 size_t                              mInputBytes;
-                const int                           mReqChannelCount;
+                const uint32_t                      mReqChannelCount;
                 const uint32_t                      mReqSampleRate;
                 ssize_t                             mBytesRead;
                 // sync event triggering actual audio capture. Frames read before this event will
@@ -1572,6 +1603,9 @@
                 // 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;
     };
 
     // server side of the client's IAudioRecord
@@ -1607,7 +1641,7 @@
     // 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 {
+    class EffectModule : public RefBase {
     public:
         EffectModule(ThreadBase *thread,
                         const wp<AudioFlinger::EffectChain>& chain,
@@ -1781,7 +1815,8 @@
         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
+        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
@@ -1794,11 +1829,11 @@
     // 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 {
+    // 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);
@@ -2064,6 +2099,13 @@
     // 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);
 };
 
 
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/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index ffea9b9..2c3c719 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -82,10 +82,8 @@
     switch (quality) {
     case DEFAULT_QUALITY:
     case LOW_QUALITY:
-#if 0   // these have not been qualified recently so are not supported unless explicitly requested
     case MED_QUALITY:
     case HIGH_QUALITY:
-#endif
     case VERY_HIGH_QUALITY:
         return true;
     default:
@@ -190,12 +188,10 @@
         ALOGV("Create linear Resampler");
         resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
         break;
-#if 0   // disabled because it has not been qualified recently, if requested will use default:
     case MED_QUALITY:
         ALOGV("Create cubic Resampler");
         resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
         break;
-#endif
     case HIGH_QUALITY:
         ALOGV("Create HIGH_QUALITY sinc Resampler");
         resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate);
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index 9e8447a..3f22ca6 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -17,13 +17,33 @@
 #define LOG_TAG "AudioResamplerSinc"
 //#define LOG_NDEBUG 0
 
+#include <malloc.h>
 #include <string.h>
-#include "AudioResamplerSinc.h"
-#include <dlfcn.h>
-#include <cutils/properties.h>
 #include <stdlib.h>
+#include <dlfcn.h>
+
+#include <cutils/compiler.h>
+#include <cutils/properties.h>
+
 #include <utils/Log.h>
 
+#include "AudioResamplerSinc.h"
+
+
+
+#if defined(__arm__) && !defined(__thumb__)
+#define USE_INLINE_ASSEMBLY (true)
+#else
+#define USE_INLINE_ASSEMBLY (false)
+#endif
+
+#if USE_INLINE_ASSEMBLY && defined(__ARM_NEON__)
+#define USE_NEON (true)
+#else
+#define USE_NEON (false)
+#endif
+
+
 namespace android {
 // ----------------------------------------------------------------------------
 
@@ -31,37 +51,274 @@
 /*
  * These coeficients are computed with the "fir" utility found in
  * tools/resampler_tools
- * TODO: A good optimization would be to transpose this matrix, to take
- * better advantage of the data-cache.
+ * cmd-line: fir -l 7 -s 48000 -c 20478
  */
-const int32_t AudioResamplerSinc::mFirCoefsUp[] = {
-        0x7fffffff, 0x7f15d078, 0x7c5e0da6, 0x77ecd867, 0x71e2e251, 0x6a6c304a, 0x61be7269, 0x58170412, 0x4db8ab05, 0x42e92ea6, 0x37eee214, 0x2d0e3bb1, 0x22879366, 0x18951e95, 0x0f693d0d, 0x072d2621,
-        0x00000000, 0xf9f66655, 0xf51a5fd7, 0xf16bbd84, 0xeee0d9ac, 0xed67a922, 0xece70de6, 0xed405897, 0xee50e505, 0xeff3be30, 0xf203370f, 0xf45a6741, 0xf6d67d53, 0xf957db66, 0xfbc2f647, 0xfe00f2b9,
-        0x00000000, 0x01b37218, 0x0313a0c6, 0x041d930d, 0x04d28057, 0x053731b0, 0x05534dff, 0x05309bfd, 0x04da440d, 0x045c1aee, 0x03c1fcdd, 0x03173ef5, 0x02663ae8, 0x01b7f736, 0x0113ec79, 0x007fe6a9,
-        0x00000000, 0xff96b229, 0xff44f99f, 0xff0a86be, 0xfee5f803, 0xfed518fd, 0xfed521fd, 0xfee2f4fd, 0xfefb54f8, 0xff1b159b, 0xff3f4203, 0xff6539e0, 0xff8ac502, 0xffae1ddd, 0xffcdf3f9, 0xffe96798,
-        0x00000000, 0x00119de6, 0x001e6b7e, 0x0026cb7a, 0x002b4830, 0x002c83d6, 0x002b2a82, 0x0027e67a, 0x002356f9, 0x001e098e, 0x001875e4, 0x0012fbbe, 0x000de2d1, 0x00095c10, 0x00058414, 0x00026636,
-        0x00000000, 0xfffe44a9, 0xfffd206d, 0xfffc7b7f, 0xfffc3c8f, 0xfffc4ac2, 0xfffc8f2b, 0xfffcf5c4, 0xfffd6df3, 0xfffdeab2, 0xfffe6275, 0xfffececf, 0xffff2c07, 0xffff788c, 0xffffb471, 0xffffe0f2,
-        0x00000000, 0x000013e6, 0x00001f03, 0x00002396, 0x00002399, 0x000020b6, 0x00001c3c, 0x00001722, 0x00001216, 0x00000d81, 0x0000099c, 0x0000067c, 0x00000419, 0x0000025f, 0x00000131, 0x00000070,
-        0x00000000, 0xffffffc7, 0xffffffb3, 0xffffffb3, 0xffffffbe, 0xffffffcd, 0xffffffdb, 0xffffffe7, 0xfffffff0, 0xfffffff7, 0xfffffffb, 0xfffffffe, 0xffffffff, 0x00000000, 0x00000000, 0x00000000,
-        0x00000000 // this one is needed for lerping the last coefficient
+const int32_t AudioResamplerSinc::mFirCoefsUp[] __attribute__ ((aligned (32))) = {
+        0x6d374bc7, 0x111c6ba0, 0xf3240e61, 0x07d14a38, 0xfc509e64, 0x0139cee9, 0xffc8c866, 0xfffcc300,
+        0x6d35278a, 0x103e8192, 0xf36b9dfd, 0x07bdfaa5, 0xfc5102d0, 0x013d618d, 0xffc663b9, 0xfffd9592,
+        0x6d2ebafe, 0x0f62811a, 0xf3b3d8ac, 0x07a9f399, 0xfc51d9a6, 0x0140bea5, 0xffc41212, 0xfffe631e,
+        0x6d24069d, 0x0e8875ad, 0xf3fcb43e, 0x07953976, 0xfc53216f, 0x0143e67c, 0xffc1d373, 0xffff2b9f,
+        0x6d150b35, 0x0db06a89, 0xf4462690, 0x077fd0ac, 0xfc54d8ae, 0x0146d965, 0xffbfa7d9, 0xffffef10,
+        0x6d01c9e3, 0x0cda6ab5, 0xf4902587, 0x0769bdaf, 0xfc56fdda, 0x014997bb, 0xffbd8f40, 0x0000ad6e,
+        0x6cea4418, 0x0c0680fe, 0xf4daa718, 0x07530501, 0xfc598f60, 0x014c21db, 0xffbb89a1, 0x000166b6,
+        0x6cce7b97, 0x0b34b7f5, 0xf525a143, 0x073bab28, 0xfc5c8ba5, 0x014e782a, 0xffb996f3, 0x00021ae5,
+        0x6cae7272, 0x0a6519f4, 0xf5710a17, 0x0723b4b4, 0xfc5ff105, 0x01509b14, 0xffb7b728, 0x0002c9fd,
+        0x6c8a2b0f, 0x0997b116, 0xf5bcd7b1, 0x070b2639, 0xfc63bdd3, 0x01528b08, 0xffb5ea31, 0x000373fb,
+        0x6c61a823, 0x08cc873c, 0xf609003f, 0x06f20453, 0xfc67f05a, 0x0154487b, 0xffb42ffc, 0x000418e2,
+        0x6c34ecb5, 0x0803a60a, 0xf6557a00, 0x06d853a2, 0xfc6c86dd, 0x0155d3e8, 0xffb28876, 0x0004b8b3,
+        0x6c03fc1c, 0x073d16e7, 0xf6a23b44, 0x06be18cd, 0xfc717f97, 0x01572dcf, 0xffb0f388, 0x00055371,
+        0x6bced9ff, 0x0678e2fc, 0xf6ef3a6e, 0x06a3587e, 0xfc76d8bc, 0x015856b6, 0xffaf7118, 0x0005e921,
+        0x6b958a54, 0x05b71332, 0xf73c6df4, 0x06881761, 0xfc7c9079, 0x01594f25, 0xffae010b, 0x000679c5,
+        0x6b581163, 0x04f7b037, 0xf789cc61, 0x066c5a27, 0xfc82a4f4, 0x015a17ab, 0xffaca344, 0x00070564,
+        0x6b1673c1, 0x043ac276, 0xf7d74c53, 0x06502583, 0xfc89144d, 0x015ab0db, 0xffab57a1, 0x00078c04,
+        0x6ad0b652, 0x0380521c, 0xf824e480, 0x06337e2a, 0xfc8fdc9f, 0x015b1b4e, 0xffaa1e02, 0x00080dab,
+        0x6a86de48, 0x02c86715, 0xf8728bb3, 0x061668d2, 0xfc96fbfc, 0x015b579e, 0xffa8f641, 0x00088a62,
+        0x6a38f123, 0x0213090c, 0xf8c038d0, 0x05f8ea30, 0xfc9e7074, 0x015b666c, 0xffa7e039, 0x00090230,
+        0x69e6f4b1, 0x01603f6e, 0xf90de2d1, 0x05db06fc, 0xfca63810, 0x015b485b, 0xffa6dbc0, 0x0009751e,
+        0x6990ef0b, 0x00b01162, 0xf95b80cb, 0x05bcc3ed, 0xfcae50d6, 0x015afe14, 0xffa5e8ad, 0x0009e337,
+        0x6936e697, 0x000285d0, 0xf9a909ea, 0x059e25b5, 0xfcb6b8c4, 0x015a8843, 0xffa506d2, 0x000a4c85,
+        0x68d8e206, 0xff57a35e, 0xf9f67577, 0x057f310a, 0xfcbf6dd8, 0x0159e796, 0xffa43603, 0x000ab112,
+        0x6876e855, 0xfeaf706f, 0xfa43bad2, 0x055fea9d, 0xfcc86e09, 0x01591cc0, 0xffa3760e, 0x000b10ec,
+        0x681100c9, 0xfe09f323, 0xfa90d17b, 0x0540571a, 0xfcd1b74c, 0x01582878, 0xffa2c6c2, 0x000b6c1d,
+        0x67a732f4, 0xfd673159, 0xfaddb10c, 0x05207b2f, 0xfcdb4793, 0x01570b77, 0xffa227ec, 0x000bc2b3,
+        0x673986ac, 0xfcc730aa, 0xfb2a513b, 0x05005b82, 0xfce51ccb, 0x0155c678, 0xffa19957, 0x000c14bb,
+        0x66c80413, 0xfc29f670, 0xfb76a9dd, 0x04dffcb6, 0xfcef34e1, 0x01545a3c, 0xffa11acb, 0x000c6244,
+        0x6652b392, 0xfb8f87bd, 0xfbc2b2e4, 0x04bf6369, 0xfcf98dbe, 0x0152c783, 0xffa0ac11, 0x000cab5c,
+        0x65d99dd5, 0xfaf7e963, 0xfc0e6461, 0x049e9433, 0xfd04254a, 0x01510f13, 0xffa04cf0, 0x000cf012,
+        0x655ccbd3, 0xfa631fef, 0xfc59b685, 0x047d93a8, 0xfd0ef969, 0x014f31b2, 0xff9ffd2c, 0x000d3075,
+        0x64dc46c3, 0xf9d12fab, 0xfca4a19f, 0x045c6654, 0xfd1a0801, 0x014d3029, 0xff9fbc89, 0x000d6c97,
+        0x64581823, 0xf9421c9d, 0xfcef1e20, 0x043b10bd, 0xfd254ef4, 0x014b0b45, 0xff9f8ac9, 0x000da486,
+        0x63d049b4, 0xf8b5ea87, 0xfd392498, 0x04199760, 0xfd30cc24, 0x0148c3d2, 0xff9f67ae, 0x000dd854,
+        0x6344e578, 0xf82c9ce7, 0xfd82adba, 0x03f7feb4, 0xfd3c7d73, 0x01465a9f, 0xff9f52f7, 0x000e0812,
+        0x62b5f5b2, 0xf7a636fa, 0xfdcbb25a, 0x03d64b27, 0xfd4860c2, 0x0143d07f, 0xff9f4c65, 0x000e33d3,
+        0x622384e8, 0xf722bbb5, 0xfe142b6e, 0x03b4811d, 0xfd5473f3, 0x01412643, 0xff9f53b4, 0x000e5ba7,
+        0x618d9ddc, 0xf6a22dcf, 0xfe5c120f, 0x0392a4f4, 0xfd60b4e7, 0x013e5cc0, 0xff9f68a1, 0x000e7fa1,
+        0x60f44b91, 0xf6248fb6, 0xfea35f79, 0x0370bafc, 0xfd6d2180, 0x013b74ca, 0xff9f8ae9, 0x000e9fd5,
+        0x60579947, 0xf5a9e398, 0xfeea0d0c, 0x034ec77f, 0xfd79b7a1, 0x01386f3a, 0xff9fba47, 0x000ebc54,
+        0x5fb79278, 0xf5322b61, 0xff30144a, 0x032ccebb, 0xfd86752e, 0x01354ce7, 0xff9ff674, 0x000ed533,
+        0x5f1442dc, 0xf4bd68b6, 0xff756edc, 0x030ad4e1, 0xfd93580d, 0x01320ea9, 0xffa03f2b, 0x000eea84,
+        0x5e6db665, 0xf44b9cfe, 0xffba168d, 0x02e8de19, 0xfda05e23, 0x012eb55a, 0xffa09425, 0x000efc5c,
+        0x5dc3f93c, 0xf3dcc959, 0xfffe054e, 0x02c6ee7f, 0xfdad855b, 0x012b41d3, 0xffa0f519, 0x000f0ace,
+        0x5d1717c4, 0xf370eea9, 0x00413536, 0x02a50a22, 0xfdbacb9e, 0x0127b4f1, 0xffa161bf, 0x000f15ef,
+        0x5c671e96, 0xf3080d8c, 0x0083a081, 0x02833506, 0xfdc82edb, 0x01240f8e, 0xffa1d9cf, 0x000f1dd2,
+        0x5bb41a80, 0xf2a2265e, 0x00c54190, 0x02617321, 0xfdd5ad01, 0x01205285, 0xffa25cfe, 0x000f228d,
+        0x5afe1886, 0xf23f393b, 0x010612eb, 0x023fc85c, 0xfde34403, 0x011c7eb2, 0xffa2eb04, 0x000f2434,
+        0x5a4525df, 0xf1df45fd, 0x01460f41, 0x021e3891, 0xfdf0f1d6, 0x011894f0, 0xffa38395, 0x000f22dc,
+        0x59894ff3, 0xf1824c3e, 0x01853165, 0x01fcc78f, 0xfdfeb475, 0x0114961b, 0xffa42668, 0x000f1e99,
+        0x58caa45b, 0xf1284b58, 0x01c37452, 0x01db7914, 0xfe0c89db, 0x0110830f, 0xffa4d332, 0x000f1781,
+        0x580930e1, 0xf0d14267, 0x0200d32c, 0x01ba50d2, 0xfe1a7009, 0x010c5ca6, 0xffa589a6, 0x000f0da8,
+        0x5745037c, 0xf07d3043, 0x023d493c, 0x0199526b, 0xfe286505, 0x010823ba, 0xffa6497c, 0x000f0125,
+        0x567e2a51, 0xf02c138a, 0x0278d1f2, 0x01788170, 0xfe3666d5, 0x0103d927, 0xffa71266, 0x000ef20b,
+        0x55b4b3af, 0xefddea9a, 0x02b368e6, 0x0157e166, 0xfe447389, 0x00ff7dc4, 0xffa7e41a, 0x000ee070,
+        0x54e8ae13, 0xef92b393, 0x02ed09d7, 0x013775bf, 0xfe528931, 0x00fb126b, 0xffa8be4c, 0x000ecc69,
+        0x541a281e, 0xef4a6c58, 0x0325b0ad, 0x011741df, 0xfe60a5e5, 0x00f697f3, 0xffa9a0b1, 0x000eb60b,
+        0x5349309e, 0xef051290, 0x035d5977, 0x00f7491a, 0xfe6ec7c0, 0x00f20f32, 0xffaa8afe, 0x000e9d6b,
+        0x5275d684, 0xeec2a3a3, 0x0394006a, 0x00d78eb3, 0xfe7cece2, 0x00ed78ff, 0xffab7ce7, 0x000e829e,
+        0x51a028e8, 0xee831cc3, 0x03c9a1e5, 0x00b815da, 0xfe8b1373, 0x00e8d62d, 0xffac7621, 0x000e65ba,
+        0x50c83704, 0xee467ae1, 0x03fe3a6f, 0x0098e1b3, 0xfe99399f, 0x00e4278f, 0xffad7662, 0x000e46d3,
+        0x4fee1037, 0xee0cbab9, 0x0431c6b5, 0x0079f54c, 0xfea75d97, 0x00df6df7, 0xffae7d5f, 0x000e25fd,
+        0x4f11c3fe, 0xedd5d8ca, 0x0464438c, 0x005b53a4, 0xfeb57d92, 0x00daaa34, 0xffaf8acd, 0x000e034f,
+        0x4e3361f7, 0xeda1d15c, 0x0495adf2, 0x003cffa9, 0xfec397cf, 0x00d5dd16, 0xffb09e63, 0x000ddedb,
+        0x4d52f9df, 0xed70a07d, 0x04c6030d, 0x001efc35, 0xfed1aa92, 0x00d10769, 0xffb1b7d8, 0x000db8b7,
+        0x4c709b8e, 0xed424205, 0x04f54029, 0x00014c12, 0xfedfb425, 0x00cc29f7, 0xffb2d6e1, 0x000d90f6,
+        0x4b8c56f8, 0xed16b196, 0x052362ba, 0xffe3f1f7, 0xfeedb2da, 0x00c7458a, 0xffb3fb37, 0x000d67ae,
+        0x4aa63c2c, 0xecedea99, 0x0550685d, 0xffc6f08a, 0xfefba508, 0x00c25ae8, 0xffb52490, 0x000d3cf1,
+        0x49be5b50, 0xecc7e845, 0x057c4ed4, 0xffaa4a5d, 0xff09890f, 0x00bd6ad7, 0xffb652a7, 0x000d10d5,
+        0x48d4c4a2, 0xeca4a59b, 0x05a7140b, 0xff8e01f1, 0xff175d53, 0x00b87619, 0xffb78533, 0x000ce36b,
+        0x47e98874, 0xec841d68, 0x05d0b612, 0xff7219b3, 0xff252042, 0x00b37d70, 0xffb8bbed, 0x000cb4c8,
+        0x46fcb72d, 0xec664a48, 0x05f93324, 0xff5693fe, 0xff32d04f, 0x00ae8198, 0xffb9f691, 0x000c84ff,
+        0x460e6148, 0xec4b26a2, 0x0620899e, 0xff3b731b, 0xff406bf8, 0x00a9834e, 0xffbb34d8, 0x000c5422,
+        0x451e9750, 0xec32acb0, 0x0646b808, 0xff20b93e, 0xff4df1be, 0x00a4834c, 0xffbc767f, 0x000c2245,
+        0x442d69de, 0xec1cd677, 0x066bbd0d, 0xff066889, 0xff5b602c, 0x009f8249, 0xffbdbb42, 0x000bef79,
+        0x433ae99c, 0xec099dcf, 0x068f9781, 0xfeec830d, 0xff68b5d5, 0x009a80f8, 0xffbf02dd, 0x000bbbd2,
+        0x4247273f, 0xebf8fc64, 0x06b2465b, 0xfed30ac5, 0xff75f153, 0x0095800c, 0xffc04d0f, 0x000b8760,
+        0x41523389, 0xebeaebaf, 0x06d3c8bb, 0xfeba0199, 0xff831148, 0x00908034, 0xffc19996, 0x000b5235,
+        0x405c1f43, 0xebdf6500, 0x06f41de3, 0xfea16960, 0xff90145e, 0x008b821b, 0xffc2e832, 0x000b1c64,
+        0x3f64fb40, 0xebd6617b, 0x0713453d, 0xfe8943dc, 0xff9cf947, 0x0086866b, 0xffc438a3, 0x000ae5fc,
+        0x3e6cd85b, 0xebcfda19, 0x07313e56, 0xfe7192bd, 0xffa9bebe, 0x00818dcb, 0xffc58aaa, 0x000aaf0f,
+        0x3d73c772, 0xebcbc7a7, 0x074e08e0, 0xfe5a579d, 0xffb66386, 0x007c98de, 0xffc6de09, 0x000a77ac,
+        0x3c79d968, 0xebca22cc, 0x0769a4b2, 0xfe439407, 0xffc2e669, 0x0077a845, 0xffc83285, 0x000a3fe5,
+        0x3b7f1f23, 0xebcae405, 0x078411c7, 0xfe2d496f, 0xffcf463a, 0x0072bc9d, 0xffc987e0, 0x000a07c9,
+        0x3a83a989, 0xebce03aa, 0x079d503b, 0xfe177937, 0xffdb81d6, 0x006dd680, 0xffcadde1, 0x0009cf67,
+        0x3987897f, 0xebd379eb, 0x07b56051, 0xfe0224b0, 0xffe79820, 0x0068f687, 0xffcc344c, 0x000996ce,
+        0x388acfe9, 0xebdb3ed5, 0x07cc426c, 0xfded4d13, 0xfff38806, 0x00641d44, 0xffcd8aeb, 0x00095e0e,
+        0x378d8da8, 0xebe54a4f, 0x07e1f712, 0xfdd8f38b, 0xffff507b, 0x005f4b4a, 0xffcee183, 0x00092535,
+        0x368fd397, 0xebf1941f, 0x07f67eec, 0xfdc5192d, 0x000af07f, 0x005a8125, 0xffd037e0, 0x0008ec50,
+        0x3591b28b, 0xec0013e8, 0x0809dac3, 0xfdb1befc, 0x00166718, 0x0055bf60, 0xffd18dcc, 0x0008b36e,
+        0x34933b50, 0xec10c12c, 0x081c0b84, 0xfd9ee5e7, 0x0021b355, 0x00510682, 0xffd2e311, 0x00087a9c,
+        0x33947eab, 0xec23934f, 0x082d1239, 0xfd8c8ecc, 0x002cd44d, 0x004c570f, 0xffd4377d, 0x000841e8,
+        0x32958d55, 0xec388194, 0x083cf010, 0xfd7aba74, 0x0037c922, 0x0047b186, 0xffd58ade, 0x0008095d,
+        0x319677fa, 0xec4f8322, 0x084ba654, 0xfd696998, 0x004290fc, 0x00431666, 0xffd6dd02, 0x0007d108,
+        0x30974f3b, 0xec688f02, 0x08593671, 0xfd589cdc, 0x004d2b0e, 0x003e8628, 0xffd82dba, 0x000798f5,
+        0x2f9823a8, 0xec839c22, 0x0865a1f1, 0xfd4854d3, 0x00579691, 0x003a0141, 0xffd97cd6, 0x00076130,
+        0x2e9905c1, 0xeca0a156, 0x0870ea7e, 0xfd3891fd, 0x0061d2ca, 0x00358824, 0xffdaca2a, 0x000729c4,
+        0x2d9a05f4, 0xecbf9558, 0x087b11de, 0xfd2954c8, 0x006bdf05, 0x00311b41, 0xffdc1588, 0x0006f2bb,
+        0x2c9b349e, 0xece06ecb, 0x088419f6, 0xfd1a9d91, 0x0075ba95, 0x002cbb03, 0xffdd5ec6, 0x0006bc21,
+        0x2b9ca203, 0xed032439, 0x088c04c8, 0xfd0c6ca2, 0x007f64da, 0x002867d2, 0xffdea5bb, 0x000685ff,
+        0x2a9e5e57, 0xed27ac16, 0x0892d470, 0xfcfec233, 0x0088dd38, 0x00242213, 0xffdfea3c, 0x0006505f,
+        0x29a079b2, 0xed4dfcc2, 0x08988b2a, 0xfcf19e6b, 0x0092231e, 0x001fea27, 0xffe12c22, 0x00061b4b,
+        0x28a30416, 0xed760c88, 0x089d2b4a, 0xfce50161, 0x009b3605, 0x001bc06b, 0xffe26b48, 0x0005e6cb,
+        0x27a60d6a, 0xed9fd1a2, 0x08a0b740, 0xfcd8eb17, 0x00a4156b, 0x0017a53b, 0xffe3a788, 0x0005b2e8,
+        0x26a9a57b, 0xedcb4237, 0x08a33196, 0xfccd5b82, 0x00acc0da, 0x001398ec, 0xffe4e0bf, 0x00057faa,
+        0x25addbf9, 0xedf8545b, 0x08a49cf0, 0xfcc25285, 0x00b537e1, 0x000f9bd2, 0xffe616c8, 0x00054d1a,
+        0x24b2c075, 0xee26fe17, 0x08a4fc0d, 0xfcb7cff0, 0x00bd7a1c, 0x000bae3c, 0xffe74984, 0x00051b3e,
+        0x23b86263, 0xee573562, 0x08a451c0, 0xfcadd386, 0x00c5872a, 0x0007d075, 0xffe878d3, 0x0004ea1d,
+        0x22bed116, 0xee88f026, 0x08a2a0f8, 0xfca45cf7, 0x00cd5eb7, 0x000402c8, 0xffe9a494, 0x0004b9c0,
+        0x21c61bc0, 0xeebc2444, 0x089fecbb, 0xfc9b6be5, 0x00d50075, 0x00004579, 0xffeaccaa, 0x00048a2b,
+        0x20ce516f, 0xeef0c78d, 0x089c3824, 0xfc92ffe1, 0x00dc6c1e, 0xfffc98c9, 0xffebf0fa, 0x00045b65,
+        0x1fd7810f, 0xef26cfca, 0x08978666, 0xfc8b186d, 0x00e3a175, 0xfff8fcf7, 0xffed1166, 0x00042d74,
+        0x1ee1b965, 0xef5e32bd, 0x0891dac8, 0xfc83b4fc, 0x00eaa045, 0xfff5723d, 0xffee2dd7, 0x0004005e,
+        0x1ded0911, 0xef96e61c, 0x088b38a9, 0xfc7cd4f0, 0x00f16861, 0xfff1f8d2, 0xffef4632, 0x0003d426,
+        0x1cf97e8b, 0xefd0df9a, 0x0883a378, 0xfc76779e, 0x00f7f9a3, 0xffee90eb, 0xfff05a60, 0x0003a8d2,
+        0x1c072823, 0xf00c14e1, 0x087b1ebc, 0xfc709c4d, 0x00fe53ef, 0xffeb3ab8, 0xfff16a4a, 0x00037e65,
+        0x1b1613ff, 0xf0487b98, 0x0871ae0d, 0xfc6b4233, 0x0104772e, 0xffe7f666, 0xfff275db, 0x000354e5,
+        0x1a26501b, 0xf0860962, 0x08675516, 0xfc66687a, 0x010a6353, 0xffe4c41e, 0xfff37d00, 0x00032c54,
+        0x1937ea47, 0xf0c4b3e0, 0x085c1794, 0xfc620e3d, 0x01101858, 0xffe1a408, 0xfff47fa5, 0x000304b7,
+        0x184af025, 0xf10470b0, 0x084ff957, 0xfc5e328c, 0x0115963d, 0xffde9646, 0xfff57db8, 0x0002de0e,
+        0x175f6f2b, 0xf1453571, 0x0842fe3d, 0xfc5ad465, 0x011add0b, 0xffdb9af8, 0xfff67729, 0x0002b85f,
+        0x1675749e, 0xf186f7c0, 0x08352a35, 0xfc57f2be, 0x011fecd3, 0xffd8b23b, 0xfff76be9, 0x000293aa,
+        0x158d0d95, 0xf1c9ad40, 0x0826813e, 0xfc558c7c, 0x0124c5ab, 0xffd5dc28, 0xfff85be8, 0x00026ff2,
+        0x14a646f6, 0xf20d4b92, 0x08170767, 0xfc53a07b, 0x012967b1, 0xffd318d6, 0xfff9471b, 0x00024d39,
+        0x13c12d73, 0xf251c85d, 0x0806c0cb, 0xfc522d88, 0x012dd30a, 0xffd06858, 0xfffa2d74, 0x00022b7f,
+        0x12ddcd8f, 0xf297194d, 0x07f5b193, 0xfc513266, 0x013207e4, 0xffcdcabe, 0xfffb0ee9, 0x00020ac7,
+        0x11fc3395, 0xf2dd3411, 0x07e3ddf7, 0xfc50adcc, 0x01360670, 0xffcb4014, 0xfffbeb70, 0x0001eb10,
+        0x111c6ba0, 0xf3240e61, 0x07d14a38, 0xfc509e64, 0x0139cee9, 0xffc8c866, 0xfffcc300, 0x0001cc5c,
 };
 
 /*
- * These coefficients are optimized for 48KHz -> 44.1KHz (stop-band at 22.050KHz)
- * It's possible to use the above coefficient for any down-sampling
- * at the expense of a slower processing loop (we can interpolate
- * these coefficient from the above by "Stretching" them in time).
+ * These coefficients are optimized for 48KHz -> 44.1KHz
+ * cmd-line: fir -l 7 -s 48000 -c 17189
  */
-const int32_t AudioResamplerSinc::mFirCoefsDown[] = {
-        0x7fffffff, 0x7f55e46d, 0x7d5b4c60, 0x7a1b4b98, 0x75a7fb14, 0x7019f0bd, 0x698f875a, 0x622bfd59, 0x5a167256, 0x5178cc54, 0x487e8e6c, 0x3f53aae8, 0x36235ad4, 0x2d17047b, 0x245539ab, 0x1c00d540,
-        0x14383e57, 0x0d14d5ca, 0x06aa910b, 0x0107c38b, 0xfc351654, 0xf835abae, 0xf5076b45, 0xf2a37202, 0xf0fe9faa, 0xf00a3bbd, 0xefb4aa81, 0xefea2b05, 0xf0959716, 0xf1a11e83, 0xf2f6f7a0, 0xf481fff4,
-        0xf62e48ce, 0xf7e98ca5, 0xf9a38b4c, 0xfb4e4bfa, 0xfcde456f, 0xfe4a6d30, 0xff8c2fdf, 0x009f5555, 0x0181d393, 0x0233940f, 0x02b62f06, 0x030ca07d, 0x033afa62, 0x03461725, 0x03334f83, 0x030835fa,
-        0x02ca59cc, 0x027f12d1, 0x022b570d, 0x01d39a49, 0x017bb78f, 0x0126e414, 0x00d7aaaf, 0x008feec7, 0x0050f584, 0x001b73e3, 0xffefa063, 0xffcd46ed, 0xffb3ddcd, 0xffa29aaa, 0xff988691, 0xff949066,
-        0xff959d24, 0xff9a959e, 0xffa27195, 0xffac4011, 0xffb72d2b, 0xffc28569, 0xffcdb706, 0xffd85171, 0xffe20364, 0xffea97e9, 0xfff1f2b2, 0xfff80c06, 0xfffcec92, 0x0000a955, 0x00035fd8, 0x000532cf,
-        0x00064735, 0x0006c1f9, 0x0006c62d, 0x000673ba, 0x0005e68f, 0x00053630, 0x000475a3, 0x0003b397, 0x0002fac1, 0x00025257, 0x0001be9e, 0x0001417a, 0x0000dafd, 0x000089eb, 0x00004c28, 0x00001f1d,
-        0x00000000, 0xffffec10, 0xffffe0be, 0xffffdbc5, 0xffffdb39, 0xffffdd8b, 0xffffe182, 0xffffe638, 0xffffeb0a, 0xffffef8f, 0xfffff38b, 0xfffff6e3, 0xfffff993, 0xfffffba6, 0xfffffd30, 0xfffffe4a,
-        0xffffff09, 0xffffff85, 0xffffffd1, 0xfffffffb, 0x0000000f, 0x00000016, 0x00000015, 0x00000012, 0x0000000d, 0x00000009, 0x00000006, 0x00000003, 0x00000002, 0x00000001, 0x00000000, 0x00000000,
-        0x00000000 // this one is needed for lerping the last coefficient
+const int32_t AudioResamplerSinc::mFirCoefsDown[] __attribute__ ((aligned (32))) = {
+        0x5bacb6f4, 0x1ded1a1d, 0xf0398d56, 0x0394f674, 0x0193a5f9, 0xfe66dbeb, 0x00791043, 0xfffe6631,
+        0x5bab6c81, 0x1d3ddccd, 0xf0421d2c, 0x03af9995, 0x01818dc9, 0xfe6bb63e, 0x0079812a, 0xfffdc37d,
+        0x5ba78d37, 0x1c8f2cf9, 0xf04beb1d, 0x03c9a04a, 0x016f8aca, 0xfe70a511, 0x0079e34d, 0xfffd2545,
+        0x5ba1194f, 0x1be11231, 0xf056f2c7, 0x03e309fe, 0x015d9e64, 0xfe75a79f, 0x007a36e2, 0xfffc8b86,
+        0x5b981122, 0x1b3393f8, 0xf0632fb7, 0x03fbd625, 0x014bc9fa, 0xfe7abd23, 0x007a7c20, 0xfffbf639,
+        0x5b8c7530, 0x1a86b9bf, 0xf0709d74, 0x04140449, 0x013a0ee9, 0xfe7fe4db, 0x007ab33d, 0xfffb655b,
+        0x5b7e461a, 0x19da8ae5, 0xf07f3776, 0x042b93fd, 0x01286e86, 0xfe851e05, 0x007adc72, 0xfffad8e4,
+        0x5b6d84a8, 0x192f0eb7, 0xf08ef92d, 0x044284e6, 0x0116ea22, 0xfe8a67dd, 0x007af7f6, 0xfffa50ce,
+        0x5b5a31c6, 0x18844c70, 0xf09fddfe, 0x0458d6b7, 0x01058306, 0xfe8fc1a5, 0x007b0603, 0xfff9cd12,
+        0x5b444e81, 0x17da4b37, 0xf0b1e143, 0x046e8933, 0x00f43a74, 0xfe952a9b, 0x007b06d4, 0xfff94da9,
+        0x5b2bdc0e, 0x17311222, 0xf0c4fe50, 0x04839c29, 0x00e311a9, 0xfe9aa201, 0x007afaa1, 0xfff8d28c,
+        0x5b10dbc2, 0x1688a832, 0xf0d9306d, 0x04980f79, 0x00d209db, 0xfea02719, 0x007ae1a7, 0xfff85bb1,
+        0x5af34f18, 0x15e11453, 0xf0ee72db, 0x04abe310, 0x00c12439, 0xfea5b926, 0x007abc20, 0xfff7e910,
+        0x5ad337af, 0x153a5d5e, 0xf104c0d2, 0x04bf16e9, 0x00b061eb, 0xfeab576d, 0x007a8a49, 0xfff77a9f,
+        0x5ab09748, 0x14948a16, 0xf11c1583, 0x04d1ab0d, 0x009fc413, 0xfeb10134, 0x007a4c5d, 0xfff71057,
+        0x5a8b6fc7, 0x13efa12c, 0xf1346c17, 0x04e39f93, 0x008f4bcb, 0xfeb6b5c0, 0x007a029a, 0xfff6aa2b,
+        0x5a63c336, 0x134ba937, 0xf14dbfb1, 0x04f4f4a2, 0x007efa29, 0xfebc745c, 0x0079ad3d, 0xfff64812,
+        0x5a3993c0, 0x12a8a8bb, 0xf1680b6e, 0x0505aa6a, 0x006ed038, 0xfec23c50, 0x00794c82, 0xfff5ea02,
+        0x5a0ce3b2, 0x1206a625, 0xf1834a63, 0x0515c12d, 0x005ecf01, 0xfec80ce8, 0x0078e0a9, 0xfff58ff0,
+        0x59ddb57f, 0x1165a7cc, 0xf19f77a0, 0x05253938, 0x004ef782, 0xfecde571, 0x007869ee, 0xfff539cf,
+        0x59ac0bba, 0x10c5b3ef, 0xf1bc8e31, 0x053412e4, 0x003f4ab4, 0xfed3c538, 0x0077e891, 0xfff4e794,
+        0x5977e919, 0x1026d0b8, 0xf1da891b, 0x05424e9b, 0x002fc98a, 0xfed9ab8f, 0x00775ccf, 0xfff49934,
+        0x59415075, 0x0f890437, 0xf1f96360, 0x054feccf, 0x002074ed, 0xfedf97c6, 0x0076c6e8, 0xfff44ea3,
+        0x590844c9, 0x0eec5465, 0xf21917ff, 0x055cee03, 0x00114dc3, 0xfee58932, 0x00762719, 0xfff407d2,
+        0x58ccc930, 0x0e50c723, 0xf239a1ef, 0x056952c3, 0x000254e8, 0xfeeb7f27, 0x00757da3, 0xfff3c4b7,
+        0x588ee0ea, 0x0db6623b, 0xf25afc29, 0x05751baa, 0xfff38b32, 0xfef178fc, 0x0074cac4, 0xfff38542,
+        0x584e8f56, 0x0d1d2b5d, 0xf27d219f, 0x0580495c, 0xffe4f171, 0xfef7760c, 0x00740ebb, 0xfff34968,
+        0x580bd7f4, 0x0c85281f, 0xf2a00d43, 0x058adc8d, 0xffd6886d, 0xfefd75af, 0x007349c7, 0xfff3111b,
+        0x57c6be67, 0x0bee5dff, 0xf2c3ba04, 0x0594d5fa, 0xffc850e6, 0xff037744, 0x00727c27, 0xfff2dc4c,
+        0x577f4670, 0x0b58d262, 0xf2e822ce, 0x059e366c, 0xffba4b98, 0xff097a29, 0x0071a61b, 0xfff2aaef,
+        0x573573f2, 0x0ac48a92, 0xf30d428e, 0x05a6feb9, 0xffac7936, 0xff0f7dbf, 0x0070c7e1, 0xfff27cf3,
+        0x56e94af1, 0x0a318bc1, 0xf333142f, 0x05af2fbf, 0xff9eda6d, 0xff15816a, 0x006fe1b8, 0xfff2524c,
+        0x569acf90, 0x099fdb04, 0xf359929a, 0x05b6ca6b, 0xff916fe1, 0xff1b848e, 0x006ef3df, 0xfff22aea,
+        0x564a0610, 0x090f7d57, 0xf380b8ba, 0x05bdcfb2, 0xff843a32, 0xff218692, 0x006dfe94, 0xfff206bf,
+        0x55f6f2d3, 0x0880779d, 0xf3a88179, 0x05c44095, 0xff7739f7, 0xff2786e1, 0x006d0217, 0xfff1e5bb,
+        0x55a19a5c, 0x07f2ce9b, 0xf3d0e7c2, 0x05ca1e1f, 0xff6a6fc1, 0xff2d84e5, 0x006bfea4, 0xfff1c7d0,
+        0x554a0148, 0x076686fc, 0xf3f9e680, 0x05cf6965, 0xff5ddc1a, 0xff33800e, 0x006af47b, 0xfff1acef,
+        0x54f02c56, 0x06dba551, 0xf42378a0, 0x05d42387, 0xff517f86, 0xff3977cb, 0x0069e3d9, 0xfff19508,
+        0x54942061, 0x06522e0f, 0xf44d9912, 0x05d84daf, 0xff455a80, 0xff3f6b8f, 0x0068ccfa, 0xfff1800b,
+        0x5435e263, 0x05ca258f, 0xf47842c5, 0x05dbe90f, 0xff396d7f, 0xff455acf, 0x0067b01e, 0xfff16de9,
+        0x53d57774, 0x0543900d, 0xf4a370ad, 0x05def6e4, 0xff2db8f2, 0xff4b4503, 0x00668d80, 0xfff15e93,
+        0x5372e4c6, 0x04be71ab, 0xf4cf1dbf, 0x05e17873, 0xff223d40, 0xff5129a3, 0x0065655d, 0xfff151f9,
+        0x530e2fac, 0x043ace6e, 0xf4fb44f4, 0x05e36f0d, 0xff16faca, 0xff57082e, 0x006437f1, 0xfff1480b,
+        0x52a75d90, 0x03b8aa40, 0xf527e149, 0x05e4dc08, 0xff0bf1ed, 0xff5ce021, 0x00630577, 0xfff140b9,
+        0x523e73fd, 0x033808eb, 0xf554edbd, 0x05e5c0c6, 0xff0122fc, 0xff62b0fd, 0x0061ce2c, 0xfff13bf3,
+        0x51d37897, 0x02b8ee22, 0xf5826555, 0x05e61eae, 0xfef68e45, 0xff687a47, 0x00609249, 0xfff139aa,
+        0x5166711c, 0x023b5d76, 0xf5b0431a, 0x05e5f733, 0xfeec340f, 0xff6e3b84, 0x005f520a, 0xfff139cd,
+        0x50f76368, 0x01bf5a5e, 0xf5de8218, 0x05e54bcd, 0xfee2149b, 0xff73f43d, 0x005e0da8, 0xfff13c4c,
+        0x5086556f, 0x0144e834, 0xf60d1d63, 0x05e41dfe, 0xfed83023, 0xff79a3fe, 0x005cc55c, 0xfff14119,
+        0x50134d3e, 0x00cc0a36, 0xf63c1012, 0x05e26f4e, 0xfece86db, 0xff7f4a54, 0x005b7961, 0xfff14821,
+        0x4f9e50ff, 0x0054c382, 0xf66b5544, 0x05e0414d, 0xfec518f1, 0xff84e6d0, 0x005a29ed, 0xfff15156,
+        0x4f2766f2, 0xffdf171b, 0xf69ae81d, 0x05dd9593, 0xfebbe68c, 0xff8a7905, 0x0058d738, 0xfff15ca8,
+        0x4eae9571, 0xff6b07e7, 0xf6cac3c7, 0x05da6dbe, 0xfeb2efcd, 0xff900089, 0x0057817b, 0xfff16a07,
+        0x4e33e2ee, 0xfef898ae, 0xf6fae373, 0x05d6cb72, 0xfeaa34d0, 0xff957cf4, 0x005628ec, 0xfff17962,
+        0x4db755f3, 0xfe87cc1b, 0xf72b425b, 0x05d2b05c, 0xfea1b5a9, 0xff9aede0, 0x0054cdc0, 0xfff18aab,
+        0x4d38f520, 0xfe18a4bc, 0xf75bdbbd, 0x05ce1e2d, 0xfe997268, 0xffa052ec, 0x0053702d, 0xfff19dd1,
+        0x4cb8c72e, 0xfdab2501, 0xf78caae0, 0x05c9169d, 0xfe916b15, 0xffa5abb8, 0x00521068, 0xfff1b2c5,
+        0x4c36d2eb, 0xfd3f4f3d, 0xf7bdab16, 0x05c39b6a, 0xfe899fb2, 0xffaaf7e6, 0x0050aea5, 0xfff1c976,
+        0x4bb31f3c, 0xfcd525a5, 0xf7eed7b4, 0x05bdae57, 0xfe82103f, 0xffb0371c, 0x004f4b17, 0xfff1e1d6,
+        0x4b2db31a, 0xfc6caa53, 0xf8202c1c, 0x05b7512e, 0xfe7abcb1, 0xffb56902, 0x004de5f1, 0xfff1fbd5,
+        0x4aa69594, 0xfc05df40, 0xf851a3b6, 0x05b085bc, 0xfe73a4fb, 0xffba8d44, 0x004c7f66, 0xfff21764,
+        0x4a1dcdce, 0xfba0c64b, 0xf88339f5, 0x05a94dd5, 0xfe6cc909, 0xffbfa38d, 0x004b17a6, 0xfff23473,
+        0x499362ff, 0xfb3d6133, 0xf8b4ea55, 0x05a1ab52, 0xfe6628c1, 0xffc4ab8f, 0x0049aee3, 0xfff252f3,
+        0x49075c72, 0xfadbb19a, 0xf8e6b059, 0x0599a00e, 0xfe5fc405, 0xffc9a4fc, 0x0048454b, 0xfff272d6,
+        0x4879c185, 0xfa7bb908, 0xf9188793, 0x05912dea, 0xfe599aaf, 0xffce8f8a, 0x0046db0f, 0xfff2940b,
+        0x47ea99a9, 0xfa1d78e3, 0xf94a6b9b, 0x058856cd, 0xfe53ac97, 0xffd36af1, 0x0045705c, 0xfff2b686,
+        0x4759ec60, 0xf9c0f276, 0xf97c5815, 0x057f1c9e, 0xfe4df98e, 0xffd836eb, 0x00440561, 0xfff2da36,
+        0x46c7c140, 0xf96626f0, 0xf9ae48af, 0x0575814c, 0xfe48815e, 0xffdcf336, 0x00429a4a, 0xfff2ff0d,
+        0x46341fed, 0xf90d1761, 0xf9e03924, 0x056b86c6, 0xfe4343d0, 0xffe19f91, 0x00412f43, 0xfff324fd,
+        0x459f101d, 0xf8b5c4be, 0xfa122537, 0x05612f00, 0xfe3e40a6, 0xffe63bc0, 0x003fc478, 0xfff34bf9,
+        0x45089996, 0xf8602fdc, 0xfa4408ba, 0x05567bf1, 0xfe39779a, 0xffeac787, 0x003e5a12, 0xfff373f0,
+        0x4470c42d, 0xf80c5977, 0xfa75df87, 0x054b6f92, 0xfe34e867, 0xffef42af, 0x003cf03d, 0xfff39cd7,
+        0x43d797c7, 0xf7ba422b, 0xfaa7a586, 0x05400be1, 0xfe3092bf, 0xfff3ad01, 0x003b871f, 0xfff3c69f,
+        0x433d1c56, 0xf769ea78, 0xfad956ab, 0x053452dc, 0xfe2c7650, 0xfff8064b, 0x003a1ee3, 0xfff3f13a,
+        0x42a159dc, 0xf71b52c4, 0xfb0aeef6, 0x05284685, 0xfe2892c5, 0xfffc4e5c, 0x0038b7ae, 0xfff41c9c,
+        0x42045865, 0xf6ce7b57, 0xfb3c6a73, 0x051be8dd, 0xfe24e7c3, 0x00008507, 0x003751a7, 0xfff448b7,
+        0x4166200e, 0xf683645a, 0xfb6dc53c, 0x050f3bec, 0xfe2174ec, 0x0004aa1f, 0x0035ecf4, 0xfff4757e,
+        0x40c6b8fd, 0xf63a0ddf, 0xfb9efb77, 0x050241b6, 0xfe1e39da, 0x0008bd7c, 0x003489b9, 0xfff4a2e5,
+        0x40262b65, 0xf5f277d9, 0xfbd00956, 0x04f4fc46, 0xfe1b3628, 0x000cbef7, 0x0033281a, 0xfff4d0de,
+        0x3f847f83, 0xf5aca21f, 0xfc00eb1b, 0x04e76da3, 0xfe18696a, 0x0010ae6e, 0x0031c83a, 0xfff4ff5d,
+        0x3ee1bda2, 0xf5688c6d, 0xfc319d13, 0x04d997d8, 0xfe15d32f, 0x00148bbd, 0x00306a3b, 0xfff52e57,
+        0x3e3dee13, 0xf5263665, 0xfc621b9a, 0x04cb7cf2, 0xfe137304, 0x001856c7, 0x002f0e3f, 0xfff55dbf,
+        0x3d991932, 0xf4e59f8a, 0xfc926319, 0x04bd1efb, 0xfe114872, 0x001c0f6e, 0x002db466, 0xfff58d89,
+        0x3cf34766, 0xf4a6c748, 0xfcc27008, 0x04ae8000, 0xfe0f52fc, 0x001fb599, 0x002c5cd0, 0xfff5bdaa,
+        0x3c4c811c, 0xf469aced, 0xfcf23eec, 0x049fa20f, 0xfe0d9224, 0x0023492f, 0x002b079a, 0xfff5ee17,
+        0x3ba4cec9, 0xf42e4faf, 0xfd21cc59, 0x04908733, 0xfe0c0567, 0x0026ca1c, 0x0029b4e4, 0xfff61ec5,
+        0x3afc38eb, 0xf3f4aea6, 0xfd5114f0, 0x0481317a, 0xfe0aac3f, 0x002a384c, 0x002864c9, 0xfff64fa8,
+        0x3a52c805, 0xf3bcc8d3, 0xfd801564, 0x0471a2ef, 0xfe098622, 0x002d93ae, 0x00271766, 0xfff680b5,
+        0x39a884a1, 0xf3869d1a, 0xfdaeca73, 0x0461dda0, 0xfe089283, 0x0030dc34, 0x0025ccd7, 0xfff6b1e4,
+        0x38fd774e, 0xf3522a49, 0xfddd30eb, 0x0451e396, 0xfe07d0d3, 0x003411d2, 0x00248535, 0xfff6e329,
+        0x3851a8a2, 0xf31f6f0f, 0xfe0b45aa, 0x0441b6dd, 0xfe07407d, 0x0037347d, 0x0023409a, 0xfff7147a,
+        0x37a52135, 0xf2ee6a07, 0xfe39059b, 0x0431597d, 0xfe06e0eb, 0x003a442e, 0x0021ff1f, 0xfff745cd,
+        0x36f7e9a4, 0xf2bf19ae, 0xfe666dbc, 0x0420cd80, 0xfe06b184, 0x003d40e0, 0x0020c0dc, 0xfff7771a,
+        0x364a0a90, 0xf2917c6d, 0xfe937b15, 0x041014eb, 0xfe06b1ac, 0x00402a8e, 0x001f85e6, 0xfff7a857,
+        0x359b8c9d, 0xf265908f, 0xfec02ac2, 0x03ff31c3, 0xfe06e0c4, 0x00430137, 0x001e4e56, 0xfff7d97a,
+        0x34ec786f, 0xf23b544b, 0xfeec79ec, 0x03ee260d, 0xfe073e2a, 0x0045c4dd, 0x001d1a3f, 0xfff80a7c,
+        0x343cd6af, 0xf212c5be, 0xff1865cd, 0x03dcf3ca, 0xfe07c93a, 0x00487582, 0x001be9b7, 0xfff83b52,
+        0x338cb004, 0xf1ebe2ec, 0xff43ebac, 0x03cb9cf9, 0xfe08814e, 0x004b132b, 0x001abcd0, 0xfff86bf6,
+        0x32dc0d17, 0xf1c6a9c3, 0xff6f08e4, 0x03ba2398, 0xfe0965bc, 0x004d9dde, 0x0019939d, 0xfff89c60,
+        0x322af693, 0xf1a3181a, 0xff99badb, 0x03a889a1, 0xfe0a75da, 0x005015a5, 0x00186e31, 0xfff8cc86,
+        0x3179751f, 0xf1812bb0, 0xffc3ff0c, 0x0396d10c, 0xfe0bb0f9, 0x00527a8a, 0x00174c9c, 0xfff8fc62,
+        0x30c79163, 0xf160e22d, 0xffedd2fd, 0x0384fbd1, 0xfe0d166b, 0x0054cc9a, 0x00162eef, 0xfff92bec,
+        0x30155404, 0xf1423924, 0x00173447, 0x03730be0, 0xfe0ea57e, 0x00570be4, 0x00151538, 0xfff95b1e,
+        0x2f62c5a7, 0xf1252e0f, 0x00402092, 0x0361032a, 0xfe105d7e, 0x00593877, 0x0013ff88, 0xfff989ef,
+        0x2eafeeed, 0xf109be56, 0x00689598, 0x034ee39b, 0xfe123db6, 0x005b5267, 0x0012edea, 0xfff9b85b,
+        0x2dfcd873, 0xf0efe748, 0x0090911f, 0x033caf1d, 0xfe144570, 0x005d59c6, 0x0011e06d, 0xfff9e65a,
+        0x2d498ad3, 0xf0d7a622, 0x00b81102, 0x032a6796, 0xfe1673f2, 0x005f4eac, 0x0010d71d, 0xfffa13e5,
+        0x2c960ea3, 0xf0c0f808, 0x00df1328, 0x03180ee7, 0xfe18c884, 0x0061312e, 0x000fd205, 0xfffa40f8,
+        0x2be26c73, 0xf0abda0e, 0x0105958c, 0x0305a6f0, 0xfe1b4268, 0x00630167, 0x000ed130, 0xfffa6d8d,
+        0x2b2eaccf, 0xf0984931, 0x012b9635, 0x02f3318a, 0xfe1de0e2, 0x0064bf71, 0x000dd4a7, 0xfffa999d,
+        0x2a7ad83c, 0xf086425a, 0x0151133e, 0x02e0b08d, 0xfe20a335, 0x00666b68, 0x000cdc74, 0xfffac525,
+        0x29c6f738, 0xf075c260, 0x01760ad1, 0x02ce25ca, 0xfe2388a1, 0x0068056b, 0x000be89f, 0xfffaf01e,
+        0x2913123c, 0xf066c606, 0x019a7b27, 0x02bb9310, 0xfe269065, 0x00698d98, 0x000af931, 0xfffb1a84,
+        0x285f31b7, 0xf05949fb, 0x01be628c, 0x02a8fa2a, 0xfe29b9c1, 0x006b0411, 0x000a0e2f, 0xfffb4453,
+        0x27ab5e12, 0xf04d4ade, 0x01e1bf58, 0x02965cdb, 0xfe2d03f2, 0x006c68f8, 0x000927a0, 0xfffb6d86,
+        0x26f79fab, 0xf042c539, 0x02048ff8, 0x0283bce6, 0xfe306e35, 0x006dbc71, 0x00084589, 0xfffb961a,
+        0x2643feda, 0xf039b587, 0x0226d2e6, 0x02711c05, 0xfe33f7c7, 0x006efea0, 0x000767f0, 0xfffbbe09,
+        0x259083eb, 0xf032182f, 0x024886ad, 0x025e7bf0, 0xfe379fe3, 0x00702fae, 0x00068ed8, 0xfffbe552,
+        0x24dd3721, 0xf02be98a, 0x0269a9e9, 0x024bde5a, 0xfe3b65c4, 0x00714fc0, 0x0005ba46, 0xfffc0bef,
+        0x242a20b3, 0xf02725dc, 0x028a3b44, 0x023944ee, 0xfe3f48a5, 0x00725f02, 0x0004ea3a, 0xfffc31df,
+        0x237748cf, 0xf023c95d, 0x02aa397b, 0x0226b156, 0xfe4347c0, 0x00735d9c, 0x00041eb9, 0xfffc571e,
+        0x22c4b795, 0xf021d031, 0x02c9a359, 0x02142533, 0xfe476250, 0x00744bba, 0x000357c2, 0xfffc7ba9,
+        0x2212751a, 0xf0213671, 0x02e877b9, 0x0201a223, 0xfe4b978e, 0x0075298a, 0x00029558, 0xfffc9f7e,
+        0x21608968, 0xf021f823, 0x0306b586, 0x01ef29be, 0xfe4fe6b3, 0x0075f739, 0x0001d779, 0xfffcc29a,
+        0x20aefc79, 0xf0241140, 0x03245bbc, 0x01dcbd96, 0xfe544efb, 0x0076b4f5, 0x00011e26, 0xfffce4fc,
+        0x1ffdd63b, 0xf0277db1, 0x03416966, 0x01ca5f37, 0xfe58cf9d, 0x007762f0, 0x0000695e, 0xfffd06a1,
+        0x1f4d1e8e, 0xf02c3953, 0x035ddd9e, 0x01b81028, 0xfe5d67d4, 0x0078015a, 0xffffb91f, 0xfffd2787,
+        0x1e9cdd43, 0xf0323ff5, 0x0379b790, 0x01a5d1ea, 0xfe6216db, 0x00789065, 0xffff0d66, 0xfffd47ae,
+        0x1ded1a1d, 0xf0398d56, 0x0394f674, 0x0193a5f9, 0xfe66dbeb, 0x00791043, 0xfffe6631, 0xfffd6713,
 };
 
 // we use 15 bits to interpolate between these samples
@@ -96,12 +353,16 @@
         return;
     }
 
-    readResampleCoefficients = (readCoefficientsFn) dlsym(resampleCoeffLib,
-            "readResamplerCoefficients");
-    readResampleFirNumCoeffFn readResampleFirNumCoeff = (readResampleFirNumCoeffFn)
+    readResampleFirNumCoeffFn readResampleFirNumCoeff;
+    readResampleFirLerpIntBitsFn readResampleFirLerpIntBits;
+
+    readResampleCoefficients = (readCoefficientsFn)
+            dlsym(resampleCoeffLib, "readResamplerCoefficients");
+    readResampleFirNumCoeff = (readResampleFirNumCoeffFn)
             dlsym(resampleCoeffLib, "readResampleFirNumCoeff");
-    readResampleFirLerpIntBitsFn readResampleFirLerpIntBits = (readResampleFirLerpIntBitsFn)
+    readResampleFirLerpIntBits = (readResampleFirLerpIntBitsFn)
             dlsym(resampleCoeffLib, "readResampleFirLerpIntBits");
+
     if (!readResampleCoefficients || !readResampleFirNumCoeff || !readResampleFirLerpIntBits) {
         readResampleCoefficients = NULL;
         dlclose(resampleCoeffLib);
@@ -111,15 +372,14 @@
     }
 
     c = &veryHighQualityConstants;
-    // we have 16 coefs samples per zero-crossing
     c->coefsBits = readResampleFirLerpIntBits();
-    ALOGV("coefsBits = %d", c->coefsBits);
     c->cShift = kNumPhaseBits - c->coefsBits;
     c->cMask = ((1<<c->coefsBits)-1) << c->cShift;
     c->pShift = kNumPhaseBits - c->coefsBits - pLerpBits;
     c->pMask = ((1<<pLerpBits)-1) << c->pShift;
     // number of zero-crossing on each side
     c->halfNumCoefs = readResampleFirNumCoeff();
+    ALOGV("coefsBits = %d", c->coefsBits);
     ALOGV("halfNumCoefs = %d", c->halfNumCoefs);
     // note that we "leak" resampleCoeffLib until the process exits
 }
@@ -129,7 +389,7 @@
 static inline
 int32_t mulRL(int left, int32_t in, uint32_t vRL)
 {
-#if defined(__arm__) && !defined(__thumb__)
+#if USE_INLINE_ASSEMBLY
     int32_t out;
     if (left) {
         asm( "smultb %[out], %[in], %[vRL] \n"
@@ -144,18 +404,15 @@
     }
     return out;
 #else
-    if (left) {
-        return int16_t(in>>16) * int16_t(vRL&0xFFFF);
-    } else {
-        return int16_t(in>>16) * int16_t(vRL>>16);
-    }
+    int16_t v = left ? int16_t(vRL) : int16_t(vRL>>16);
+    return int32_t((int64_t(in) * v) >> 16);
 #endif
 }
 
 static inline
 int32_t mulAdd(int16_t in, int32_t v, int32_t a)
 {
-#if defined(__arm__) && !defined(__thumb__)
+#if USE_INLINE_ASSEMBLY
     int32_t out;
     asm( "smlawb %[out], %[v], %[in], %[a] \n"
          : [out]"=r"(out)
@@ -163,16 +420,14 @@
          : );
     return out;
 #else
-    return a + in * (v>>16);
-    // improved precision
-    // return a + in * (v>>16) + ((in * (v & 0xffff)) >> 16);
+    return a + int32_t((int64_t(v) * in) >> 16);
 #endif
 }
 
 static inline
 int32_t mulAddRL(int left, uint32_t inRL, int32_t v, int32_t a)
 {
-#if defined(__arm__) && !defined(__thumb__)
+#if USE_INLINE_ASSEMBLY
     int32_t out;
     if (left) {
         asm( "smlawb %[out], %[v], %[inRL], %[a] \n"
@@ -187,13 +442,8 @@
     }
     return out;
 #else
-    if (left) {
-        return a + (int16_t(inRL&0xFFFF) * (v>>16));
-        //improved precision
-        // return a + (int16_t(inRL&0xFFFF) * (v>>16)) + ((int16_t(inRL&0xFFFF) * (v & 0xffff)) >> 16);
-    } else {
-        return a + (int16_t(inRL>>16) * (v>>16));
-    }
+    int16_t s = left ? int16_t(inRL) : int16_t(inRL>>16);
+    return a + int32_t((int64_t(v) * s) >> 16);
 #endif
 }
 
@@ -202,7 +452,7 @@
 AudioResamplerSinc::AudioResamplerSinc(int bitDepth,
         int inChannelCount, int32_t sampleRate, src_quality quality)
     : AudioResampler(bitDepth, inChannelCount, sampleRate, quality),
-    mState(0)
+    mState(0), mImpulse(0), mRingFull(0), mFirCoefs(0)
 {
     /*
      * Layout of the state buffer for 32 tap:
@@ -220,44 +470,48 @@
      *
      */
 
+    mVolumeSIMD[0] = 0;
+    mVolumeSIMD[1] = 0;
+
     // Load the constants for coefficients
     int ok = pthread_once(&once_control, init_routine);
     if (ok != 0) {
         ALOGE("%s pthread_once failed: %d", __func__, ok);
     }
-    mConstants = (quality == VERY_HIGH_QUALITY) ? &veryHighQualityConstants : &highQualityConstants;
+    mConstants = (quality == VERY_HIGH_QUALITY) ?
+            &veryHighQualityConstants : &highQualityConstants;
 }
 
 
-AudioResamplerSinc::~AudioResamplerSinc()
-{
-    delete[] mState;
+AudioResamplerSinc::~AudioResamplerSinc() {
+    free(mState);
 }
 
 void AudioResamplerSinc::init() {
-    const Constants *c = mConstants;
-
-    const size_t numCoefs = 2*c->halfNumCoefs;
+    const Constants& c(*mConstants);
+    const size_t numCoefs = 2 * c.halfNumCoefs;
     const size_t stateSize = numCoefs * mChannelCount * 2;
-    mState = new int16_t[stateSize];
+    mState = (int16_t*)memalign(32, stateSize*sizeof(int16_t));
     memset(mState, 0, sizeof(int16_t)*stateSize);
-    mImpulse = mState + (c->halfNumCoefs-1)*mChannelCount;
+    mImpulse  = mState   + (c.halfNumCoefs-1)*mChannelCount;
     mRingFull = mImpulse + (numCoefs+1)*mChannelCount;
 }
 
+void AudioResamplerSinc::setVolume(int16_t left, int16_t right) {
+    AudioResampler::setVolume(left, right);
+    mVolumeSIMD[0] = int32_t(left)<<16;
+    mVolumeSIMD[1] = int32_t(right)<<16;
+}
+
 void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider)
 {
-
     // FIXME store current state (up or down sample) and only load the coefs when the state
     // changes. Or load two pointers one for up and one for down in the init function.
     // Not critical now since the read functions are fast, but would be important if read was slow.
     if (mConstants == &veryHighQualityConstants && readResampleCoefficients) {
-        ALOGV("get coefficient from libmm-audio resampler library");
-        mFirCoefs = (mInSampleRate <= mSampleRate) ? readResampleCoefficients(true) :
-                readResampleCoefficients(false);
+        mFirCoefs = readResampleCoefficients( mInSampleRate <= mSampleRate );
     } else {
-        ALOGV("Use default coefficients");
         mFirCoefs = (mInSampleRate <= mSampleRate) ? mFirCoefsUp : mFirCoefsDown;
     }
 
@@ -270,7 +524,6 @@
         resample<2>(out, outFrameCount, provider);
         break;
     }
-
 }
 
 
@@ -278,7 +531,8 @@
 void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
         AudioBufferProvider* provider)
 {
-    const Constants *c = mConstants;
+    const Constants& c(*mConstants);
+    const size_t headOffset = c.halfNumCoefs*CHANNELS;
     int16_t* impulse = mImpulse;
     uint32_t vRL = mVolumeRL;
     size_t inputIndex = mInputIndex;
@@ -313,43 +567,31 @@
                 }
             }
         }
-        int16_t *in = mBuffer.i16;
+        int16_t const * const in = mBuffer.i16;
         const size_t frameCount = mBuffer.frameCount;
 
         // Always read-in the first samples from the input buffer
-        int16_t* head = impulse + c->halfNumCoefs*CHANNELS;
-        head[0] = in[inputIndex*CHANNELS + 0];
-        if (CHANNELS == 2)
-            head[1] = in[inputIndex*CHANNELS + 1];
+        int16_t* head = impulse + headOffset;
+        for (size_t i=0 ; i<CHANNELS ; i++) {
+            head[i] = in[inputIndex*CHANNELS + i];
+        }
 
         // handle boundary case
-        int32_t l, r;
-        while (outputIndex < outputSampleCount) {
-            filterCoefficient<CHANNELS>(l, r, phaseFraction, impulse);
-            out[outputIndex++] += 2 * mulRL(1, l, vRL);
-            out[outputIndex++] += 2 * mulRL(0, r, vRL);
+        while (CC_LIKELY(outputIndex < outputSampleCount)) {
+            filterCoefficient<CHANNELS>(&out[outputIndex], phaseFraction, impulse, vRL);
+            outputIndex += 2;
 
             phaseFraction += phaseIncrement;
-            const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits;
-            if (phaseIndex == 1) {
+            const size_t phaseIndex = phaseFraction >> kNumPhaseBits;
+            for (size_t i=0 ; i<phaseIndex ; i++) {
                 inputIndex++;
-                if (inputIndex >= frameCount)
-                    break;  // need a new buffer
-                read<CHANNELS>(impulse, phaseFraction, in, inputIndex);
-            } else if (phaseIndex == 2) {    // maximum value
-                inputIndex++;
-                if (inputIndex >= frameCount)
-                    break;  // 0 frame available, 2 frames needed
-                // read first frame
-                read<CHANNELS>(impulse, phaseFraction, in, inputIndex);
-                inputIndex++;
-                if (inputIndex >= frameCount)
-                    break;  // 0 frame available, 1 frame needed
-                // read second frame
+                if (inputIndex >= frameCount) {
+                    goto done;  // need a new buffer
+                }
                 read<CHANNELS>(impulse, phaseFraction, in, inputIndex);
             }
         }
-
+done:
         // if done with buffer, save samples
         if (inputIndex >= frameCount) {
             inputIndex -= frameCount;
@@ -375,66 +617,215 @@
         int16_t*& impulse, uint32_t& phaseFraction,
         const int16_t* in, size_t inputIndex)
 {
-    const Constants *c = mConstants;
-    const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits;
     impulse += CHANNELS;
     phaseFraction -= 1LU<<kNumPhaseBits;
-    if (impulse >= mRingFull) {
-        const size_t stateSize = (c->halfNumCoefs*2)*CHANNELS;
+
+    const Constants& c(*mConstants);
+    if (CC_UNLIKELY(impulse >= mRingFull)) {
+        const size_t stateSize = (c.halfNumCoefs*2)*CHANNELS;
         memcpy(mState, mState+stateSize, sizeof(int16_t)*stateSize);
         impulse -= stateSize;
     }
-    int16_t* head = impulse + c->halfNumCoefs*CHANNELS;
-    head[0] = in[inputIndex*CHANNELS + 0];
-    if (CHANNELS == 2)
-        head[1] = in[inputIndex*CHANNELS + 1];
+
+    int16_t* head = impulse + c.halfNumCoefs*CHANNELS;
+    for (size_t i=0 ; i<CHANNELS ; i++) {
+        head[i] = in[inputIndex*CHANNELS + i];
+    }
 }
 
 template<int CHANNELS>
 void AudioResamplerSinc::filterCoefficient(
-        int32_t& l, int32_t& r, uint32_t phase, const int16_t *samples)
+        int32_t* out, uint32_t phase, const int16_t *samples, uint32_t vRL)
 {
-    const Constants *c = mConstants;
+    // NOTE: be very careful when modifying the code here. register
+    // pressure is very high and a small change might cause the compiler
+    // to generate far less efficient code.
+    // Always sanity check the result with objdump or test-resample.
 
     // compute the index of the coefficient on the positive side and
     // negative side
-    uint32_t indexP = (phase & c->cMask) >> c->cShift;
-    uint16_t lerpP = (phase & c->pMask) >> c->pShift;
-    uint32_t indexN = (-phase & c->cMask) >> c->cShift;
-    uint16_t lerpN = (-phase & c->pMask) >> c->pShift;
-    if ((indexP == 0) && (lerpP == 0)) {
-        indexN = c->cMask >> c->cShift;
-        lerpN = c->pMask >> c->pShift;
-    }
+    const Constants& c(*mConstants);
+    const int32_t ONE = c.cMask | c.pMask;
+    uint32_t indexP = ( phase & c.cMask) >> c.cShift;
+    uint32_t lerpP  = ( phase & c.pMask) >> c.pShift;
+    uint32_t indexN = ((ONE-phase) & c.cMask) >> c.cShift;
+    uint32_t lerpN  = ((ONE-phase) & c.pMask) >> c.pShift;
 
-    l = 0;
-    r = 0;
-    const int32_t* coefs = mFirCoefs;
-    const int16_t *sP = samples;
-    const int16_t *sN = samples+CHANNELS;
-    for (unsigned int i=0 ; i < c->halfNumCoefs/4 ; i++) {
-        interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP);
-        interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN);
-        sP -= CHANNELS; sN += CHANNELS; coefs += 1 << c->coefsBits;
-        interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP);
-        interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN);
-        sP -= CHANNELS; sN += CHANNELS; coefs += 1 << c->coefsBits;
-        interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP);
-        interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN);
-        sP -= CHANNELS; sN += CHANNELS; coefs += 1 << c->coefsBits;
-        interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP);
-        interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN);
-        sP -= CHANNELS; sN += CHANNELS; coefs += 1 << c->coefsBits;
+    const size_t offset = c.halfNumCoefs;
+    indexP *= offset;
+    indexN *= offset;
+
+    int32_t const* coefsP = mFirCoefs + indexP;
+    int32_t const* coefsN = mFirCoefs + indexN;
+    int16_t const* sP = samples;
+    int16_t const* sN = samples + CHANNELS;
+
+    size_t count = offset;
+
+    if (!USE_NEON) {
+        int32_t l = 0;
+        int32_t r = 0;
+        for (size_t i=0 ; i<count ; i++) {
+            interpolate<CHANNELS>(l, r, coefsP++, offset, lerpP, sP);
+            sP -= CHANNELS;
+            interpolate<CHANNELS>(l, r, coefsN++, offset, lerpN, sN);
+            sN += CHANNELS;
+        }
+        out[0] += 2 * mulRL(1, l, vRL);
+        out[1] += 2 * mulRL(0, r, vRL);
+    } else if (CHANNELS == 1) {
+        int32_t const* coefsP1 = coefsP + offset;
+        int32_t const* coefsN1 = coefsN + offset;
+        sP -= CHANNELS*3;
+        asm (
+            "vmov.32        d2[0], %[lerpP]          \n"    // load the positive phase
+            "vmov.32        d2[1], %[lerpN]          \n"    // load the negative phase
+            "veor           q0, q0, q0               \n"    // result, initialize to 0
+            "vshl.s32       d2, d2, #16              \n"    // convert to 32 bits
+
+            "1:                                      \n"
+            "vld1.16        { d4}, [%[sP]]           \n"    // load 4 16-bits stereo samples
+            "vld1.32        { q8}, [%[coefsP0]:128]! \n"    // load 4 32-bits coefs
+            "vld1.32        { q9}, [%[coefsP1]:128]! \n"    // load 4 32-bits coefs for interpolation
+            "vld1.16        { d6}, [%[sN]]!          \n"    // load 4 16-bits stereo samples
+            "vld1.32        {q10}, [%[coefsN0]:128]! \n"    // load 4 32-bits coefs
+            "vld1.32        {q11}, [%[coefsN1]:128]! \n"    // load 4 32-bits coefs for interpolation
+
+            "vrev64.16      d4, d4                   \n"    // reverse 2 frames of the positive side
+
+            "vsub.s32        q9,  q9,  q8            \n"    // interpolate (step1) 1st set of coefs
+            "vsub.s32       q11, q11, q10            \n"    // interpolate (step1) 2nd set of coets
+            "vshll.s16      q12,  d4, #15            \n"    // extend samples to 31 bits
+
+            "vqrdmulh.s32    q9,  q9, d2[0]          \n"    // interpolate (step2) 1st set of coefs
+            "vqrdmulh.s32   q11, q11, d2[1]          \n"    // interpolate (step3) 2nd set of coefs
+            "vshll.s16      q14,  d6, #15            \n"    // extend samples to 31 bits
+
+            "vadd.s32        q8,  q8,  q9            \n"    // interpolate (step3) 1st set
+            "vadd.s32       q10, q10, q11            \n"    // interpolate (step4) 2nd set
+            "subs           %[count], %[count], #4   \n"    // update loop counter
+
+            "vqrdmulh.s32   q12, q12, q8             \n"    // multiply samples by interpolated coef
+            "vqrdmulh.s32   q14, q14, q10            \n"    // multiply samples by interpolated coef
+            "sub            %[sP], %[sP], #8         \n"    // move pointer to next set of samples
+
+            "vadd.s32       q0, q0, q12              \n"    // accumulate result
+            "vadd.s32       q0, q0, q14              \n"    // accumulate result
+
+            "bne            1b                       \n"    // loop
+
+            "vld1.s32       {d2}, [%[vLR]]           \n"    // load volumes
+            "vld1.s32       {d3}, %[out]             \n"    // load the output
+            "vpadd.s32      d0, d0, d1               \n"    // add all 4 partial sums
+            "vpadd.s32      d0, d0, d0               \n"    // together
+            "vdup.i32       d0, d0[0]                \n"    // interleave L,R channels
+            "vqrdmulh.s32   d0, d0, d2               \n"    // apply volume
+            "vadd.s32       d3, d3, d0               \n"    // accumulate result
+            "vst1.s32       {d3}, %[out]             \n"    // store result
+
+            : [out]     "=Uv" (out[0]),
+              [count]   "+r" (count),
+              [coefsP0] "+r" (coefsP),
+              [coefsP1] "+r" (coefsP1),
+              [coefsN0] "+r" (coefsN),
+              [coefsN1] "+r" (coefsN1),
+              [sP]      "+r" (sP),
+              [sN]      "+r" (sN)
+            : [lerpP]   "r" (lerpP),
+              [lerpN]   "r" (lerpN),
+              [vLR]     "r" (mVolumeSIMD)
+            : "cc", "memory",
+              "q0", "q1", "q2", "q3",
+              "q8", "q9", "q10", "q11",
+              "q12", "q14"
+        );
+    } else if (CHANNELS == 2) {
+        int32_t const* coefsP1 = coefsP + offset;
+        int32_t const* coefsN1 = coefsN + offset;
+        sP -= CHANNELS*3;
+        asm (
+            "vmov.32        d2[0], %[lerpP]          \n"    // load the positive phase
+            "vmov.32        d2[1], %[lerpN]          \n"    // load the negative phase
+            "veor           q0, q0, q0               \n"    // result, initialize to 0
+            "veor           q4, q4, q4               \n"    // result, initialize to 0
+            "vshl.s32       d2, d2, #16              \n"    // convert to 32 bits
+
+            "1:                                      \n"
+            "vld2.16        {d4,d5}, [%[sP]]         \n"    // load 4 16-bits stereo samples
+            "vld1.32        { q8}, [%[coefsP0]:128]! \n"    // load 4 32-bits coefs
+            "vld1.32        { q9}, [%[coefsP1]:128]! \n"    // load 4 32-bits coefs for interpolation
+            "vld2.16        {d6,d7}, [%[sN]]!        \n"    // load 4 16-bits stereo samples
+            "vld1.32        {q10}, [%[coefsN0]:128]! \n"    // load 4 32-bits coefs
+            "vld1.32        {q11}, [%[coefsN1]:128]! \n"    // load 4 32-bits coefs for interpolation
+
+            "vrev64.16      d4, d4                   \n"    // reverse 2 frames of the positive side
+            "vrev64.16      d5, d5                   \n"    // reverse 2 frames of the positive side
+
+            "vsub.s32        q9,  q9,  q8            \n"    // interpolate (step1) 1st set of coefs
+            "vsub.s32       q11, q11, q10            \n"    // interpolate (step1) 2nd set of coets
+            "vshll.s16      q12,  d4, #15            \n"    // extend samples to 31 bits
+            "vshll.s16      q13,  d5, #15            \n"    // extend samples to 31 bits
+
+            "vqrdmulh.s32    q9,  q9, d2[0]          \n"    // interpolate (step2) 1st set of coefs
+            "vqrdmulh.s32   q11, q11, d2[1]          \n"    // interpolate (step3) 2nd set of coefs
+            "vshll.s16      q14,  d6, #15            \n"    // extend samples to 31 bits
+            "vshll.s16      q15,  d7, #15            \n"    // extend samples to 31 bits
+
+            "vadd.s32        q8,  q8,  q9            \n"    // interpolate (step3) 1st set
+            "vadd.s32       q10, q10, q11            \n"    // interpolate (step4) 2nd set
+            "subs           %[count], %[count], #4   \n"    // update loop counter
+
+            "vqrdmulh.s32   q12, q12, q8             \n"    // multiply samples by interpolated coef
+            "vqrdmulh.s32   q13, q13, q8             \n"    // multiply samples by interpolated coef
+            "vqrdmulh.s32   q14, q14, q10            \n"    // multiply samples by interpolated coef
+            "vqrdmulh.s32   q15, q15, q10            \n"    // multiply samples by interpolated coef
+            "sub            %[sP], %[sP], #16        \n"    // move pointer to next set of samples
+
+            "vadd.s32       q0, q0, q12              \n"    // accumulate result
+            "vadd.s32       q4, q4, q13              \n"    // accumulate result
+            "vadd.s32       q0, q0, q14              \n"    // accumulate result
+            "vadd.s32       q4, q4, q15              \n"    // accumulate result
+
+            "bne            1b                       \n"    // loop
+
+            "vld1.s32       {d2}, [%[vLR]]           \n"    // load volumes
+            "vld1.s32       {d3}, %[out]             \n"    // load the output
+            "vpadd.s32      d0, d0, d1               \n"    // add all 4 partial sums from q0
+            "vpadd.s32      d8, d8, d9               \n"    // add all 4 partial sums from q4
+            "vpadd.s32      d0, d0, d0               \n"    // together
+            "vpadd.s32      d8, d8, d8               \n"    // together
+            "vtrn.s32       d0, d8                   \n"    // interlace L,R channels
+            "vqrdmulh.s32   d0, d0, d2               \n"    // apply volume
+            "vadd.s32       d3, d3, d0               \n"    // accumulate result
+            "vst1.s32       {d3}, %[out]             \n"    // store result
+
+            : [out]     "=Uv" (out[0]),
+              [count]   "+r" (count),
+              [coefsP0] "+r" (coefsP),
+              [coefsP1] "+r" (coefsP1),
+              [coefsN0] "+r" (coefsN),
+              [coefsN1] "+r" (coefsN1),
+              [sP]      "+r" (sP),
+              [sN]      "+r" (sN)
+            : [lerpP]   "r" (lerpP),
+              [lerpN]   "r" (lerpN),
+              [vLR]     "r" (mVolumeSIMD)
+            : "cc", "memory",
+              "q0", "q1", "q2", "q3", "q4",
+              "q8", "q9", "q10", "q11",
+              "q12", "q13", "q14", "q15"
+        );
     }
 }
 
 template<int CHANNELS>
 void AudioResamplerSinc::interpolate(
         int32_t& l, int32_t& r,
-        const int32_t* coefs, int16_t lerp, const int16_t* samples)
+        const int32_t* coefs, size_t offset,
+        int32_t lerp, const int16_t* samples)
 {
     int32_t c0 = coefs[0];
-    int32_t c1 = coefs[1];
+    int32_t c1 = coefs[offset];
     int32_t sinc = mulAdd(lerp, (c1-c0)<<1, c0);
     if (CHANNELS == 2) {
         uint32_t rl = *reinterpret_cast<const uint32_t*>(samples);
diff --git a/services/audioflinger/AudioResamplerSinc.h b/services/audioflinger/AudioResamplerSinc.h
index 25fc025..09c6866 100644
--- a/services/audioflinger/AudioResamplerSinc.h
+++ b/services/audioflinger/AudioResamplerSinc.h
@@ -44,18 +44,21 @@
 private:
     void init();
 
+    virtual void setVolume(int16_t left, int16_t right);
+
     template<int CHANNELS>
     void resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
 
     template<int CHANNELS>
     inline void filterCoefficient(
-            int32_t& l, int32_t& r, uint32_t phase, const int16_t *samples);
+            int32_t* out, uint32_t phase, const int16_t *samples, uint32_t vRL);
 
     template<int CHANNELS>
     inline void interpolate(
             int32_t& l, int32_t& r,
-            const int32_t* coefs, int16_t lerp, const int16_t* samples);
+            const int32_t* coefs, size_t offset,
+            int32_t lerp, const int16_t* samples);
 
     template<int CHANNELS>
     inline void read(int16_t*& impulse, uint32_t& phaseFraction,
@@ -64,6 +67,7 @@
     int16_t *mState;
     int16_t *mImpulse;
     int16_t *mRingFull;
+    int32_t mVolumeSIMD[2];
 
     const int32_t * mFirCoefs;
     static const int32_t mFirCoefsDown[];
@@ -71,17 +75,14 @@
 
     // ----------------------------------------------------------------------------
     static const int32_t RESAMPLE_FIR_NUM_COEF       = 8;
-    static const int32_t RESAMPLE_FIR_LERP_INT_BITS  = 4;
+    static const int32_t RESAMPLE_FIR_LERP_INT_BITS  = 7;
 
     struct Constants {
-        // we have 16 coefs samples per zero-crossing
         int coefsBits;
         int cShift;
         uint32_t cMask;
-
         int pShift;
         uint32_t pMask;
-
         // number of zero-crossing on each side
         unsigned int halfNumCoefs;
     };
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/audio-resampler/AudioResamplerCoefficients.cpp b/services/audioflinger/audio-resampler/AudioResamplerCoefficients.cpp
index ade58a7..af3e40d 100644
--- a/services/audioflinger/audio-resampler/AudioResamplerCoefficients.cpp
+++ b/services/audioflinger/audio-resampler/AudioResamplerCoefficients.cpp
@@ -14,42 +14,41 @@
  * limitations under the License.
  */
 
-#include <dnsampler_filter_coefficients_x128_10112011.h>
-#include <resampler_filter_coefficients_10042011.h>
-#undef LOG_TAG
-#include <utils/Log.h>
-//#include "common_log.h"
 #define LOG_TAG "ResamplerCoefficients"
 #define LOG_NDEBUG 0
 
-const int32_t RESAMPLE_FIR_NUM_COEF       = 16;
-const int32_t RESAMPLE_FIR_LERP_INT_BITS  = 7;
+#include <utils/Log.h>
+
+#include "filter_coefficients.h"
+
+const int32_t RESAMPLE_FIR_NUM_COEF = 16;
+const int32_t RESAMPLE_FIR_LERP_INT_BITS = 7;
 
 using namespace android;
+
 #ifdef __cplusplus
 extern "C" {
 #endif
+
 const int32_t* readResamplerCoefficients(bool upSample) {
 
     ALOGV("readResamplerCoefficients");
-    if(upSample) {
-        return resampler_filter_coefficients_10042011;
+    if (upSample) {
+        return up_sampler_filter_coefficients;
+    } else {
+        return dn_sampler_filter_coefficients;
     }
-    else {
-        return dnsampler_filter_coefficients_x128_10112011;
-   }
 
 }
 
 int32_t readResampleFirNumCoeff() {
-
     return RESAMPLE_FIR_NUM_COEF;
 }
 
 int32_t readResampleFirLerpIntBits() {
-
-   return RESAMPLE_FIR_LERP_INT_BITS;
+    return RESAMPLE_FIR_LERP_INT_BITS;
 }
+
 #ifdef __cplusplus
 }
 #endif
diff --git a/services/audioflinger/audio-resampler/dnsampler_filter_coefficients_x128_10112011.h b/services/audioflinger/audio-resampler/dnsampler_filter_coefficients_x128_10112011.h
deleted file mode 100644
index eb2944c..0000000
--- a/services/audioflinger/audio-resampler/dnsampler_filter_coefficients_x128_10112011.h
+++ /dev/null
@@ -1,2585 +0,0 @@
-
-/*
- * 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.
- */
-
-#include <stdlib.h>
-
-namespace android {
-
-const int32_t dnsampler_filter_coefficients_x128_10112011[] = {
-1849391518,
-1849249650,
-1848824221,
-1848115177,
-1847122891,
-1845847499,
-1844289491,
-1842449129,
-1840327103,
-1837923861,
-1835240203,
-1832276710,
-1829034388,
-1825513999,
-1821716652,
-1817643240,
-1813295074,
-1808673214,
-1803779065,
-1798613825,
-1793179109,
-1787476282,
-1781507048,
-1775272909,
-1768775774,
-1762017295,
-1754999460,
-1747724062,
-1740193297,
-1732409109,
-1724373764,
-1716089338,
-1707558293,
-1698782831,
-1689765473,
-1680508558,
-1671014814,
-1661286715,
-1651327042,
-1641138399,
-1630723762,
-1620085842,
-1609227651,
-1598152036,
-1586862206,
-1575361116,
-1563652006,
-1551737947,
-1539622358,
-1527308388,
-1514799466,
-1502098869,
-1489210218,
-1476136877,
-1462882477,
-1449450498,
-1435844739,
-1422068735,
-1408126274,
-1394021008,
-1379756900,
-1365337657,
-1350767225,
-1336049408,
-1321188297,
-1306187721,
-1291051731,
-1275784259,
-1260389523,
-1244871491,
-1229234343,
-1213482137,
-1197619187,
-1181649550,
-1165577481,
-1149407125,
-1133142876,
-1116788884,
-1100349476,
-1083828868,
-1067231493,
-1050561533,
-1033823337,
-1017021160,
-1000159475,
-983242522,
-966274693,
-949260282,
-932203771,
-915109401,
-897981548,
-880824504,
-863642743,
-846440509,
-829222164,
-811991981,
-794754382,
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-674240023,
-657095935,
-639982476,
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-605863979,
-588867231,
-571917545,
-555018972,
-538175499,
-521391190,
-504669905,
-488015541,
-471431958,
-454923090,
-438492688,
-422144531,
-405882352,
-389709927,
-373630848,
-357648715,
-341767096,
-325989596,
-310319649,
-294760687,
-279316098,
-263989284,
-248783473,
-233701878,
-218747691,
-203924116,
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-104068338,
-90396230,
-76880575,
-63523937,
-50328860,
-37297842,
-24433248,
-11737378,
--787473,
--13139049,
--25315210,
--37313887,
--49133021,
--60770615,
--72224791,
--83493750,
--94575698,
--105468905,
--116171744,
--126682678,
--137000178,
--147122799,
--157049194,
--166778112,
--176308299,
--185638576,
--194767849,
--203695121,
--212419395,
--220939772,
--229255432,
--237365662,
--245269744,
--252967058,
--260457050,
--267739278,
--274813300,
--281678782,
--288335451,
--294783151,
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--307051072,
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--323884189,
--329077326,
--334061875,
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--343406662,
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--355869785,
--359612019,
--363149327,
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--372539768,
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--382245500,
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--390921477,
--391698507,
--392288434,
--392692926,
--392913649,
--392952381,
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--392491020,
--391994654,
--391323748,
--390480249,
--389466214,
--388283683,
--386934801,
--385421695,
--383746599,
--381911725,
--379919384,
--377771869,
--375471577,
--373020872,
--370422215,
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--340793645,
--336856270,
--332799742,
--328626863,
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--315438225,
--310828168,
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--301304533,
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--275861163,
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--259592541,
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--253046,
--241972,
--231199,
--220725,
--210546,
--200659,
--191060,
--181746,
--172713,
--163958,
--155477,
--147266,
--139321,
--131639,
--124215,
--117045,
--110126,
--103453,
--97022,
--90829,
--84870,
--79140,
--73636,
--68352,
--63285,
--58431,
--53784,
--49341,
--45097,
--41048,
--37188,
--33515,
--30023,
--26708,
--23566,
--20592,
--17782,
--15130,
--12634,
--10289,
--8089,
--6031,
--4111,
--2324,
--666,
-868,
-2281,
-3578,
-4763,
-5839,
-6812,
-7685,
-8462,
-9146,
-9743,
-10255,
-10686,
-11041,
-11322,
-11533,
-11679,
-11762,
-11785,
-11753,
-11668,
-11533,
-11352,
-11128,
-10864,
-10563,
-10227,
-9860,
-9464,
-9042,
-8596,
-8129,
-7644,
-7142,
-6626,
-6099,
-5562,
-5017,
-4467,
-3913,
-3357,
-2800,
-2245,
-1694,
-1146,
-605,
-71,
--454,
--970,
--1474,
--1966,
--2446,
--2911,
--3362,
--3797,
--4215,
--4617,
--5001,
--5367,
--5714,
--6043,
--6352,
--6641,
--6910,
--7160,
--7389,
--7598,
--7786,
--7954,
--8102,
--8230,
--8338,
--8426,
--8495,
--8545,
--8575,
--8587,
--8582,
--8558,
--8517,
--8460,
--8386,
--8297,
--8192,
--8073,
--7940,
--7794,
--7635,
--7464,
--7281,
--7088,
--6885,
--6672,
--6450,
--6221,
--5984,
--5740,
--5490,
--5235,
--4975,
--4711,
--4443,
--4173,
--3901,
--3627,
--3352,
--3077,
--2803,
--2529,
--2257,
--1986,
--1719,
--1454,
--1193,
--935,
--683,
--435,
--192,
-45,
-276,
-501,
-719,
-930,
-1134,
-1331,
-1519,
-1700,
-1873,
-2038,
-2194,
-2342,
-2481,
-2611,
-2733,
-2846,
-2950,
-3046,
-3133,
-3211,
-3281,
-3343,
-3396,
-3441,
-3477,
-3506,
-3527,
-3541,
-3547,
-3546,
-3538,
-3523,
-3502,
-3474,
-3441,
-3401,
-3357,
-3307,
-3252,
-3192,
-3128,
-3060,
-2989,
-2913,
-2835,
-2753,
-2669,
-2583,
-2494,
-2403,
-2311,
-2218,
-2124,
-2028,
-1933,
-1837,
-1741,
-1645,
-1550,
-1455,
-1361,
-1268,
-1176,
-1086,
-997,
-910,
-825,
-741,
-660,
-581,
-504,
-429,
-357,
-287,
-220,
-156,
-94,
-35,
--22,
--75,
--126,
--175,
--220,
--263,
--303,
--341,
--375,
--408,
--437,
--464,
--489,
--511,
--531,
--548,
--564,
--577,
--588,
--597,
--604,
--610,
--613,
--615,
--616,
--614,
--612,
--608,
--603,
--597,
--590,
--582,
--573,
--563,
--553,
--542,
--530,
--518,
--506,
--493,
--480,
--466,
--453,
--439,
--425,
--411,
--398,
--384,
--370,
--357,
--344,
--331,
--318,
--305,
--293,
--281,
--269,
--258,
--247,
--237,
--227,
--217,
--208,
--199,
--190,
--182,
--174,
--167,
--160,
--154,
--147,
--142,
--136,
--131,
--126,
--121,
--117,
--113,
--109,
--106,
--102,
--99,
--96,
--93,
--90,
--87,
--85,
--82,
--80,
--78,
--76,
--74,
--72,
--70,
--68,
--66,
--64,
--62,
--60,
--58,
--57,
--55,
--53,
--51,
--50,
--48,
--46,
--45,
--43,
--41,
--40,
--38,
--36,
--35,
--33,
--31,
--30,
--28,
--27,
--25,
--24,
--22,
--21,
--20,
--18,
--17,
--16,
--15,
--13,
--12,
--11,
--10,
--9,
--9,
--8,
--7,
--6,
-};
-}
diff --git a/services/audioflinger/audio-resampler/filter_coefficients.h b/services/audioflinger/audio-resampler/filter_coefficients.h
new file mode 100644
index 0000000..bf70c63
--- /dev/null
+++ b/services/audioflinger/audio-resampler/filter_coefficients.h
@@ -0,0 +1,285 @@
+/*
+ * 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.
+ */
+#include <stdlib.h>
+
+namespace android {
+
+// cmd-line: fir -l 7 -s 48000 -c 23400 -n 16 -b 9.62
+const int32_t up_sampler_filter_coefficients[] __attribute__ ((aligned (32))) = {
+        0x7ccccccd, 0x0323eb7f, 0xfd086246, 0x02b2aa5c, 0xfda45e2c, 0x01fa5183, 0xfe694e12, 0x0137e672, 0xff1c87d3, 0x009ce6d8, 0xff9a68b0, 0x003d150d, 0xffde727a, 0x00106595, 0xfff93679, 0x00021fc5,
+        0x7cc9b757, 0x022ac835, 0xfd7e3a71, 0x026b7da1, 0xfdd2b905, 0x01db7c90, 0xfe7db77c, 0x012aa7bf, 0xff24dc32, 0x0097dfc9, 0xff9d4ae9, 0x003b8742, 0xffdf38e5, 0x00100be5, 0xfff959f5, 0x0002144b,
+        0x7cc0773c, 0x01354bc1, 0xfdf365e8, 0x0224726d, 0xfe011d2e, 0x01bc908b, 0xfe923a2b, 0x011d528d, 0xff2d426f, 0x0092cbc0, 0xffa035cc, 0x0039f42e, 0xffe00236, 0x000fb0d2, 0xfff97dfa, 0x000208b0,
+        0x7cb10d52, 0x0043843f, 0xfe67d5a8, 0x01dd92df, 0xfe2f83c1, 0x019d9230, 0xfea6d2e5, 0x010fe901, 0xff35b924, 0x008dab9d, 0xffa328d4, 0x00385c1d, 0xffe0ce46, 0x000f5471, 0xfff9a27f, 0x0001fcf5,
+        0x7c9b7afd, 0xff557f58, 0xfedb7ae9, 0x0196e8fe, 0xfe5de5e3, 0x017e8635, 0xfebb7e75, 0x01026d40, 0xff3e3eed, 0x0088803e, 0xffa6237a, 0x0036bf58, 0xffe19cec, 0x000ef6d4, 0xfff9c77d, 0x0001f11e,
+        0x7c7fc22f, 0xfe6b4a44, 0xff4e471d, 0x01507eb8, 0xfe8c3cc3, 0x015f714d, 0xfed039a8, 0x00f4e16f, 0xff46d266, 0x00834a83, 0xffa9253b, 0x00351e2d, 0xffe26e01, 0x000e980f, 0xfff9eceb, 0x0001e52e,
+        0x7c5de56a, 0xfd84f1c8, 0xffc02bf2, 0x010a5de2, 0xfeba819d, 0x01405821, 0xfee5014c, 0x00e747b0, 0xff4f722b, 0x007e0b4b, 0xffac2d8f, 0x003378e7, 0xffe3415d, 0x000e3834, 0xfffa12c0, 0x0001d927,
+        0x7c35e7bb, 0xfca28234, 0x00311b54, 0x00c49034, 0xfee8adba, 0x01213f58, 0xfef9d232, 0x00d9a226, 0xff581cd8, 0x0078c375, 0xffaf3bf2, 0x0031cfd1, 0xffe416d8, 0x000dd758, 0xfffa38f5, 0x0001cd0d,
+        0x7c07ccbe, 0xfbc40766, 0x00a1076e, 0x007f1f4b, 0xff16ba71, 0x01022b90, 0xff0ea931, 0x00cbf2f0, 0xff60d10b, 0x007373de, 0xffb24fde, 0x00302337, 0xffe4ee4b, 0x000d758d, 0xfffa5f81, 0x0001c0e1,
+        0x7bd3989d, 0xfae98cc5, 0x010fe2ab, 0x003a14a6, 0xff44a128, 0x00e3215e, 0xff238322, 0x00be3c2d, 0xff698d62, 0x006e1d66, 0xffb568ce, 0x002e7363, 0xffe5c78d, 0x000d12e6, 0xfffa865d, 0x0001b4a8,
+        0x7b99500c, 0xfa131d41, 0x017d9fb8, 0xfff579a3, 0xff725b54, 0x00c42551, 0xff385ce3, 0x00b07ff8, 0xff72507e, 0x0068c0e9, 0xffb8863e, 0x002cc0a2, 0xffe6a277, 0x000caf76, 0xfffaad81, 0x0001a863,
+        0x7b58f84d, 0xf940c355, 0x01ea3184, 0xffb15783, 0xff9fe27d, 0x00a53bed, 0xff4d3358, 0x00a2c06b, 0xff7b18fe, 0x00635f45, 0xffbba7aa, 0x002b0b3d, 0xffe77ee2, 0x000c4b50, 0xfffad4e4, 0x00019c15,
+        0x7b12972d, 0xf8728902, 0x02558b43, 0xff6db764, 0xffcd303b, 0x008669ae, 0xff620368, 0x0094ff9b, 0xff83e586, 0x005df954, 0xffbecc8d, 0x00295380, 0xffe85ca7, 0x000be687, 0xfffafc7f, 0x00018fc1,
+        0x7ac63304, 0xf7a877d4, 0x02bfa06d, 0xff2aa243, 0xfffa3e37, 0x0067b303, 0xff76ca02, 0x00873f9b, 0xff8cb4bb, 0x00588ff1, 0xffc1f465, 0x002799b3, 0xffe93b9e, 0x000b812d, 0xfffb244a, 0x0001836a,
+        0x7a73d2b5, 0xf6e298db, 0x032864c1, 0xfee820f8, 0x00270631, 0x00491c54, 0xff8b841a, 0x0079827a, 0xff958542, 0x005323f7, 0xffc51eaf, 0x0025de22, 0xffea1ba2, 0x000b1b55, 0xfffb4c3e, 0x00017712,
+        0x7a1b7daa, 0xf620f4b2, 0x038fcc44, 0xfea63c38, 0x005381fa, 0x002aa9fa, 0xffa02eac, 0x006bca44, 0xff9e55c6, 0x004db63c, 0xffc84ae9, 0x00242115, 0xffeafc8b, 0x000ab510, 0xfffb7452, 0x00016abb,
+        0x79bd3bd8, 0xf5639376, 0x03f5cb46, 0xfe64fc93, 0x007fab77, 0x000c6043, 0xffb4c6b9, 0x005e1900, 0xffa724f0, 0x00484799, 0xffcb7893, 0x002262d6, 0xffebde33, 0x000a4e72, 0xfffb9c80, 0x00015e68,
+        0x795915bc, 0xf4aa7cce, 0x045a565c, 0xfe246a72, 0x00ab7ca6, 0xffee4372, 0xffc9494b, 0x005070b0, 0xffaff16f, 0x0042d8e1, 0xffcea72c, 0x0020a3ad, 0xffecc075, 0x0009e78c, 0xfffbc4bf, 0x0001521b,
+        0x78ef1457, 0xf3f5b7e4, 0x04bd6269, 0xfde48e17, 0x00d6ef99, 0xffd057bb, 0xffddb374, 0x0042d353, 0xffb8b9f3, 0x003d6aea, 0xffd1d635, 0x001ee3e1, 0xffeda32a, 0x00098070, 0xfffbed0a, 0x000145d7,
+        0x787f4134, 0xf3454b6a, 0x051ee498, 0xfda56f9c, 0x0101fe7a, 0xffb2a145, 0xfff2024e, 0x003542e2, 0xffc17d30, 0x0037fe85, 0xffd50530, 0x001d23b9, 0xffee862e, 0x0009192f, 0xfffc1558, 0x0001399e,
+        0x7809a65e, 0xf2993d95, 0x057ed264, 0xfd6716f2, 0x012ca389, 0xff952429, 0x000632fa, 0x0027c151, 0xffca39dd, 0x00329483, 0xffd833a0, 0x001b637e, 0xffef695c, 0x0008b1db, 0xfffc3da2, 0x00012d72,
+        0x778e4e68, 0xf1f19421, 0x05dd218f, 0xfd298be0, 0x0156d920, 0xff77e470, 0x001a42a4, 0x001a508e, 0xffd2eeb3, 0x002d2db0, 0xffdb6109, 0x0019a373, 0xfff04c8f, 0x00084a86, 0xfffc65e2, 0x00012155,
+        0x770d4466, 0xf14e544f, 0x0639c82d, 0xfcecd602, 0x018099b2, 0xff5ae614, 0x002e2e82, 0x000cf281, 0xffdb9a70, 0x0027cada, 0xffde8cf1, 0x0017e3df, 0xfff12fa3, 0x0007e33f, 0xfffc8e11, 0x0001154a,
+        0x768693ec, 0xf0af82e4, 0x0694bca0, 0xfcb0fcca, 0x01a9dfcc, 0xff3e2d01, 0x0041f3d2, 0xffffa90e, 0xffe43bd5, 0x00226ccb, 0xffe1b6dd, 0x00162507, 0xfff21275, 0x00077c17, 0xfffcb628, 0x00010952,
+        0x75fa4911, 0xf015242b, 0x06edf595, 0xfc76077b, 0x01d2a615, 0xff21bd11, 0x00558fdc, 0xfff27611, 0xffecd1a6, 0x001d144a, 0xffe4de56, 0x0014672d, 0xfff2f4e0, 0x00071520, 0xfffcde20, 0x0000fd6f,
+        0x75687068, 0xef7f3bf5, 0x07456a0e, 0xfc3bfd2e, 0x01fae74e, 0xff059a0e, 0x0068fff3, 0xffe55b60, 0xfff55aae, 0x0017c21c, 0xffe802e6, 0x0012aa95, 0xfff3d6c3, 0x0006ae6a, 0xfffd05f3, 0x0000f1a4,
+        0x74d11703, 0xeeedcd98, 0x079b1158, 0xfc02e4cc, 0x02229e57, 0xfee9c7af, 0x007c4177, 0xffd85ac9, 0xfffdd5b8, 0x00127704, 0xffeb2416, 0x0010ef82, 0xfff4b7fb, 0x00064804, 0xfffd2d9b, 0x0000e5f3,
+        0x74344a70, 0xee60dbee, 0x07eee314, 0xfbcac510, 0x0249c629, 0xfece499d, 0x008f51cf, 0xffcb7615, 0x00064197, 0x000d33c3, 0xffee4174, 0x000f3633, 0xfff59866, 0x0005e1fe, 0xfffd5511, 0x0000da5c,
+        0x739218b8, 0xedd86958, 0x0840d732, 0xfb93a486, 0x027059da, 0xfeb3236b, 0x00a22e71, 0xffbeaf06, 0x000e9d1f, 0x0007f915, 0xfff15a8d, 0x000d7eea, 0xfff677e2, 0x00057c68, 0xfffd7c4f, 0x0000cee3,
+        0x72ea905a, 0xed5477be, 0x0890e5f7, 0xfb5d898c, 0x029654a0, 0xfe98589b, 0x00b4d4dd, 0xffb20754, 0x0016e72c, 0x0002c7b6, 0xfff46ef1, 0x000bc9e6, 0xfff75650, 0x00051750, 0xfffda350, 0x0000c388,
+        0x723dc051, 0xecd5088e, 0x08df07f6, 0xfb287a4d, 0x02bbb1cc, 0xfe7dec9c, 0x00c7429f, 0xffa580b1, 0x001f1e9b, 0xfffda05c, 0xfff77e31, 0x000a1765, 0xfff8338e, 0x0004b2c7, 0xfffdca0d, 0x0000b84d,
+        0x718bb80b, 0xec5a1cbc, 0x092b3617, 0xfaf47cc4, 0x02e06ccf, 0xfe63e2cc, 0x00d97550, 0xff991cc9, 0x00274253, 0xfff883be, 0xfffa87df, 0x000867a5, 0xfff90f7c, 0x00044eda, 0xfffdf080, 0x0000ad34,
+        0x70d4876b, 0xebe3b4c5, 0x09756994, 0xfac196bb, 0x03048139, 0xfe4a3e70, 0x00eb6a95, 0xff8cdd3c, 0x002f513a, 0xfff3728d, 0xfffd8b92, 0x0006bae1, 0xfff9e9fd, 0x0003eb98, 0xfffe16a6, 0x0000a23f,
+        0x70183ec5, 0xeb71d0ab, 0x09bd9bfb, 0xfa8fcdca, 0x0327eab8, 0xfe3102bd, 0x00fd2022, 0xff80c3a4, 0x00374a40, 0xffee6d78, 0x000088df, 0x00051157, 0xfffac2f0, 0x0003890e, 0xfffe3c76, 0x0000976e,
+        0x6f56eee1, 0xeb046ffc, 0x0a03c72b, 0xfa5f2755, 0x034aa51b, 0xfe1832d4, 0x010e93b5, 0xff74d194, 0x003f2c57, 0xffe97529, 0x00037f60, 0x00036b3f, 0xfffb9a38, 0x0003274c, 0xfffe61ee, 0x00008cc4,
+        0x6e90a8f2, 0xea9b91cc, 0x0a47e559, 0xfa2fa890, 0x036cac52, 0xfdffd1bd, 0x011fc31c, 0xff690894, 0x0046f679, 0xffe48a4a, 0x00066eae, 0x0001c8d2, 0xfffc6fb8, 0x0002c65d, 0xfffe8707, 0x00008241,
+        0x6dc57e9b, 0xea3734bb, 0x0a89f10c, 0xfa015679, 0x038dfc6c, 0xfde7e26f, 0x0130ac31, 0xff5d6a24, 0x004ea7a3, 0xffdfad7f, 0x00095666, 0x00002a4a, 0xfffd4352, 0x00026650, 0xfffeabbd, 0x000077e8,
+        0x6cf581e8, 0xe9d756f3, 0x0ac9e521, 0xf9d435dc, 0x03ae919a, 0xfdd067ca, 0x01414cdd, 0xff51f7bb, 0x00563edb, 0xffdadf69, 0x000c3627, 0xfffe8fdc, 0xfffe14eb, 0x00020730, 0xfffed00a, 0x00006db9,
+        0x6c20c550, 0xe97bf627, 0x0b07bcc6, 0xf9a84b50, 0x03ce682d, 0xfdb96498, 0x0151a317, 0xff46b2c7, 0x005dbb29, 0xffd620a6, 0x000f0d91, 0xfffcf9be, 0xfffee466, 0x0001a90b, 0xfffef3ea, 0x000063b5,
+        0x6b475bb0, 0xe9250f99, 0x0b437380, 0xf97d9b37, 0x03ed7c9a, 0xfda2db8c, 0x0161ace5, 0xff3b9cad, 0x00651b9c, 0xffd171d1, 0x0011dc47, 0xfffb6825, 0xffffb1aa, 0x00014bed, 0xffff1759, 0x000059dd,
+        0x6a69584a, 0xe8d2a017, 0x0b7d0525, 0xf95429c0, 0x040bcb77, 0xfd8ccf46, 0x01716859, 0xff30b6c8, 0x006c5f4b, 0xffccd380, 0x0014a1ee, 0xfff9db44, 0x00007c9c, 0x0000efe1, 0xffff3a53, 0x00005033,
+        0x6986cec4, 0xe884a3fb, 0x0bb46de2, 0xf92bfae4, 0x0429517b, 0xfd77424c, 0x0180d397, 0xff260269, 0x00738551, 0xffc84645, 0x00175e2d, 0xfff8534d, 0x00014521, 0x000094f3, 0xffff5cd2, 0x000046b8,
+        0x689fd324, 0xe83b1731, 0x0be9aa34, 0xf9051266, 0x04460b81, 0xfd62370e, 0x018fecd1, 0xff1b80da, 0x007a8cd0, 0xffc3cab1, 0x001a10ad, 0xfff6d070, 0x00020b23, 0x00003b2e, 0xffff7ed3, 0x00003d6c,
+        0x67b479cf, 0xe7f5f531, 0x0c1cb6ef, 0xf8df73d6, 0x0461f688, 0xfd4dafe6, 0x019eb246, 0xff113358, 0x008174ef, 0xffbf614e, 0x001cb91a, 0xfff552de, 0x0002ce87, 0xffffe29d, 0xffffa052, 0x00003450,
+        0x66c4d787, 0xe7b53908, 0x0c4d913a, 0xf8bb228c, 0x047d0fb1, 0xfd39af17, 0x01ad2249, 0xff071b16, 0x00883cdc, 0xffbb0aa3, 0x001f5723, 0xfff3dac3, 0x00038f37, 0xffff8b4b, 0xffffc14b, 0x00002b66,
+        0x65d10168, 0xe778dd50, 0x0c7c368d, 0xf89821ac, 0x0497543f, 0xfd2636ca, 0x01bb3b37, 0xfefd3941, 0x008ee3cd, 0xffb6c735, 0x0021ea76, 0xfff2684e, 0x00044d1b, 0xffff3540, 0xffffe1bc, 0x000022ad,
+        0x64d90ce7, 0xe740dc3c, 0x0ca8a4b7, 0xf8767422, 0x04b0c19a, 0xfd134913, 0x01c8fb81, 0xfef38ef6, 0x009568fc, 0xffb29782, 0x002472c8, 0xfff0fba9, 0x0005081f, 0xfffee088, 0x0000019f, 0x00001a28,
+        0x63dd0fcd, 0xe70d2f8d, 0x0cd2d9d5, 0xf8561ca7, 0x04c9554e, 0xfd00e7ec, 0x01d661a6, 0xfeea1d4c, 0x009bcbab, 0xffae7c06, 0x0026efcc, 0xffef94fe, 0x0005c02c, 0xfffe8d2c, 0x000020f3, 0x000011d5,
+        0x62dd2039, 0xe6ddd09f, 0x0cfad45a, 0xf8371dbb, 0x04e10d0a, 0xfcef153a, 0x01e36c34, 0xfee0e54e, 0x00a20b23, 0xffaa7538, 0x0029613a, 0xffee3477, 0x0006752d, 0xfffe3b35, 0x00003fb3, 0x000009b6,
+        0x61d95497, 0xe6b2b862, 0x0d209309, 0xf81979ab, 0x04f7e6a2, 0xfcddd2c7, 0x01f019cb, 0xfed7e7fd, 0x00a826b2, 0xffa6838c, 0x002bc6cd, 0xffecda3b, 0x0007270f, 0xfffdeaaa, 0x00005ddd, 0x000001cc,
+        0x60d1c3a6, 0xe68bdf5e, 0x0d4414f9, 0xf7fd328c, 0x050de00d, 0xfccd2246, 0x01fc691b, 0xfecf2650, 0x00ae1dae, 0xffa2a770, 0x002e2040, 0xffeb866f, 0x0007d5bf, 0xfffd9b96, 0x00007b6f, 0xfffffa17,
+        0x5fc68470, 0xe6693db5, 0x0d65598f, 0xf7e24a3c, 0x0522f766, 0xfcbd0551, 0x020858e2, 0xfec6a130, 0x00b3ef73, 0xff9ee150, 0x00306d52, 0xffea3939, 0x0008812a, 0xfffd4dff, 0x00009865, 0xfffff297,
+        0x5eb7ae46, 0xe64acb24, 0x0d846084, 0xf7c8c267, 0x05372aee, 0xfcad7d6b, 0x0213e7f0, 0xfebe5980, 0x00b99b65, 0xff9b3192, 0x0032adc4, 0xffe8f2bb, 0x0009293e, 0xfffd01ee, 0x0000b4bd, 0xffffeb4c,
+        0x5da558c5, 0xe6307f05, 0x0da129df, 0xf7b09c7f, 0x054a7909, 0xfc9e8bfd, 0x021f1526, 0xfeb65015, 0x00bf20ee, 0xff979898, 0x0034e15b, 0xffe7b317, 0x0009cdeb, 0xfffcb769, 0x0000d074, 0xffffe438,
+        0x5c8f9bcb, 0xe61a504f, 0x0dbbb5f6, 0xf799d9c4, 0x055ce03f, 0xfc903258, 0x0229df75, 0xfeae85bb, 0x00c47f7f, 0xff9416c1, 0x003707dc, 0xffe67a6f, 0x000a6f20, 0xfffc6e78, 0x0000eb89, 0xffffdd5a,
+        0x5b768f7a, 0xe6083599, 0x0dd40571, 0xf7847b3d, 0x056e5f3d, 0xfc8271b4, 0x023445dd, 0xfea6fb32, 0x00c9b691, 0xff90ac66, 0x00392111, 0xffe548e0, 0x000b0cce, 0xfffc2720, 0x000105f9, 0xffffd6b2,
+        0x5a5a4c32, 0xe5fa2519, 0x0dea1943, 0xf77081be, 0x057ef4d3, 0xfc754b32, 0x023e4772, 0xfe9fb12e, 0x00cec5a1, 0xff8d59dd, 0x003b2cc5, 0xffe41e88, 0x000ba6e5, 0xfffbe169, 0x00011fc3, 0xffffd041,
+        0x593aea93, 0xe5f014aa, 0x0dfdf2ae, 0xf75dede5, 0x058e9ff8, 0xfc68bfd7, 0x0247e354, 0xfe98a85b, 0x00d3ac38, 0xff8a1f77, 0x003d2ac6, 0xffe2fb83, 0x000c3d59, 0xfffb9d59, 0x000138e4, 0xffffca06,
+        0x58188376, 0xe5e9f9ca, 0x0e0f9342, 0xf74cc01c, 0x059d5fc5, 0xfc5cd092, 0x025118b8, 0xfe91e159, 0x00d869e1, 0xff86fd81, 0x003f1ae4, 0xffe1dfec, 0x000cd01b, 0xfffb5af3, 0x0001515c, 0xffffc402,
+        0x56f32fea, 0xe5e7c99e, 0x0e1efcdb, 0xf73cf898, 0x05ab3377, 0xfc517e38, 0x0259e6e1, 0xfe8b5cba, 0x00dcfe32, 0xff83f443, 0x0040fcf3, 0xffe0cbdc, 0x000d5f1f, 0xfffb1a3f, 0x00016928, 0xffffbe35,
+        0x55cb0935, 0xe5e978f0, 0x0e2c319d, 0xf72e9758, 0x05b81a70, 0xfc46c987, 0x02624d23, 0xfe851b09, 0x00e168c5, 0xff810401, 0x0042d0c9, 0xffdfbf6b, 0x000dea5a, 0xfffadb40, 0x00018048, 0xffffb89f,
+        0x54a028d0, 0xe5eefc35, 0x0e3733fc, 0xf7219c2a, 0x05c41435, 0xfc3cb323, 0x026a4ae5, 0xfe7f1cc4, 0x00e5a93c, 0xff7e2cfb, 0x0044963d, 0xffdebaaf, 0x000e71c1, 0xfffa9dfa, 0x000196ba, 0xffffb340,
+        0x5372a862, 0xe5f8478d, 0x0e4006b2, 0xf71606a6, 0x05cf2070, 0xfc333b97, 0x0271df9c, 0xfe79625e, 0x00e9bf43, 0xff7b6f6c, 0x00464d2b, 0xffddbdbd, 0x000ef549, 0xfffa6273, 0x0001ac7d, 0xffffae17,
+        0x5242a1c1, 0xe6054ec6, 0x0e46acc4, 0xf70bd632, 0x05d93eee, 0xfc2a6356, 0x02790ace, 0xfe73ec40, 0x00edaa88, 0xff78cb8c, 0x0047f571, 0xffdcc8a9, 0x000f74e9, 0xfffa28ad, 0x0001c191, 0xffffa924,
+        0x51102eec, 0xe616055a, 0x0e4b297c, 0xf7030a01, 0x05e26f9f, 0xfc222abb, 0x027fcc12, 0xfe6ebac6, 0x00f16ac4, 0xff76418b, 0x00498eed, 0xffdbdb84, 0x000ff098, 0xfff9f0ac, 0x0001d5f4, 0xffffa467,
+        0x4fdb6a09, 0xe62a5e76, 0x0e4d806f, 0xf6fba113, 0x05eab296, 0xfc1a9208, 0x02862311, 0xfe69ce43, 0x00f4ffb6, 0xff73d199, 0x004b1984, 0xffdaf65e, 0x0010684e, 0xfff9ba73, 0x0001e9a7, 0xffff9fe0,
+        0x4ea46d66, 0xe6424cf8, 0x0e4db575, 0xf6f59a36, 0x05f20809, 0xfc139968, 0x028c0f83, 0xfe6526fe, 0x00f86924, 0xff717bdf, 0x004c951b, 0xffda1948, 0x0010dc05, 0xfff98604, 0x0001fca8, 0xffff9b8f,
+        0x4d6b536f, 0xe65dc373, 0x0e4bccac, 0xf6f0f407, 0x05f87053, 0xfc0d40ec, 0x0291912f, 0xfe60c533, 0x00fba6da, 0xff6f4083, 0x004e0199, 0xffd9444e, 0x00114bb4, 0xfff95363, 0x00020ef7, 0xffff9773,
+        0x4c3036b2, 0xe67cb42f, 0x0e47ca78, 0xf6edacf2, 0x05fdebee, 0xfc07888e, 0x0296a7f0, 0xfe5ca913, 0x00feb8ad, 0xff6d1fa5, 0x004f5ee9, 0xffd8777d, 0x0011b757, 0xfff92290, 0x00022095, 0xffff938c,
+        0x4af331d9, 0xe69f112f, 0x0e41b37c, 0xf6ebc332, 0x06027b78, 0xfc027031, 0x029b53af, 0xfe58d2c5, 0x01019e78, 0xff6b1961, 0x0050acf7, 0xffd7b2e0, 0x00121ee9, 0xfff8f38e, 0x00023181, 0xffff8fd9,
+        0x49b45fa8, 0xe6c4cc2e, 0x0e398c9f, 0xf6eb34d4, 0x06061fb2, 0xfbfdf79e, 0x029f9466, 0xfe554265, 0x0104581c, 0xff692dd2, 0x0051ebb4, 0xffd6f67f, 0x00128265, 0xfff8c65d, 0x000241bb, 0xffff8c5a,
+        0x4873daf7, 0xe6edd6a4, 0x0e2f5b0b, 0xf6ebffb2, 0x0608d97c, 0xfbfa1e88, 0x02a36a1e, 0xfe51f802, 0x0106e583, 0xff675d09, 0x00531b12, 0xffd64264, 0x0012e1c8, 0xfff89b00, 0x00025143, 0xffff890e,
+        0x4731beb7, 0xe71a21c7, 0x0e232425, 0xf6ee217b, 0x060aa9da, 0xfbf6e48c, 0x02a6d4f0, 0xfe4ef3a4, 0x0109469d, 0xff65a718, 0x00543b04, 0xffd59695, 0x00133d0e, 0xfff87176, 0x0002601b, 0xffff85f5,
+        0x45ee25e7, 0xe7499e8f, 0x0e14ed93, 0xf6f197ad, 0x060b91ee, 0xfbf4492d, 0x02a9d508, 0xfe4c3546, 0x010b7b61, 0xff640c08, 0x00554b83, 0xffd4f316, 0x00139436, 0xfff849c0, 0x00026e41, 0xffff830e,
+        0x44a92b96, 0xe77c3db4, 0x0e04bd39, 0xf6f65f9b, 0x060b92ff, 0xfbf24bd9, 0x02ac6a9e, 0xfe49bcd9, 0x010d83cb, 0xff628be3, 0x00564c88, 0xffd457ec, 0x0013e73e, 0xfff823dd, 0x00027bb8, 0xffff805a,
+        0x4362eadc, 0xe7b1efb4, 0x0df29936, 0xf6fc766a, 0x060aae6e, 0xfbf0ebe7, 0x02ae95fb, 0xfe478a42, 0x010f5fe2, 0xff6126a9, 0x00573e0f, 0xffd3c519, 0x00143626, 0xfff7ffce, 0x0002887f, 0xffff7dd6,
+        0x421b7edf, 0xe7eaa4d4, 0x0dde87e2, 0xf703d912, 0x0608e5c2, 0xfbf02896, 0x02b05779, 0xfe459d5e, 0x01110faf, 0xff5fdc5b, 0x00582016, 0xffd33a9e, 0x001480ec, 0xfff7dd92, 0x00029497, 0xffff7b82,
+        0x40d302c5, 0xe8264d21, 0x0dc88fd2, 0xf70c8461, 0x06063a9d, 0xfbf00112, 0x02b1af7f, 0xfe43f5ff, 0x01129344, 0xff5eacf3, 0x0058f29f, 0xffd2b87c, 0x0014c792, 0xfff7bd28, 0x0002a002, 0xffff795f,
+        0x3f8991bd, 0xe864d874, 0x0db0b7d1, 0xf71674fa, 0x0602aec3, 0xfbf0746e, 0x02b29e84, 0xfe4293ec, 0x0113eabb, 0xff5d9867, 0x0059b5ad, 0xffd23eaf, 0x00150a19, 0xfff79e8f, 0x0002aac0, 0xffff776a,
+        0x3e3f46f2, 0xe8a63671, 0x0d9706e1, 0xf721a756, 0x05fe4414, 0xfbf181a9, 0x02b3250f, 0xfe4176e2, 0x01151632, 0xff5c9eaa, 0x005a6946, 0xffd1cd37, 0x00154883, 0xfff781c5, 0x0002b4d2, 0xffff75a3,
+        0x3cf43d8f, 0xe8ea568f, 0x0d7b843b, 0xf72e17c4, 0x05f8fc8f, 0xfbf327ab, 0x02b343b5, 0xfe409e95, 0x011615ce, 0xff5bbfaa, 0x005b0d72, 0xffd1640e, 0x001582d3, 0xfff766c8, 0x0002be3b, 0xffff740a,
+        0x3ba890b9, 0xe9312813, 0x0d5e3749, 0xf73bc26b, 0x05f2da52, 0xfbf56549, 0x02b2fb1a, 0xfe400aae, 0x0116e9bc, 0xff5afb53, 0x005ba23b, 0xffd1032f, 0x0015b90b, 0xfff74d97, 0x0002c6fa, 0xffff729e,
+        0x3a5c5b8e, 0xe97a9a17, 0x0d3f27ab, 0xf74aa34c, 0x05ebdf97, 0xfbf83941, 0x02b24bf1, 0xfe3fbacd, 0x0117922f, 0xff5a5189, 0x005c27af, 0xffd0aa93, 0x0015eb2f, 0xfff7362f, 0x0002cf12, 0xffff715d,
+        0x390fb920, 0xe9c69b8c, 0x0d1e5d32, 0xf75ab63f, 0x05e40eb3, 0xfbfba23f, 0x02b136f9, 0xfe3fae87, 0x01180f5d, 0xff59c230, 0x005c9ddc, 0xffd05a33, 0x00161944, 0xfff7208d, 0x0002d684, 0xffff7047,
+        0x37c2c474, 0xea151b3a, 0x0cfbdfdd, 0xf76bf6f7, 0x05db6a19, 0xfbff9ed7, 0x02afbd02, 0xfe3fe569, 0x01186187, 0xff594d27, 0x005d04d4, 0xffd01205, 0x0016434f, 0xfff70caf, 0x0002dd53, 0xffff6f5c,
+        0x36759880, 0xea6607c4, 0x0cd7b7dd, 0xf77e6103, 0x05d1f459, 0xfc042d8e, 0x02addee8, 0xfe405ef6, 0x011888f2, 0xff58f249, 0x005d5cab, 0xffcfd1ff, 0x00166956, 0xfff6fa92, 0x0002e37e, 0xffff6e99,
+        0x35285026, 0xeab94fa9, 0x0cb1ed8c, 0xf791efcb, 0x05c7b01a, 0xfc094cd2, 0x02ab9d96, 0xfe411aa8, 0x011885e7, 0xff58b16c, 0x005da575, 0xffcf9a15, 0x00168b5e, 0xfff6ea31, 0x0002e90a, 0xffff6dff,
+        0x33db0631, 0xeb0ee148, 0x0c8a8973, 0xf7a69e96, 0x05bca021, 0xfc0efafe, 0x02a8fa03, 0xfe4217ef, 0x011858b9, 0xff588a65, 0x005ddf4c, 0xffcf6a3b, 0x0016a96f, 0xfff6db89, 0x0002edf6, 0xffff6d8d,
+        0x328dd556, 0xeb66aae0, 0x0c619444, 0xf7bc6889, 0x05b0c74b, 0xfc15365c, 0x02a5f535, 0xfe435633, 0x011801be, 0xff587d03, 0x005e0a48, 0xffcf4262, 0x0016c390, 0xfff6ce97, 0x0002f246, 0xffff6d40,
+        0x3140d82e, 0xebc09a94, 0x0c3716da, 0xf7d348a4, 0x05a42890, 0xfc1bfd22, 0x02a2903e, 0xfe44d4d3, 0x01178152, 0xff588913, 0x005e2687, 0xffcf227b, 0x0016d9c9, 0xfff6c356, 0x0002f5fc, 0xffff6d1a,
+        0x2ff42933, 0xec1c9e6d, 0x0c0b1a37, 0xf7eb39cc, 0x0596c6ff, 0xfc234d75, 0x029ecc3c, 0xfe469325, 0x0116d7d7, 0xff58ae5d, 0x005e3427, 0xffcf0a77, 0x0016ec22, 0xfff6b9c1, 0x0002f919, 0xffff6d17,
+        0x2ea7e2c0, 0xec7aa45b, 0x0bdda783, 0xf80436c0, 0x0588a5bf, 0xfc2b2567, 0x029aaa5a, 0xfe489077, 0x011605b5, 0xff58eca8, 0x005e3347, 0xffcefa44, 0x0016faa5, 0xfff6b1d5, 0x0002fba0, 0xffff6d38,
+        0x2d5c1f0e, 0xecda9a39, 0x0baec80a, 0xf81e3a25, 0x0579c812, 0xfc3382fb, 0x02962bd1, 0xfe4acc0e, 0x01150b5a, 0xff5943b4, 0x005e240a, 0xffcef1cf, 0x0017055b, 0xfff6ab8c, 0x0002fd94, 0xffff6d7c,
+        0x2c10f82d, 0xed3c6dce, 0x0b7e853c, 0xf8393e81, 0x056a314b, 0xfc3c6420, 0x029151e3, 0xfe4d4526, 0x0113e937, 0xff59b340, 0x005e0694, 0xffcef106, 0x00170c4f, 0xfff6a6e2, 0x0002fef6, 0xffff6de2,
+        0x2ac68807, 0xeda00cd1, 0x0b4ce8a8, 0xf8553e3c, 0x0559e4da, 0xfc45c6b6, 0x028c1de0, 0xfe4ffaf6, 0x01129fc5, 0xff5a3b09, 0x005ddb0b, 0xffcef7d4, 0x00170f8a, 0xfff6a3d0, 0x0002ffc9, 0xffff6e67,
+        0x297ce85a, 0xee0564e8, 0x0b19fbfe, 0xf87233a4, 0x0548e63f, 0xfc4fa88f, 0x02869122, 0xfe52ecab, 0x01112f81, 0xff5adac6, 0x005da198, 0xffcf0623, 0x00170f18, 0xfff6a252, 0x00030010, 0xffff6f0d,
+        0x283432b9, 0xee6c63ad, 0x0ae5c90b, 0xf89018eb, 0x05373912, 0xfc5a076a, 0x0280ad0f, 0xfe561969, 0x010f98eb, 0xff5b922d, 0x005d5a62, 0xffcf1bde, 0x00170b04, 0xfff6a262, 0x0002ffcd, 0xffff6fd1,
+        0x26ec8083, 0xeed4f6b0, 0x0ab059bc, 0xf8aee828, 0x0524e100, 0xfc64e0f9, 0x027a7318, 0xfe598050, 0x010ddc8c, 0xff5c60ee, 0x005d0597, 0xffcf38ec, 0x0017035a, 0xfff6a3f9, 0x0002ff03, 0xffff70b2,
+        0x25a5eae8, 0xef3f0b78, 0x0a79b814, 0xf8ce9b5d, 0x0511e1c6, 0xfc7032de, 0x0273e4b8, 0xfe5d2075, 0x010bfaee, 0xff5d46bb, 0x005ca363, 0xffcf5d36, 0x0016f828, 0xfff6a713, 0x0002fdb4, 0xffff71b0,
+        0x24608ae2, 0xefaa8f87, 0x0a41ee32, 0xf8ef2c71, 0x04fe3f39, 0xfc7bfaad, 0x026d0374, 0xfe60f8ea, 0x0109f4a2, 0xff5e433e, 0x005c33f6, 0xffcf88a2, 0x0016e979, 0xfff6aba9, 0x0002fbe4, 0xffff72c9,
+        0x231c7932, 0xf017705a, 0x0a09064e, 0xf9109535, 0x04e9fd3c, 0xfc8835ed, 0x0265d0dd, 0xfe6508b6, 0x0107ca3c, 0xff5f5621, 0x005bb77f, 0xffcfbb17, 0x0016d75b, 0xfff6b1b4, 0x0002f995, 0xffff73fc,
+        0x21d9ce63, 0xf0859b6e, 0x09cf0ab4, 0xf932cf65, 0x04d51fc6, 0xfc94e216, 0x025e4e8b, 0xfe694edd, 0x01057c57, 0xff607f0b, 0x005b2e31, 0xffcff478, 0x0016c1dc, 0xfff6b92d, 0x0002f6c9, 0xffff7549,
+        0x2098a2bf, 0xf0f4fe3d, 0x099405c6, 0xf955d4a7, 0x04bfaadf, 0xfca1fc96, 0x02567e22, 0xfe6dca58, 0x01030b8e, 0xff61bd9f, 0x005a9840, 0xffd034ac, 0x0016a90a, 0xfff6c20f, 0x0002f385, 0xffff76ae,
+        0x1f590e55, 0xf1658649, 0x095801f8, 0xf9799e8f, 0x04a9a29e, 0xfcaf82ca, 0x024e614c, 0xfe727a1f, 0x01007885, 0xff631180, 0x0059f5e1, 0xffd07b95, 0x00168cf2, 0xfff6cc52, 0x0002efca, 0xffff782a,
+        0x1e1b28f2, 0xf1d72114, 0x091b09d1, 0xf99e269e, 0x04930b2b, 0xfcbd7206, 0x0245f9bf, 0xfe775d1f, 0x00fdc3e0, 0xff647a4b, 0x0059474a, 0xffd0c915, 0x00166da5, 0xfff6d7f0, 0x0002eb9c, 0xffff79bc,
+        0x1cdf0a20, 0xf249bc2c, 0x08dd27e6, 0xf9c36642, 0x047be8bc, 0xfccbc793, 0x023d4937, 0xfe7c7243, 0x00faee49, 0xff65f79e, 0x00588cb4, 0xffd11d0f, 0x00164b32, 0xfff6e4e1, 0x0002e6fe, 0xffff7b63,
+        0x1ba4c923, 0xf2bd4523, 0x089e66dd, 0xf9e956da, 0x04643f95, 0xfcda80ad, 0x0234517a, 0xfe81b86d, 0x00f7f86e, 0xff678912, 0x0057c658, 0xffd17764, 0x001625a7, 0xfff6f31d, 0x0002e1f3, 0xffff7d1f,
+        0x1a6c7cf9, 0xf331a99b, 0x085ed167, 0xfa0ff1b6, 0x044c1409, 0xfce99a86, 0x022b1455, 0xfe872e7c, 0x00f4e2ff, 0xff692e3f, 0x0056f471, 0xffd1d7f5, 0x0015fd15, 0xfff7029f, 0x0002dc7d, 0xffff7eed,
+        0x19363c54, 0xf3a6d741, 0x081e7241, 0xfa373017, 0x04336a75, 0xfcf91246, 0x0221939d, 0xfe8cd349, 0x00f1aeb2, 0xff6ae6ba, 0x0056173b, 0xffd23ea1, 0x0015d18b, 0xfff7135d, 0x0002d6a0, 0xffff80cd,
+        0x18021d9d, 0xf41cbbd3, 0x07dd5430, 0xfa5f0b30, 0x041a4744, 0xfd08e50c, 0x0217d12d, 0xfe92a5a7, 0x00ee5c3e, 0xff6cb218, 0x00552ef3, 0xffd2ab47, 0x0015a31b, 0xfff72551, 0x0002d060, 0xffff82bf,
+        0x16d036eb, 0xf493451f, 0x079b8203, 0xfa877c29, 0x0400aeec, 0xfd190fed, 0x020dcee8, 0xfe98a466, 0x00eaec5e, 0xff6e8fe9, 0x00543bd8, 0xffd31dc7, 0x001571d5, 0xfff73873, 0x0002c9be, 0xffff84c0,
+        0x15a09e09, 0xf50a610a, 0x0759068f, 0xfab07c1d, 0x03e6a5ee, 0xfd298ff6, 0x02038eb7, 0xfe9ece4f, 0x00e75fd1, 0xff707fbd, 0x00533e29, 0xffd395fd, 0x00153dca, 0xfff74cba, 0x0002c2be, 0xffff86d0,
+        0x1473686d, 0xf581fd8b, 0x0715ecae, 0xfada0420, 0x03cc30d4, 0xfd3a622b, 0x01f9128a, 0xfea52227, 0x00e3b758, 0xff728121, 0x00523626, 0xffd413c9, 0x0015070b, 0xfff76220, 0x0002bb64, 0xffff88ee,
+        0x1348ab3a, 0xf5fa08b5, 0x06d23f3d, 0xfb040d3b, 0x03b15431, 0xfd4b8389, 0x01ee5c55, 0xfeab9eb2, 0x00dff3b7, 0xff7493a2, 0x00512412, 0xffd49705, 0x0014cdab, 0xfff7789c, 0x0002b3b3, 0xffff8b19,
+        0x12207b3e, 0xf67270b1, 0x068e091c, 0xfb2e906f, 0x039614a1, 0xfd5cf105, 0x01e36e14, 0xfeb242ac, 0x00dc15b4, 0xff76b6ca, 0x0050082f, 0xffd51f90, 0x001491b9, 0xfff79026, 0x0002abad, 0xffff8d50,
+        0x10faecee, 0xf6eb23c6, 0x0649552a, 0xfb5986b6, 0x037a76c7, 0xfd6ea790, 0x01d849c7, 0xfeb90cce, 0x00d81e1a, 0xff78ea20, 0x004ee2c1, 0xffd5ad44, 0x00145349, 0xfff7a8b6, 0x0002a357, 0xffff8f92,
+        0x0fd81464, 0xf7641059, 0x06042e45, 0xfb84e906, 0x035e7f4e, 0xfd80a411, 0x01ccf173, 0xfebffbd0, 0x00d40db3, 0xff7b2d2d, 0x004db40c, 0xffd63ffe, 0x0014126c, 0xfff7c245, 0x00029ab2, 0xffff91de,
+        0x0eb80562, 0xf7dd24ef, 0x05be9f49, 0xfbb0b04e, 0x034232e6, 0xfd92e36c, 0x01c16720, 0xfec70e64, 0x00cfe54f, 0xff7d7f76, 0x004c7c55, 0xffd6d798, 0x0013cf36, 0xfff7dcc8, 0x000291c3, 0xffff9434,
+        0x0d9ad348, 0xf856502d, 0x0578b30e, 0xfbdcd57a, 0x03259644, 0xfda5627e, 0x01b5acdd, 0xfece433a, 0x00cba5bc, 0xff7fe07f, 0x004b3be3, 0xffd773ed, 0x001389b7, 0xfff7f83a, 0x0002888c, 0xffff9691,
+        0x0c80911b, 0xf8cf80de, 0x05327467, 0xfc095174, 0x0308ae24, 0xfdb81e22, 0x01a9c4bc, 0xfed598fe, 0x00c74fce, 0xff824fca, 0x0049f2fc, 0xffd814d7, 0x00134204, 0xfff81490, 0x00027f11, 0xffff98f5,
+        0x0b69517e, 0xf948a5f0, 0x04ebee1c, 0xfc361d25, 0x02eb7f44, 0xfdcb132d, 0x019db0d0, 0xfedd0e5c, 0x00c2e457, 0xff84ccdb, 0x0048a1e7, 0xffd8ba31, 0x0012f82e, 0xfff831c3, 0x00027555, 0xffff9b60,
+        0x0a5526b0, 0xf9c1ae7b, 0x04a52af2, 0xfc633173, 0x02ce0e67, 0xfdde3e6f, 0x01917334, 0xfee4a1fa, 0x00be642f, 0xff875731, 0x004748ed, 0xffd963d4, 0x0012ac48, 0xfff84fcb, 0x00026b5b, 0xffff9dd0,
+        0x0944228e, 0xfa3a89be, 0x045e359f, 0xfc908746, 0x02b0604f, 0xfdf19cb9, 0x01850e00, 0xfeec527e, 0x00b9d02b, 0xff89ee4d, 0x0045e856, 0xffda1199, 0x00125e66, 0xfff86e9e, 0x00026126, 0xffffa045,
+        0x08365690, 0xfab32723, 0x041718d2, 0xfcbe1789, 0x029279c4, 0xfe052ad4, 0x01788354, 0xfef41e8c, 0x00b52925, 0xff8c91ad, 0x0044806c, 0xffdac35a, 0x00120e9b, 0xfff88e35, 0x000256b9, 0xffffa2be,
+        0x072bd3c5, 0xfb2b7641, 0x03cfdf29, 0xfcebdb26, 0x02745f8c, 0xfe18e58c, 0x016bd54f, 0xfefc04c6, 0x00b06ff7, 0xff8f40d0, 0x00431177, 0xffdb78ef, 0x0011bcf9, 0xfff8ae88, 0x00024c18, 0xffffa539,
+        0x0624aad6, 0xfba366df, 0x03889336, 0xfd19cb0e, 0x02561670, 0xfe2cc9a7, 0x015f0612, 0xff0403cc, 0x00aba57c, 0xff91fb31, 0x00419bc2, 0xffdc3231, 0x00116994, 0xfff8cf8d, 0x00024146, 0xffffa7b7,
+        0x0520ec00, 0xfc1ae8f2, 0x03413f7b, 0xfd47e035, 0x0237a337, 0xfe40d3ed, 0x015217c0, 0xff0c1a3c, 0x00a6ca90, 0xff94c04f, 0x00401f98, 0xffdceef9, 0x00111480, 0xfff8f13c, 0x00023645, 0xffffaa35,
+        0x0420a716, 0xfc91eca1, 0x02f9ee68, 0xfd761395, 0x02190aa6, 0xfe550124, 0x01450c7f, 0xff1446b5, 0x00a1e00f, 0xff978fa6, 0x003e9d42, 0xffddaf1e, 0x0010bdcf, 0xfff9138e, 0x00022b19, 0xffffacb4,
+        0x0323eb7f, 0xfd086246, 0x02b2aa5c, 0xfda45e2c, 0x01fa5183, 0xfe694e12, 0x0137e672, 0xff1c87d3, 0x009ce6d8, 0xff9a68b0, 0x003d150d, 0xffde727a, 0x00106595, 0xfff93679, 0x00021fc5, 0xffffaf33,
+};
+
+// cmd-line: fir -l 7 -s 44100 -c 19876 -n 16 -b 9.62
+const int32_t dn_sampler_filter_coefficients[] __attribute__ ((aligned (32))) = {
+        0x736144b5, 0x0c333a22, 0xf4fca390, 0x09424904, 0xf8c92a41, 0x052ac04c, 0xfca4fc64, 0x01ed8cc7, 0xff119cc0, 0x0053ba6e, 0xfff9a80d, 0xffeaeaab, 0x001690d9, 0xfff11dcd, 0x000715d9, 0xfffdb4b9,
+        0x735ed3aa, 0x0b433de8, 0xf560f0f3, 0x091282c4, 0xf8dd5ccf, 0x0525cb66, 0xfca23e3d, 0x01f33960, 0xff0bc9c2, 0x00586127, 0xfff68603, 0xffecbad5, 0x0015ab8b, 0xfff17c10, 0x0006f71a, 0xfffdbc2f,
+        0x735780bb, 0x0a55a98f, 0xf5c5b2a1, 0x08e1ea27, 0xf8f25767, 0x0520366d, 0xfc9ff262, 0x01f89c98, 0xff0620a4, 0x005cf349, 0xfff36c0d, 0xffee8913, 0x0014c5dc, 0xfff1db1a, 0x0006d7d7, 0xfffdc3db,
+        0x734b4c77, 0x096a8a51, 0xf62adb7c, 0x08b086aa, 0xf9081629, 0x051a030f, 0xfc9e186a, 0x01fdb637, 0xff00a1d8, 0x00617065, 0xfff05a84, 0xfff0552d, 0x0013dfed, 0xfff23ada, 0x0006b817, 0xfffdcbba,
+        0x733a37d2, 0x0881ed1f, 0xf6905e79, 0x087e5fd7, 0xf91e9521, 0x05133308, 0xfc9cafe0, 0x0202860e, 0xfefb4dc7, 0x0065d80c, 0xffed51bc, 0xfff21ee8, 0x0012f9de, 0xfff29b40, 0x000697e0, 0xfffdd3ca,
+        0x7324441e, 0x079bdea7, 0xf6f62e9d, 0x084b7d43, 0xf935d048, 0x050bc828, 0xfc9bb83e, 0x02070bf9, 0xfef624d8, 0x006a29d6, 0xffea520a, 0xfff3e60f, 0x001213d0, 0xfff2fc3d, 0x00067739, 0xfffddc07,
+        0x7309730f, 0x06b86b52, 0xf75c3eff, 0x0817e68c, 0xf94dc388, 0x0503c44d, 0xfc9b30f3, 0x020b47dd, 0xfef12766, 0x006e655c, 0xffe75bbe, 0xfff5aa69, 0x00112de1, 0xfff35dc1, 0x00065629, 0xfffde470,
+        0x72e9c6b8, 0x05d79f40, 0xf7c282cb, 0x07e3a35a, 0xf9666ab7, 0x04fb2969, 0xfc9b195f, 0x020f39ab, 0xfeec55cc, 0x00728a3d, 0xffe46f2a, 0xfff76bc2, 0x00104831, 0xfff3bfbc, 0x000634b6, 0xfffded03,
+        0x72c5418e, 0x04f98649, 0xf828ed43, 0x07aebb5d, 0xf97fc19e, 0x04f1f97c, 0xfc9b70d6, 0x0212e15c, 0xfee7b059, 0x0076981a, 0xffe18c9a, 0xfff929e3, 0x000f62de, 0xfff4221f, 0x000612e8, 0xfffdf5bc,
+        0x729be665, 0x041e2bfe, 0xf88f71bf, 0x0779364a, 0xf999c3f4, 0x04e83697, 0xfc9c369c, 0x02163ef1, 0xfee33759, 0x007a8e98, 0xffdeb45b, 0xfffae49b, 0x000e7e08, 0xfff484db, 0x0005f0c4, 0xfffdfe9b,
+        0x726db871, 0x03459ba4, 0xf8f603ae, 0x07431bdf, 0xf9b46d64, 0x04dde2da, 0xfc9d69eb, 0x02195278, 0xfedeeb11, 0x007e6d61, 0xffdbe6b6, 0xfffc9bb4, 0x000d99cc, 0xfff4e7e1, 0x0005ce51, 0xfffe079b,
+        0x723abb44, 0x026fe039, 0xf95c9699, 0x070c73dd, 0xf9cfb988, 0x04d30074, 0xfc9f09ee, 0x021c1c06, 0xfedacbbf, 0x00823422, 0xffd923f4, 0xfffe4efd, 0x000cb647, 0xfff54b20, 0x0005ab95, 0xfffe10bc,
+        0x7202f2d3, 0x019d046d, 0xf9c31e22, 0x06d5460b, 0xf9eba3ef, 0x04c791a4, 0xfca115c5, 0x021e9bbb, 0xfed6d99c, 0x0085e28b, 0xffd66c59, 0xfffffe46, 0x000bd397, 0xfff5ae8c, 0x00058898, 0xfffe19fa,
+        0x71c6636d, 0x00cd12a4, 0xfa298e07, 0x069d9a31, 0xfa082817, 0x04bb98b5, 0xfca38c83, 0x0220d1bf, 0xfed314da, 0x00897851, 0xffd3c02a, 0x0001a95d, 0x000af1d9, 0xfff61214, 0x0005655e, 0xfffe2354,
+        0x718511c2, 0x000014f8, 0xfa8fda21, 0x0665781b, 0xfa254176, 0x04af1804, 0xfca66d2e, 0x0222be45, 0xfecf7da3, 0x008cf52d, 0xffd11fa9, 0x00035015, 0x000a1129, 0xfff675ab, 0x000541f0, 0xfffe2cc8,
+        0x713f02e0, 0xff361534, 0xfaf5f669, 0x062ce795, 0xfa42eb75, 0x04a211f8, 0xfca9b6bf, 0x02246187, 0xfecc141d, 0x009058da, 0xffce8b13, 0x0004f23e, 0x000931a3, 0xfff6d942, 0x00051e52, 0xfffe3652,
+        0x70f43c32, 0xfe6f1cd7, 0xfb5bd6f4, 0x05f3f06b, 0xfa61216f, 0x04948906, 0xfcad6827, 0x0225bbca, 0xfec8d867, 0x0093a31a, 0xffcc02a8, 0x00068fad, 0x00085362, 0xfff73ccb, 0x0004fa8b, 0xfffe3ff2,
+        0x70a4c37f, 0xfdab350f, 0xfbc16ff6, 0x05ba9a6b, 0xfa7fdeba, 0x04867fb3, 0xfcb18047, 0x0226cd5b, 0xfec5ca9a, 0x0096d3af, 0xffc986a1, 0x00082835, 0x00077681, 0xfff7a037, 0x0004d6a1, 0xfffe49a4,
+        0x70509eec, 0xfcea66be, 0xfc26b5c5, 0x0580ed5f, 0xfa9f1e9e, 0x0477f88d, 0xfcb5fdf7, 0x02279691, 0xfec2eaca, 0x0099ea62, 0xffc71738, 0x0009bbab, 0x00069b1b, 0xfff8037a, 0x0004b29a, 0xfffe5367,
+        0x6ff7d4f8, 0xfc2cba75, 0xfc8b9cda, 0x0546f10f, 0xfabedc5a, 0x0468f62e, 0xfcbae002, 0x022817ca, 0xfec03901, 0x009ce6fe, 0xffc4b4a4, 0x000b49e6, 0x0005c149, 0xfff86686, 0x00048e7c, 0xfffe5d38,
+        0x6f9a6c7f, 0xfb723876, 0xfcf019cd, 0x050cad3f, 0xfadf1328, 0x04597b40, 0xfcc0252b, 0x0228516f, 0xfebdb547, 0x009fc954, 0xffc25f1a, 0x000cd2bd, 0x0004e926, 0xfff8c94c, 0x00046a4c, 0xfffe6716,
+        0x6f386cb6, 0xfabae8b2, 0xfd54215c, 0x04d229b1, 0xfaffbe36, 0x04498a72, 0xfcc5cc26, 0x022843f0, 0xfebb5f9b, 0x00a29136, 0xffc016cb, 0x000e5609, 0x000412c9, 0xfff92bc0, 0x00044612, 0xfffe70ff,
+        0x6ed1dd2e, 0xfa06d2ca, 0xfdb7a869, 0x04976e20, 0xfb20d8ad, 0x04392684, 0xfccbd3a0, 0x0227efc6, 0xfeb937f9, 0x00a53e7b, 0xffbddbe8, 0x000fd3a3, 0x00033e4c, 0xfff98dd6, 0x000421d2, 0xfffe7aef,
+        0x6e66c5ce, 0xf955fe0c, 0xfe1aa3fc, 0x045c8240, 0xfb425db0, 0x0428523d, 0xfcd23a3a, 0x02275572, 0xfeb73e54, 0x00a7d0ff, 0xffbbae9f, 0x00114b67, 0x00026bc6, 0xfff9ef80, 0x0003fd92, 0xfffe84e7,
+        0x6df72ed9, 0xf8a87178, 0xfe7d0942, 0x04216dc0, 0xfb64485b, 0x0417106e, 0xfcd8fe8b, 0x0226757e, 0xfeb5729b, 0x00aa48a0, 0xffb98f1c, 0x0012bd30, 0x00019b4e, 0xfffa50b1, 0x0003d957, 0xfffe8ee3,
+        0x6d8320e6, 0xf7fe33ba, 0xfedecd90, 0x03e63846, 0xfb8693c6, 0x040563f4, 0xfce01f21, 0x0225507c, 0xfeb3d4b7, 0x00aca542, 0xffb77d88, 0x001428db, 0x0000ccfc, 0xfffab15e, 0x0003b527, 0xfffe98e2,
+        0x6d0aa4e6, 0xf7574b2b, 0xff3fe663, 0x03aae970, 0xfba93b01, 0x03f34fb2, 0xfce79a7f, 0x0223e706, 0xfeb26489, 0x00aee6ca, 0xffb57a0b, 0x00158e47, 0x000000e6, 0xfffb117a, 0x00039108, 0xfffea2e1,
+        0x6c8dc41f, 0xf6b3bdd3, 0xffa04963, 0x036f88d2, 0xfbcc391d, 0x03e0d697, 0xfcef6f20, 0x022239bc, 0xfeb121ee, 0x00b10d23, 0xffb384ca, 0x0016ed53, 0xffff3721, 0xfffb70fa, 0x00036cfe, 0xfffeacdf,
+        0x6c0c882a, 0xf6139169, 0xffffec5f, 0x03341df4, 0xfbef8924, 0x03cdfb99, 0xfcf79b75, 0x02204949, 0xfeb00cbf, 0x00b3183c, 0xffb19de7, 0x001845e0, 0xfffe6fc3, 0xfffbcfd2, 0x00034910, 0xfffeb6db,
+        0x6b86faf8, 0xf576cb4e, 0x005ec552, 0x02f8b055, 0xfc13261f, 0x03bac1b4, 0xfd001de8, 0x021e165d, 0xfeaf24cc, 0x00b50805, 0xffafc584, 0x001997d0, 0xfffdaadf, 0xfffc2df6, 0x00032541, 0xfffec0d2,
+        0x6afd26cb, 0xf4dd7092, 0x00bcca63, 0x02bd4768, 0xfc370b14, 0x03a72bf0, 0xfd08f4d6, 0x021ba1b2, 0xfeae69e1, 0x00b6dc75, 0xffadfbbe, 0x001ae306, 0xfffce88b, 0xfffc8b5c, 0x00030196, 0xfffecac3,
+        0x6a6f1638, 0xf44785f1, 0x0119f1e4, 0x0281ea90, 0xfc5b3309, 0x03933d58, 0xfd121e99, 0x0218ec06, 0xfeaddbc4, 0x00b89584, 0xffac40b3, 0x001c2765, 0xfffc28d9, 0xfffce7f8, 0x0002de16, 0xfffed4ab,
+        0x69dcd425, 0xf3b50fd6, 0x01763256, 0x0246a125, 0xfc7f9902, 0x037ef900, 0xfd1b9980, 0x0215f621, 0xfead7a37, 0x00ba3330, 0xffaa947c, 0x001d64d5, 0xfffb6bdd, 0xfffd43c1, 0x0002bac4, 0xfffede8a,
+        0x69466bc8, 0xf3261255, 0x01d18265, 0x020b726f, 0xfca43803, 0x036a6201, 0xfd2563d3, 0x0212c0d2, 0xfead44f4, 0x00bbb579, 0xffa8f730, 0x001e9b3a, 0xfffab1a8, 0xfffd9eab, 0x000297a5, 0xfffee85e,
+        0x68abe8a8, 0xf29a9133, 0x022bd8ee, 0x01d065a8, 0xfcc90b12, 0x03557b7a, 0xfd2f7bd1, 0x020f4cec, 0xfead3bb2, 0x00bd1c63, 0xffa768e6, 0x001fca7d, 0xfff9fa4d, 0xfffdf8ae, 0x000274be, 0xfffef225,
+        0x680d5698, 0xf2128fde, 0x02852cfc, 0x019581f9, 0xfcee0d33, 0x03404890, 0xfd39dfb4, 0x020b9b4c, 0xfead5e22, 0x00be67f6, 0xffa5e9b1, 0x0020f288, 0xfff945dc, 0xfffe51be, 0x00025214, 0xfffefbde,
+        0x676ac1bb, 0xf18e1174, 0x02dd75ca, 0x015ace79, 0xfd133970, 0x032acc6d, 0xfd448dae, 0x0207acd4, 0xfeadabef, 0x00bf983d, 0xffa479a2, 0x00221344, 0xfff89465, 0xfffea9d2, 0x00022fa9, 0xffff0587,
+        0x66c4367d, 0xf10d18bd, 0x0334aac4, 0x0120522f, 0xfd388ad1, 0x03150a3f, 0xfd4f83eb, 0x0203826c, 0xfeae24c1, 0x00c0ad48, 0xffa318c7, 0x00232c9d, 0xfff7e5f9, 0xffff00e1, 0x00020d84, 0xffff0f1f,
+        0x6619c197, 0xf08fa82f, 0x038ac385, 0x00e6140f, 0xfd5dfc63, 0x02ff0538, 0xfd5ac08e, 0x01ff1d04, 0xfeaec838, 0x00c1a728, 0xffa1c72f, 0x00243e7f, 0xfff73aa7, 0xffff56e3, 0x0001eba8, 0xffff18a4,
+        0x656b700a, 0xf015c1ee, 0x03dfb7dd, 0x00ac1af9, 0xfd838938, 0x02e8c08e, 0xfd6641b8, 0x01fa7d91, 0xfeaf95f2, 0x00c285f4, 0xffa084e3, 0x002548d9, 0xfff6927e, 0xffffabcd, 0x0001ca18, 0xffff2215,
+        0x64b94f22, 0xef9f67cb, 0x04337fcb, 0x00726dbb, 0xfda92c63, 0x02d23f7a, 0xfd720581, 0x01f5a50d, 0xfeb08d86, 0x00c349c4, 0xff9f51eb, 0x00264b9a, 0xfff5ed8b, 0xffffff99, 0x0001a8da, 0xffff2b70,
+        0x64036c6f, 0xef2c9b43, 0x04861383, 0x0039130c, 0xfdcee0ff, 0x02bb8537, 0xfd7e09fc, 0x01f0947a, 0xfeb1ae87, 0x00c3f2b6, 0xff9e2e50, 0x002746b2, 0xfff54bdc, 0x0000523d, 0x000187f0, 0xffff34b6,
+        0x6349d5c9, 0xeebd5d81, 0x04d76b6b, 0x00001191, 0xfdf4a22a, 0x02a49505, 0xfd8a4d37, 0x01eb4cde, 0xfeb2f884, 0x00c480e9, 0xff9d1a14, 0x00283a12, 0xfff4ad7e, 0x0000a3b3, 0x0001675f, 0xffff3de3,
+        0x628c994c, 0xee51af5f, 0x0527801d, 0xffc76fd5, 0xfe1a6b08, 0x028d7223, 0xfd96cd3d, 0x01e5cf44, 0xfeb46b07, 0x00c4f480, 0xff9c1539, 0x002925ae, 0xfff4127d, 0x0000f3f1, 0x00014729, 0xffff46f7,
+        0x61cbc559, 0xede99165, 0x05764a68, 0xff8f344f, 0xfe4036c5, 0x02761fd3, 0xfda3880f, 0x01e01cbe, 0xfeb60596, 0x00c54da2, 0xff9b1fc1, 0x002a0979, 0xfff37ae4, 0x000142f1, 0x00012754, 0xffff4ff1,
+        0x61076890, 0xed8503c7, 0x05c3c34e, 0xff576560, 0xfe660094, 0x025ea157, 0xfdb07bb0, 0x01da3661, 0xfeb7c7b0, 0x00c58c79, 0xff9a39a9, 0x002ae568, 0xfff2e6bf, 0x000190ac, 0x000107e1, 0xffff58d0,
+        0x603f91d5, 0xed24066b, 0x060fe408, 0xff20094d, 0xfe8bc3ad, 0x0246f9f3, 0xfdbda61a, 0x01d41d4a, 0xfeb9b0d3, 0x00c5b132, 0xff9962ec, 0x002bb971, 0xfff25619, 0x0001dd1b, 0x0000e8d4, 0xffff6192,
+        0x5f745049, 0xecc698e6, 0x065aa604, 0xfee92646, 0xfeb17b53, 0x022f2cea, 0xfdcb0546, 0x01cdd297, 0xfebbc078, 0x00c5bbfc, 0xff989b85, 0x002c858d, 0xfff1c8fa, 0x00022837, 0x0000ca30, 0xffff6a38,
+        0x5ea5b34c, 0xec6cba79, 0x06a402e4, 0xfeb2c261, 0xfed722d0, 0x02173d81, 0xfdd89727, 0x01c7576d, 0xfebdf613, 0x00c5ad0a, 0xff97e36c, 0x002d49b4, 0xfff13f6c, 0x000271fa, 0x0000abf8, 0xffff72be,
+        0x5dd3ca7a, 0xec166a19, 0x06ebf483, 0xfe7ce399, 0xfefcb57a, 0x01ff2ef9, 0xfde659af, 0x01c0acf5, 0xfec05114, 0x00c58494, 0xff973a96, 0x002e05df, 0xfff0b977, 0x0002ba5f, 0x00008e30, 0xffff7b26,
+        0x5cfea5aa, 0xebc3a669, 0x073274f1, 0xfe478fd2, 0xff222eac, 0x01e70494, 0xfdf44acc, 0x01b9d45b, 0xfec2d0e8, 0x00c542d1, 0xff96a0f8, 0x002eba0a, 0xfff03724, 0x0003015f, 0x000070d9, 0xffff836d,
+        0x5c2654ed, 0xeb746dbe, 0x07777e74, 0xfe12ccd1, 0xff4789d1, 0x01cec194, 0xfe026869, 0x01b2ced1, 0xfec574f9, 0x00c4e7fe, 0xff961684, 0x002f6630, 0xffefb87a, 0x000346f6, 0x000053f7, 0xffff8b93,
+        0x5b4ae88d, 0xeb28be1f, 0x07bb0b8b, 0xfddea042, 0xff6cc25a, 0x01b66936, 0xfe10b06f, 0x01ab9d8b, 0xfec83caa, 0x00c47459, 0xff959b29, 0x00300a4f, 0xffef3d7f, 0x00038b1d, 0x0000378c, 0xffff9398,
+        0x5a6c7108, 0xeae09544, 0x07fd16eb, 0xfdab0fb6, 0xff91d3c6, 0x019dfeb6, 0xfe1f20c5, 0x01a441c2, 0xfecb275e, 0x00c3e824, 0xff952ed7, 0x0030a665, 0xffeec63a, 0x0003cdd1, 0x00001b9a, 0xffff9b7a,
+        0x598aff13, 0xea9bf097, 0x083d9b81, 0xfd7820a0, 0xffb6b99f, 0x0185854f, 0xfe2db74f, 0x019cbcb1, 0xfece3472, 0x00c343a4, 0xff94d178, 0x00313a72, 0xffee52b1, 0x00040f0d, 0x00000024, 0xffffa339,
+        0x58a6a397, 0xea5acd38, 0x087c9471, 0xfd45d856, 0xffdb6f7c, 0x016d0037, 0xfe3c71f1, 0x01950f98, 0xfed16342, 0x00c2871f, 0xff9482f8, 0x0031c677, 0xffede2e7, 0x00044ecb, 0xffffe52d, 0xffffaad3,
+        0x57bf6fae, 0xea1d27f7, 0x08b9fd18, 0xfd143c12, 0xfffff100, 0x015472a1, 0xfe4b4e8c, 0x018d3bb8, 0xfed4b325, 0x00c1b2e0, 0xff944340, 0x00324a74, 0xffed76e3, 0x00048d0a, 0xffffcab5, 0xffffb249,
+        0x56d574a2, 0xe9e2fd5b, 0x08f5d10a, 0xfce350f0, 0x002439db, 0x013bdfbc, 0xfe5a4b03, 0x01854258, 0xfed82370, 0x00c0c731, 0xff941236, 0x0032c66e, 0xffed0ea7, 0x0004c9c4, 0xffffb0bf, 0xffffb99a,
+        0x55e8c3ee, 0xe9ac49a0, 0x09300c14, 0xfcb31bec, 0x004845cc, 0x01234ab4, 0xfe696534, 0x017d24bf, 0xfedbb373, 0x00bfc463, 0xff93efbf, 0x00333a67, 0xffecaa36, 0x000504f6, 0xffff974d, 0xffffc0c5,
+        0x54f96f37, 0xe97908b8, 0x0968aa3b, 0xfc83a1e5, 0x006c10a0, 0x010ab6b0, 0xfe789b01, 0x0174e437, 0xfedf627d, 0x00beaac6, 0xff93dbc0, 0x0033a665, 0xffec4994, 0x00053e9e, 0xffff7e61, 0xffffc7ca,
+        0x54078851, 0xe9493649, 0x099fa7bb, 0xfc54e79a, 0x008f9631, 0x00f226d0, 0xfe87ea47, 0x016c820d, 0xfee32fdb, 0x00bd7aae, 0xff93d618, 0x00340a6d, 0xffebecc2, 0x000576b8, 0xffff65fc, 0xffffcea8,
+        0x53132138, 0xe91ccdb5, 0x09d5010b, 0xfc26f1ad, 0x00b2d26b, 0x00d99e31, 0xfe9750e8, 0x0163ff90, 0xfee71ad4, 0x00bc3470, 0xff93deaa, 0x00346687, 0xffeb93c3, 0x0005ad41, 0xffff4e20, 0xffffd55f,
+        0x521c4c10, 0xe8f3ca12, 0x0a08b2d9, 0xfbf9c49d, 0x00d5c147, 0x00c11feb, 0xfea6ccc3, 0x015b5e11, 0xfeeb22af, 0x00bad866, 0xff93f552, 0x0034babb, 0xffeb3e96, 0x0005e238, 0xffff36ce, 0xffffdbee,
+        0x51231b26, 0xe8ce2631, 0x0a3aba09, 0xfbcd64ca, 0x00f85ecf, 0x00a8af0c, 0xfeb65bb9, 0x01529ee3, 0xfeef46b0, 0x00b966e9, 0xff9419ef, 0x00350711, 0xffeaed3c, 0x00061599, 0xffff2007, 0xffffe255,
+        0x5027a0e9, 0xe8abdc9d, 0x0a6b13bc, 0xfba1d673, 0x011aa71d, 0x00904ea0, 0xfec5fbac, 0x0149c35a, 0xfef3861a, 0x00b7e055, 0xff944c5a, 0x00354b94, 0xffea9fb6, 0x00064764, 0xffff09ce, 0xffffe894,
+        0x4f29efed, 0xe88ce79a, 0x0a99bd47, 0xfb771db9, 0x013c965b, 0x007801aa, 0xfed5aa7e, 0x0140cccb, 0xfef7e02a, 0x00b6450a, 0xff948c6e, 0x0035884f, 0xffea5602, 0x00067797, 0xfffef421, 0xffffeeaa,
+        0x4e2a1ae8, 0xe871412a, 0x0ac6b43a, 0xfb4d3e97, 0x015e28c7, 0x005fcb26, 0xfee56614, 0x0137bc8f, 0xfefc541e, 0x00b49568, 0xff94da03, 0x0035bd4e, 0xffea1020, 0x0006a630, 0xfffedf04, 0xfffff498,
+        0x4d2834b0, 0xe858e30a, 0x0af1f65d, 0xfb243cea, 0x017f5aad, 0x0047ae09, 0xfef52c54, 0x012e93fc, 0xff00e133, 0x00b2d1d1, 0xff9534f0, 0x0035ea9d, 0xffe9ce0d, 0x0006d32f, 0xfffeca76, 0xfffffa5d,
+        0x4c245038, 0xe843c6b5, 0x0b1b81ad, 0xfafc1c6e, 0x01a0286c, 0x002fad3f, 0xff04fb25, 0x0125546c, 0xff0586a0, 0x00b0faaa, 0xff959d0a, 0x0036104b, 0xffe98fc8, 0x0006fe92, 0xfffeb678, 0xfffffff8,
+        0x4b1e8091, 0xe831e563, 0x0b435462, 0xfad4e0b9, 0x01c08e78, 0x0017cbae, 0xff14d073, 0x011bff38, 0xff0a439e, 0x00af1059, 0xff961224, 0x00362e66, 0xffe9554c, 0x00072859, 0xfffea30b, 0x0000056a,
+        0x4a16d8e5, 0xe823380d, 0x0b696ceb, 0xfaae8d43, 0x01e08952, 0x00000c33, 0xff24aa2a, 0x011295bb, 0xff0f1762, 0x00ad1346, 0xff969412, 0x003644fd, 0xffe91e99, 0x00075084, 0xfffe9030, 0x00000ab3,
+        0x490d6c79, 0xe817b76c, 0x0b8dc9ed, 0xfa89255f, 0x02001593, 0xffe871a0, 0xff348639, 0x0109194f, 0xff140121, 0x00ab03da, 0xff9722a5, 0x00365422, 0xffe8eba8, 0x00077712, 0xfffe7de7, 0x00000fd2,
+        0x48024ea7, 0xe80f5bfb, 0x0bb06a47, 0xfa64ac3f, 0x021f2fe5, 0xffd0fec1, 0xff446293, 0x00ff8b4f, 0xff19000e, 0x00a8e282, 0xff97bdac, 0x00365be6, 0xffe8bc77, 0x00079c04, 0xfffe6c2f, 0x000014c8,
+        0x46f592e2, 0xe80a1df5, 0x0bd14d0b, 0xfa4124f2, 0x023dd505, 0xffb9b656, 0xff543d2e, 0x00f5ed15, 0xff1e135b, 0x00a6afa8, 0xff9864f6, 0x00365c5b, 0xffe89101, 0x0007bf5b, 0xfffe5b0b, 0x00001994,
+        0x45e74cad, 0xe807f55b, 0x0bf07186, 0xfa1e9262, 0x025c01c5, 0xffa29b18, 0xff641402, 0x00ec3ffc, 0xff233a39, 0x00a46bbc, 0xff991851, 0x00365594, 0xffe8693f, 0x0007e116, 0xfffe4a79, 0x00001e37,
+        0x44d78fa0, 0xe808d9f1, 0x0c0dd738, 0xf9fcf758, 0x0279b30b, 0xff8bafb3, 0xff73e50e, 0x00e2855d, 0xff2873d6, 0x00a2172d, 0xff99d789, 0x003647a5, 0xffe8452d, 0x00080137, 0xfffe3a79, 0x000022b1,
+        0x43c66f62, 0xe80cc342, 0x0c297dd9, 0xf9dc567b, 0x0296e5d0, 0xff74f6cc, 0xff83ae52, 0x00d8be92, 0xff2dbf61, 0x009fb26c, 0xff9aa268, 0x003632a2, 0xffe824c5, 0x00081fbf, 0xfffe2b0d, 0x00002701,
+        0x42b3ffa9, 0xe813a89f, 0x0c436557, 0xf9bcb24a, 0x02b39724, 0xff5e72fb, 0xff936dd2, 0x00ceecf5, 0xff331c08, 0x009d3deb, 0xff9b78ba, 0x003616a2, 0xffe807ff, 0x00083cb0, 0xfffe1c32, 0x00002b28,
+        0x41a05437, 0xe81d8122, 0x0c5b8dd4, 0xf99e0d26, 0x02cfc429, 0xff4826cf, 0xffa3219a, 0x00c511dc, 0xff3888f8, 0x009aba1d, 0xff9c5a47, 0x0035f3b9, 0xffe7eed5, 0x0008580a, 0xfffe0dea, 0x00002f26,
+        0x408b80d9, 0xe82a43ac, 0x0c71f7a9, 0xf980694a, 0x02eb6a18, 0xff3214c9, 0xffb2c7b6, 0x00bb2e9f, 0xff3e055d, 0x00982778, 0xff9d46d6, 0x0035ca00, 0xffe7d93f, 0x000871cf, 0xfffe0034, 0x000032fb,
+        0x3f759967, 0xe839e6e9, 0x0c86a361, 0xf963c8cc, 0x03068640, 0xff1c3f63, 0xffc25e3b, 0x00b14493, 0xff439064, 0x0095866f, 0xff9e3e30, 0x0035998d, 0xffe7c735, 0x00088a02, 0xfffdf310, 0x000036a8,
+        0x3e5eb1bd, 0xe84c6152, 0x0c9991be, 0xf9482da0, 0x03211603, 0xff06a907, 0xffd1e340, 0x00a7550c, 0xff492937, 0x0092d77b, 0xff9f4019, 0x00356279, 0xffe7b8af, 0x0008a0a5, 0xfffde67c, 0x00003a2d,
+        0x3d46ddc1, 0xe861a92b, 0x0caac3b5, 0xf92d9997, 0x033b16dc, 0xfef15417, 0xffe154e3, 0x009d615d, 0xff4ecf02, 0x00901b11, 0xffa04c57, 0x003524dd, 0xffe7ada5, 0x0008b5ba, 0xfffdda79, 0x00003d89,
+        0x3c2e315a, 0xe879b487, 0x0cba3a6d, 0xf9140e5e, 0x03548659, 0xfedc42e7, 0xfff0b148, 0x00936ad6, 0xff5480f0, 0x008d51ab, 0xffa162ae, 0x0034e0d3, 0xffe7a60d, 0x0008c944, 0xfffdcf05, 0x000040be,
+        0x3b14c072, 0xe8947947, 0x0cc7f742, 0xf8fb8d7d, 0x036d621f, 0xfec777be, 0xfffff697, 0x008972c7, 0xff5a3e2c, 0x008a7bc1, 0xffa282e1, 0x00349674, 0xffe7a1de, 0x0008db46, 0xfffdc421, 0x000043cc,
+        0x39fa9ef3, 0xe8b1ed1c, 0x0cd3fbc0, 0xf8e4185a, 0x0385a7eb, 0xfeb2f4d9, 0x000f22fe, 0x007f7a7c, 0xff6005e1, 0x008799cd, 0xffa3acb4, 0x003445dc, 0xffe7a10d, 0x0008ebc1, 0xfffdb9cb, 0x000046b2,
+        0x38dfe0c6, 0xe8d2058b, 0x0cde49a8, 0xf8cdb036, 0x039d558e, 0xfe9ebc66, 0x001e34b4, 0x00758341, 0xff65d73a, 0x0084ac48, 0xffa4dfe8, 0x0033ef25, 0xffe7a391, 0x0008fabb, 0xfffdb002, 0x00004972,
+        0x37c499d0, 0xe8f4b7e9, 0x0ce6e2ea, 0xf8b85631, 0x03b468f1, 0xfe8ad087, 0x002d29f3, 0x006b8e5c, 0xff6bb163, 0x0081b3af, 0xffa61c3e, 0x0033926d, 0xffe7a95f, 0x00090836, 0xfffda6c5, 0x00004c0b,
+        0x36a8ddf3, 0xe919f961, 0x0cedc9a7, 0xf8a40b44, 0x03cae014, 0xfe773351, 0x003c00fd, 0x00619d15, 0xff719388, 0x007eb07b, 0xffa76176, 0x00332fcf, 0xffe7b26c, 0x00091435, 0xfffd9e13, 0x00004e7f,
+        0x358cc109, 0xe941bef3, 0x0cf30031, 0xf890d048, 0x03e0b90d, 0xfe63e6cb, 0x004ab81b, 0x0057b0ae, 0xff777cd6, 0x007ba32a, 0xffa8af51, 0x0032c769, 0xffe7bead, 0x00091ebd, 0xfffd95eb, 0x000050cd,
+        0x347056e3, 0xe96bfd76, 0x0cf6890a, 0xf87ea5f1, 0x03f5f20a, 0xfe50ecf0, 0x00594d9d, 0x004dca68, 0xff7d6c79, 0x00788c36, 0xffaa058d, 0x00325958, 0xffe7ce16, 0x000927d1, 0xfffd8e4d, 0x000052f7,
+        0x3353b349, 0xe998a999, 0x0cf866e1, 0xf86d8cd1, 0x040a894e, 0xfe3e47ac, 0x0067bfd8, 0x0043eb7f, 0xff83619f, 0x00756c1d, 0xffab63ea, 0x0031e5ba, 0xffe7e09c, 0x00092f75, 0xfffd8735, 0x000054fc,
+        0x3236e9f7, 0xe9c7b7e3, 0x0cf89c96, 0xf85d8555, 0x041e7d34, 0xfe2bf8de, 0x00760d2a, 0x003a152f, 0xff895b77, 0x0072435b, 0xffacca25, 0x00316cae, 0xffe7f631, 0x000935ad, 0xfffd80a4, 0x000056dd,
+        0x311a0e9b, 0xe9f91cb9, 0x0cf72d34, 0xf84e8fc9, 0x0431cc31, 0xfe1a0256, 0x008433f9, 0x003048ae, 0xff8f5930, 0x006f126b, 0xffae37fd, 0x0030ee53, 0xffe80eca, 0x00093a7f, 0xfffd7a98, 0x0000589b,
+        0x2ffd34d4, 0xea2ccc59, 0x0cf41bf7, 0xf840ac57, 0x044474ce, 0xfe0865d7, 0x009232b2, 0x0026872f, 0xff9559fb, 0x006bd9cd, 0xffafad2e, 0x00306ac8, 0xffe82a59, 0x00093ded, 0xfffd750f, 0x00005a36,
+        0x2ee07030, 0xea62bae0, 0x0cef6c43, 0xf833db04, 0x045675ab, 0xfdf72515, 0x00a007c9, 0x001cd1e4, 0xff9b5d0a, 0x006899fb, 0xffb12976, 0x002fe22c, 0xffe848d3, 0x00093ffe, 0xfffd7008, 0x00005baf,
+        0x2dc3d429, 0xea9adc49, 0x0ce921ab, 0xf8281bb6, 0x0467cd83, 0xfde641b7, 0x00adb1bb, 0x001329f7, 0xffa16190, 0x00655372, 0xffb2ac90, 0x002f54a1, 0xffe86a29, 0x000940b6, 0xfffd6b81, 0x00005d06,
+        0x2ca77428, 0xead52471, 0x0ce13feb, 0xf81d6e2e, 0x04787b24, 0xfdd5bd53, 0x00bb2f0b, 0x00099093, 0xffa766c0, 0x006206b1, 0xffb4363a, 0x002ec246, 0xffe88e4d, 0x00094019, 0xfffd6779, 0x00005e3d,
+        0x2b8b637b, 0xeb118714, 0x0cd7caec, 0xf813d20d, 0x04887d76, 0xfdc59972, 0x00c87e47, 0x000006db, 0xffad6bd0, 0x005eb431, 0xffb5c630, 0x002e2b3c, 0xffe8b532, 0x00093e2e, 0xfffd63ed, 0x00005f52,
+        0x2a6fb55e, 0xeb4ff7d4, 0x0cccc6bc, 0xf80b46d3, 0x0497d378, 0xfdb5d78f, 0x00d59e03, 0xfff68df1, 0xffb36ff9, 0x005b5c71, 0xffb75c2c, 0x002d8fa4, 0xffe8decb, 0x00093af8, 0xfffd60dd, 0x00006048,
+        0x29547ced, 0xeb906a35, 0x0cc03797, 0xf803cbdc, 0x04a67c41, 0xfda67913, 0x00e28cdd, 0xffed26f0, 0xffb97271, 0x0057ffec, 0xffb8f7ea, 0x002cefa1, 0xffe90b08, 0x0009367e, 0xfffd5e46, 0x0000611f,
+        0x2839cd30, 0xebd2d1a1, 0x0cb221de, 0xf7fd6065, 0x04b476fe, 0xfd977f5d, 0x00ef497a, 0xffe3d2f2, 0xffbf7274, 0x00549f1c, 0xffba9927, 0x002c4b53, 0xffe939db, 0x000930c4, 0xfffd5c26, 0x000061d8,
+        0x271fb90d, 0xec17216b, 0x0ca28a1a, 0xf7f8038c, 0x04c1c2f3, 0xfd88ebb9, 0x00fbd28a, 0xffda930a, 0xffc56f3e, 0x00513a7e, 0xffbc3f9d, 0x002ba2dc, 0xffe96b35, 0x000929d1, 0xfffd5a7c, 0x00006272,
+        0x2606534e, 0xec5d4ccd, 0x0c9174fa, 0xf7f3b44b, 0x04ce5f7d, 0xfd7abf64, 0x010826c4, 0xffd16848, 0xffcb680e, 0x004dd28c, 0xffbdeb07, 0x002af65f, 0xffe99f08, 0x000921aa, 0xfffd5945, 0x000062f0,
+        0x24edae9c, 0xeca546eb, 0x0c7ee754, 0xf7f0717e, 0x04da4c10, 0xfd6cfb8e, 0x011444e7, 0xffc853b6, 0xffd15c22, 0x004a67c0, 0xffbf9b21, 0x002a45fe, 0xffe9d545, 0x00091854, 0xfffd5880, 0x00006351,
+        0x23d5dd81, 0xecef02d5, 0x0c6ae622, 0xf7ee39e2, 0x04e58836, 0xfd5fa157, 0x01202bbe, 0xffbf565a, 0xffd74abe, 0x0046fa93, 0xffc14fa5, 0x002991db, 0xffea0ddc, 0x00090dd6, 0xfffd582a, 0x00006396,
+        0x22bef262, 0xed3a7388, 0x0c557681, 0xf7ed0c12, 0x04f01392, 0xfd52b1cf, 0x012bda1b, 0xffb67137, 0xffdd3325, 0x00438b7e, 0xffc3084f, 0x0028da1a, 0xffea48be, 0x00090236, 0xfffd5842, 0x000063c0,
+        0x21a8ff7e, 0xed878bf0, 0x0c3e9db5, 0xf7ece68c, 0x04f9edda, 0xfd462df6, 0x01374eda, 0xffada547, 0xffe3149e, 0x00401af9, 0xffc4c4da, 0x00281edd, 0xffea85dc, 0x0008f57a, 0xfffd58c5, 0x000063d0,
+        0x209416f2, 0xedd63ee5, 0x0c26611f, 0xf7edc7af, 0x050316e0, 0xfd3a16c0, 0x014288e0, 0xffa4f383, 0xffe8ee72, 0x003ca97b, 0xffc68502, 0x00276046, 0xffeac525, 0x0008e7a7, 0xfffd59b2, 0x000063c6,
+        0x1f804ab0, 0xee267f35, 0x0c0cc646, 0xf7efadbd, 0x050b8e8a, 0xfd2e6d0d, 0x014d871b, 0xff9c5cdc, 0xffeebfec, 0x0039377a, 0xffc84881, 0x00269e7a, 0xffeb068a, 0x0008d8c4, 0xfffd5b05, 0x000063a3,
+        0x1e6dac83, 0xee783f9e, 0x0bf1d2d0, 0xf7f296d7, 0x051354d5, 0xfd2331b0, 0x01584883, 0xff93e241, 0xfff48859, 0x0035c56c, 0xffca0f14, 0x0025d99b, 0xffeb49fc, 0x0008c8d7, 0xfffd5cbe, 0x00006368,
+        0x1d5c4e09, 0xeecb72d1, 0x0bd58c81, 0xf7f68103, 0x051a69d4, 0xfd18656f, 0x0162cc19, 0xff8b8498, 0xfffa470a, 0x003253c6, 0xffcbd876, 0x002511cd, 0xffeb8f6a, 0x0008b7e7, 0xfffd5ed8, 0x00006316,
+        0x1c4c40b6, 0xef200b76, 0x0bb7f940, 0xf7fb6a29, 0x0520cdb1, 0xfd0e08fb, 0x016d10e9, 0xff8344c4, 0xfffffb51, 0x002ee2fa, 0xffcda463, 0x00244733, 0xffebd6c4, 0x0008a5fa, 0xfffd6154, 0x000062ad,
+        0x1b3d95d1, 0xef75fc2b, 0x0b991f0f, 0xf8015015, 0x052680ae, 0xfd041cfa, 0x01771608, 0xff7b23a1, 0x0005a483, 0x002b737b, 0xffcf7299, 0x002379ef, 0xffec1ffa, 0x00089316, 0xfffd642d, 0x0000622e,
+        0x1a305e70, 0xefcd3787, 0x0b79040c, 0xf8083077, 0x052b8320, 0xfcfaa200, 0x0180da94, 0xff732209, 0x000b41fa, 0x002805ba, 0xffd142d3, 0x0022aa26, 0xffec6afc, 0x00087f43, 0xfffd6762, 0x0000619a,
+        0x1924ab7b, 0xf025b01a, 0x0b57ae75, 0xf81008e2, 0x052fd573, 0xfcf19894, 0x018a5db5, 0xff6b40cb, 0x0010d30e, 0x00249a28, 0xffd314cf, 0x0021d7fa, 0xffecb7b9, 0x00086a86, 0xfffd6af1, 0x000060f1,
+        0x181a8da5, 0xf07f586e, 0x0b3524a0, 0xf818d6cf, 0x0533782a, 0xfce9012c, 0x01939e9e, 0xff6380b5, 0x00165720, 0x00213134, 0xffd4e84a, 0x00210390, 0xffed0621, 0x000854e6, 0xfffd6ed6, 0x00006035,
+        0x17121573, 0xf0da230b, 0x0b116cff, 0xf822979b, 0x05366bdc, 0xfce0dc2f, 0x019c9c8b, 0xff5be28d, 0x001bcd8e, 0x001dcb4a, 0xffd6bd01, 0x00202d09, 0xffed5624, 0x00083e6a, 0xfffd7310, 0x00005f66,
+        0x160b5331, 0xf1360276, 0x0aec8e1c, 0xf82d488c, 0x0538b136, 0xfcd929f4, 0x01a556c1, 0xff546713, 0x002135bd, 0x001a68d8, 0xffd892b4, 0x001f5489, 0xffeda7b1, 0x00082718, 0xfffd779d, 0x00005e84,
+        0x150656f8, 0xf192e932, 0x0ac68e9b, 0xf838e6c9, 0x053a48fa, 0xfcd1eac3, 0x01adcc91, 0xff4d0f02, 0x00268f13, 0x00170a47, 0xffda6921, 0x001e7a33, 0xffedfab8, 0x00080ef7, 0xfffd7c7a, 0x00005d92,
+        0x140330a9, 0xf1f0c9c5, 0x0a9f7537, 0xf8456f65, 0x053b3400, 0xfccb1ed7, 0x01b5fd54, 0xff45db10, 0x002bd8fa, 0x0013b003, 0xffdc4007, 0x001d9e2a, 0xffee4f29, 0x0007f60f, 0xfffd81a4, 0x00005c8e,
+        0x1301efed, 0xf24f96b5, 0x0a7748c0, 0xf852df56, 0x053b7332, 0xfcc4c658, 0x01bde86f, 0xff3ecbea, 0x003112e0, 0x00105a72, 0xffde1726, 0x001cc091, 0xffeea4f2, 0x0007dc65, 0xfffd8719, 0x00005b7b,
+        0x1202a434, 0xf2af428c, 0x0a4e101f, 0xf861337c, 0x053b0791, 0xfcbee162, 0x01c58d50, 0xff37e23b, 0x00363c35, 0x000d09fc, 0xffdfee3f, 0x001be18a, 0xffeefc04, 0x0007c201, 0xfffd8cd7, 0x00005a58,
+        0x11055cb4, 0xf30fbfd7, 0x0a23d24e, 0xf870689f, 0x0539f231, 0xfcb97001, 0x01cceb6e, 0xff311ea4, 0x003b546b, 0x0009bf05, 0xffe1c511, 0x001b0138, 0xffef544e, 0x0007a6e9, 0xfffd92db, 0x00005927,
+        0x100a2864, 0xf371012c, 0x09f8965d, 0xf8807b70, 0x0538343a, 0xfcb47232, 0x01d4024c, 0xff2a81c4, 0x00405afa, 0x000679f2, 0xffe39b60, 0x001a1fbc, 0xffefadc0, 0x00078b24, 0xfffd9923, 0x000057e9,
+        0x0f111603, 0xf3d2f926, 0x09cc636e, 0xf8916889, 0x0535cee9, 0xfcafe7e2, 0x01dad175, 0xff240c2f, 0x00454f5d, 0x00033b23, 0xffe570ed, 0x00193d3a, 0xfff00849, 0x00076eba, 0xfffd9fac, 0x0000569d,
+        0x0e1a340d, 0xf4359a6a, 0x099f40b5, 0xf8a32c6e, 0x0532c38c, 0xfcabd0f2, 0x01e15880, 0xff1dbe77, 0x004a310f, 0x000002f9, 0xffe7457c, 0x001859d2, 0xfff063d9, 0x000751b0, 0xfffda675, 0x00005545,
+        0x0d2590c3, 0xf498d7a5, 0x09713575, 0xf8b5c38d, 0x052f1386, 0xfca82d32, 0x01e7970e, 0xff179926, 0x004eff94, 0xfffcd1d3, 0xffe918ce, 0x001775a7, 0xfff0c060, 0x0007340d, 0xfffdad79, 0x000053e2,
+        0x0c333a22, 0xf4fca390, 0x09424904, 0xf8c92a41, 0x052ac04c, 0xfca4fc64, 0x01ed8cc7, 0xff119cc0, 0x0053ba6e, 0xfff9a80d, 0xffeaeaab, 0x001690d9, 0xfff11dcd, 0x000715d9, 0xfffdb4b9, 0x00005274,
+};
+}
diff --git a/services/audioflinger/audio-resampler/resampler_filter_coefficients_10042011.h b/services/audioflinger/audio-resampler/resampler_filter_coefficients_10042011.h
deleted file mode 100644
index 8c6a899..0000000
--- a/services/audioflinger/audio-resampler/resampler_filter_coefficients_10042011.h
+++ /dev/null
@@ -1,2071 +0,0 @@
-/*
- * 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.
- */
-#include <stdlib.h>
-
-namespace android {
-
-const int32_t resampler_filter_coefficients_10042011[] = {
-2075076504,
-2074870219,
-2074269557,
-2073262841,
-2071862786,
-2070051926,
-2067849110,
-2065243563,
-2062248465,
-2058846262,
-2055055548,
-2050866069,
-2046291635,
-2041315273,
-2035955897,
-2030204167,
-2024074532,
-2017550518,
-2010651175,
-2003368165,
-1995716923,
-1987682243,
-1979283938,
-1970514752,
-1961390076,
-1951894181,
-1942045775,
-1931837943,
-1921286728,
-1910377736,
-1899130344,
-1887538902,
-1875619770,
-1863358864,
-1850775104,
-1837863721,
-1824641940,
-1811097380,
-1797249776,
-1783095528,
-1768651291,
-1753903742,
-1738870726,
-1723548746,
-1707955054,
-1692078274,
-1675937190,
-1659529795,
-1642873313,
-1625956474,
-1608797063,
-1591393772,
-1573764495,
-1555899971,
-1537818785,
-1519520784,
-1501022796,
-1482314240,
-1463411160,
-1444313227,
-1425037741,
-1405576492,
-1385946622,
-1366149382,
-1346201706,
-1326095287,
-1305845828,
-1285455045,
-1264940308,
-1244295464,
-1223536951,
-1202667530,
-1181703085,
-1160635857,
-1139479271,
-1118235636,
-1096921134,
-1075530585,
-1054078577,
-1032568949,
-1011017224,
-989417949,
-967783970,
-946119336,
-924439752,
-902741900,
-881039249,
-859336703,
-837648175,
-815968557,
-794308053,
-772670861,
-751070988,
-729505896,
-707986911,
-686519594,
-665117063,
-643776375,
-622506966,
-601314359,
-580211607,
-559197735,
-538282662,
-517472516,
-496778402,
-476197491,
-455736426,
-435400430,
-415200497,
-395136166,
-375215078,
-355443561,
-335831583,
-316378199,
-297089183,
-277970603,
-259032151,
-240274557,
-221703918,
-203326638,
-185150446,
-167174280,
-149401179,
-131836509,
-114487708,
-97355714,
-80444376,
-63759844,
-47308498,
-31090767,
-15108809,
--631654,
--16124677,
--31368536,
--46360908,
--61095999,
--75569692,
--89781876,
--103732869,
--117417959,
--130833461,
--143977734,
--156850513,
--169446641,
--181763420,
--193799687,
--205556559,
--217029409,
--228216068,
--239114500,
--249725794,
--260045485,
--270072966,
--279807538,
--289252364,
--298404017,
--307262396,
--315825804,
--324097052,
--332072584,
--339753137,
--347137398,
--354229167,
--361025450,
--367527527,
--373733643,
--379647676,
--385266970,
--390594013,
--395628043,
--400374357,
--404831217,
--409001628,
--412884357,
--416484569,
--419800667,
--422836272,
--425590412,
--428068799,
--430270358,
--432199204,
--433854305,
--435241518,
--436360214,
--437215297,
--437806349,
--438139967,
--438216244,
--438040549,
--437612462,
--436938620,
--436019455,
--434860689,
--433462013,
--431830172,
--429966019,
--427875663,
--425559045,
--423023081,
--420269108,
--417303590,
--414126712,
--410745497,
--407161766,
--403382348,
--399407797,
--395245305,
--390897144,
--386370224,
--381665062,
--376788564,
--371743279,
--366536360,
--361168782,
--355647611,
--349975843,
--344160581,
--338202705,
--332108851,
--325882239,
--319530194,
--313054213,
--306461164,
--299754744,
--292942106,
--286024571,
--279008412,
--271897436,
--264698872,
--257414606,
--250051032,
--242612307,
--235105276,
--227531464,
--219896432,
--212204319,
--204462036,
--196671840,
--188839540,
--180969700,
--173068854,
--165138974,
--157185076,
--149211664,
--141225186,
--133228184,
--125225726,
--117222535,
--109224429,
--101233397,
--93253442,
--85289050,
--77345947,
--69426846,
--61535955,
--53678071,
--45858404,
--38079310,
--30344121,
--22657427,
--15024209,
--7447312,
-69952,
-7522929,
-14907403,
-22221174,
-29462063,
-36625813,
-43708432,
-50707112,
-57619541,
-64441301,
-71168979,
-77800149,
-84333341,
-90764454,
-97090405,
-103308422,
-109417126,
-115412500,
-121292248,
-127054242,
-132698149,
-138220442,
-143619140,
-148891701,
-154037743,
-159053739,
-163938275,
-168689170,
-173306749,
-177787867,
-182131495,
-186335286,
-190399716,
-194321853,
-198101380,
-201736519,
-205228586,
-208575166,
-211776310,
-214830056,
-217737773,
-220497194,
-223108861,
-225571114,
-227885824,
-230051117,
-232067922,
-233934597,
-235653195,
-237222146,
-238642925,
-239914316,
-241038916,
-242015623,
-242846277,
-243529706,
-244068632,
-244462216,
-244712654,
-244818983,
-244784181,
-244607765,
-244292268,
-243836913,
-243244874,
-242516010,
-241653201,
-240655918,
-239527557,
-238268354,
-236881508,
-235366743,
-233727638,
-231964766,
-230081521,
-228077731,
-225956985,
-223720128,
-221370808,
-218909157,
-216338938,
-213661333,
-210880096,
-207995414,
-205010946,
-201928111,
-198750891,
-195479864,
-192118892,
-188669741,
-185136417,
-181519490,
-177822602,
-174047687,
-170198894,
-166277172,
-162286302,
-158228508,
-154107834,
-149925117,
-145683777,
-141386129,
-137036351,
-132635752,
-128187958,
-123695608,
-119162747,
-114590586,
-109982363,
-105340765,
-100669861,
-95971242,
-91248228,
-86503703,
-81741443,
-76962776,
-72170457,
-67367307,
-62557108,
-57741653,
-52923856,
-48106777,
-43293939,
-38486954,
-33688240,
-28900789,
-24128036,
-19371931,
-14634917,
-9920080,
-5230446,
-567618,
--4066612,
--8669336,
--13237622,
--17769457,
--22262939,
--26715017,
--31123091,
--35485370,
--39800471,
--44065504,
--48278050,
--52436071,
--56538223,
--60581634,
--64564356,
--68484740,
--72342096,
--76133828,
--79858170,
--83513191,
--87098163,
--90610469,
--94048704,
--97411174,
--100697628,
--103905676,
--107034151,
--110081227,
--113046738,
--115928406,
--118725517,
--121436605,
--124062060,
--126599916,
--129049689,
--131409767,
--133680567,
--135860199,
--137948517,
--139944122,
--141847799,
--143657917,
--145374591,
--146996418,
--148524310,
--149956828,
--151294475,
--152536152,
--153683178,
--154734431,
--155690674,
--156550825,
--157316298,
--157986149,
--158561410,
--159041176,
--159427086,
--159718451,
--159916548,
--160020582,
--160032332,
--159951337,
--159779143,
--159515147,
--159161329,
--158717499,
--158185432,
--157564678,
--156857347,
--156063474,
--155185008,
--154221601,
--153175426,
--152046727,
--150837654,
--149548057,
--148180246,
--146734704,
--145213706,
--143617181,
--141947434,
--140205146,
--138392776,
--136510514,
--134560819,
--132544622,
--130464445,
--128320507,
--126115178,
--123849532,
--121526221,
--119145720,
--116710510,
--114221879,
--111682456,
--109092675,
--106454825,
--103770291,
--101041818,
--98270160,
--95457755,
--92606223,
--89718259,
--86794577,
--83837395,
--80848397,
--77830331,
--74784179,
--71712237,
--68616354,
--65499123,
--62361390,
--59205117,
--56032152,
--52845129,
--49645224,
--46434530,
--43215084,
--39989380,
--36758496,
--33524217,
--30288565,
--27054001,
--23821849,
--20593926,
--17372347,
--14159320,
--10955954,
--7763647,
--4584416,
--1420431,
-1726908,
-4856115,
-7965053,
-11051765,
-14114989,
-17153587,
-20165512,
-23148908,
-26102328,
-29024647,
-31913805,
-34768253,
-37586799,
-40368755,
-43112232,
-45815788,
-48478013,
-51098183,
-53674369,
-56205373,
-58689945,
-61127693,
-63516834,
-65856322,
-68144797,
-70381920,
-72565965,
-74696201,
-76771518,
-78791971,
-80756044,
-82663155,
-84512076,
-86302872,
-88034067,
-89705316,
-91315544,
-92865092,
-94352657,
-95778072,
-97140241,
-98439585,
-99674922,
-100846363,
-101953029,
-102995643,
-103973242,
-104886114,
-105733366,
-106515783,
-107232507,
-107884029,
-108469586,
-108990146,
-109445031,
-109834907,
-110159075,
-110418598,
-110612956,
-110743021,
-110808247,
-110809873,
-110747573,
-110622392,
-110433872,
-110183341,
-109870630,
-109496925,
-109061853,
-108566821,
-108011814,
-107398172,
-106725651,
-105995758,
-105208649,
-104365776,
-103466973,
-102513780,
-101506507,
-100446746,
-99334505,
-98171433,
-96958015,
-95695913,
-94385172,
-93027406,
-91623219,
-90174378,
-88681103,
-87145094,
-85567114,
-83948941,
-82290788,
-80594255,
-78860190,
-77090470,
-75285527,
-73447070,
-71576122,
-69674546,
-67742761,
-65782346,
-63794384,
-61780793,
-59742181,
-57680191,
-55596034,
-53491542,
-51367245,
-49224577,
-47064766,
-44889688,
-42700109,
-40497558,
-38283410,
-36059459,
-33826411,
-31585598,
-29338397,
-27086596,
-24831081,
-22573215,
-20314456,
-18056430,
-15799888,
-13545910,
-11295904,
-9051483,
-6813612,
-4583444,
-2362482,
-152205,
--2046514,
--4232760,
--6405078,
--8562051,
--10702661,
--12825994,
--14930573,
--17015184,
--19078982,
--21121361,
--23140947,
--25136591,
--27107282,
--29052375,
--30970456,
--32860543,
--34721731,
--36553615,
--38354874,
--40124623,
--41861868,
--43566233,
--45236425,
--46871781,
--48471484,
--50035443,
--51562506,
--53052109,
--54503334,
--55916090,
--57289239,
--58622387,
--59914726,
--61166369,
--62376298,
--63544241,
--64669360,
--65751822,
--66790682,
--67785875,
--68736719,
--69643614,
--70505769,
--71323237,
--72095315,
--72822435,
--73503872,
--74139832,
--74729705,
--75274073,
--75772337,
--76224828,
--76630970,
--76991410,
--77305656,
--77574197,
--77796578,
--77973591,
--78104888,
--78191082,
--78231765,
--78227787,
--78178906,
--78085845,
--77948266,
--77767093,
--77542200,
--77274423,
--76963506,
--76610446,
--76215234,
--75778805,
--75300971,
--74782775,
--74224327,
--73626669,
--72989720,
--72314602,
--71601546,
--70851655,
--70064886,
--69242354,
--68384382,
--67492158,
--66565755,
--65606354,
--64614392,
--63591082,
--62536507,
--61451798,
--60337466,
--59194800,
--58024034,
--56826376,
--55602460,
--54353584,
--53079977,
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-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
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-0,
-0,
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-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
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-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-};
-}
diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
new file mode 100644
index 0000000..b082e8c
--- /dev/null
+++ b/services/audioflinger/test-resample.cpp
@@ -0,0 +1,274 @@
+/*
+ * 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.
+ */
+
+#include "AudioResampler.h"
+#include <media/AudioBufferProvider.h>
+#include <unistd.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <fcntl.h>
+#include <string.h>
+#include <sys/mman.h>
+#include <sys/stat.h>
+#include <errno.h>
+#include <time.h>
+#include <math.h>
+
+using namespace android;
+
+struct HeaderWav {
+    HeaderWav(size_t size, int nc, int sr, int bits) {
+        strncpy(RIFF, "RIFF", 4);
+        chunkSize = size + sizeof(HeaderWav);
+        strncpy(WAVE, "WAVE", 4);
+        strncpy(fmt,  "fmt ", 4);
+        fmtSize = 16;
+        audioFormat = 1;
+        numChannels = nc;
+        samplesRate = sr;
+        byteRate = sr * numChannels * (bits/8);
+        align = nc*(bits/8);
+        bitsPerSample = bits;
+        strncpy(data, "data", 4);
+        dataSize = size;
+    }
+
+    char RIFF[4];           // RIFF
+    uint32_t chunkSize;     // File size
+    char WAVE[4];        // WAVE
+    char fmt[4];            // fmt\0
+    uint32_t fmtSize;       // fmt size
+    uint16_t audioFormat;   // 1=PCM
+    uint16_t numChannels;   // num channels
+    uint32_t samplesRate;   // sample rate in hz
+    uint32_t byteRate;      // Bps
+    uint16_t align;         // 2=16-bit mono, 4=16-bit stereo
+    uint16_t bitsPerSample; // bits per sample
+    char data[4];           // "data"
+    uint32_t dataSize;      // size
+};
+
+static int usage(const char* name) {
+    fprintf(stderr,"Usage: %s [-p] [-h] [-s] [-q {dq|lq|mq|hq|vhq}] [-i input-sample-rate] "
+                   "[-o output-sample-rate] [<input-file>] <output-file>\n", name);
+    fprintf(stderr,"    -p    enable profiling\n");
+    fprintf(stderr,"    -h    create wav file\n");
+    fprintf(stderr,"    -s    stereo\n");
+    fprintf(stderr,"    -q    resampler quality\n");
+    fprintf(stderr,"              dq  : default quality\n");
+    fprintf(stderr,"              lq  : low quality\n");
+    fprintf(stderr,"              mq  : medium quality\n");
+    fprintf(stderr,"              hq  : high quality\n");
+    fprintf(stderr,"              vhq : very high quality\n");
+    fprintf(stderr,"    -i    input file sample rate\n");
+    fprintf(stderr,"    -o    output file sample rate\n");
+    return -1;
+}
+
+int main(int argc, char* argv[]) {
+
+    const char* const progname = argv[0];
+    bool profiling = false;
+    bool writeHeader = false;
+    int channels = 1;
+    int input_freq = 0;
+    int output_freq = 0;
+    AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY;
+
+    int ch;
+    while ((ch = getopt(argc, argv, "phsq:i:o:")) != -1) {
+        switch (ch) {
+        case 'p':
+            profiling = true;
+            break;
+        case 'h':
+            writeHeader = true;
+            break;
+        case 's':
+            channels = 2;
+            break;
+        case 'q':
+            if (!strcmp(optarg, "dq"))
+                quality = AudioResampler::DEFAULT_QUALITY;
+            else if (!strcmp(optarg, "lq"))
+                quality = AudioResampler::LOW_QUALITY;
+            else if (!strcmp(optarg, "mq"))
+                quality = AudioResampler::MED_QUALITY;
+            else if (!strcmp(optarg, "hq"))
+                quality = AudioResampler::HIGH_QUALITY;
+            else if (!strcmp(optarg, "vhq"))
+                quality = AudioResampler::VERY_HIGH_QUALITY;
+            else {
+                usage(progname);
+                return -1;
+            }
+            break;
+        case 'i':
+            input_freq = atoi(optarg);
+            break;
+        case 'o':
+            output_freq = atoi(optarg);
+            break;
+        case '?':
+        default:
+            usage(progname);
+            return -1;
+        }
+    }
+    argc -= optind;
+    argv += optind;
+
+    const char* file_in = NULL;
+    const char* file_out = NULL;
+    if (argc == 1) {
+        file_out = argv[0];
+    } else if (argc == 2) {
+        file_in = argv[0];
+        file_out = argv[1];
+    } else {
+        usage(progname);
+        return -1;
+    }
+
+    // ----------------------------------------------------------
+
+    size_t input_size;
+    void* input_vaddr;
+    if (argc == 2) {
+        struct stat st;
+        if (stat(file_in, &st) < 0) {
+            fprintf(stderr, "stat: %s\n", strerror(errno));
+            return -1;
+        }
+
+        int input_fd = open(file_in, O_RDONLY);
+        if (input_fd < 0) {
+            fprintf(stderr, "open: %s\n", strerror(errno));
+            return -1;
+        }
+
+        input_size = st.st_size;
+        input_vaddr = mmap(0, input_size, PROT_READ, MAP_PRIVATE, input_fd, 0);
+        if (input_vaddr == MAP_FAILED ) {
+            fprintf(stderr, "mmap: %s\n", strerror(errno));
+            return -1;
+        }
+    } else {
+        double k = 1000; // Hz / s
+        double time = (input_freq / 2) / k;
+        size_t input_frames = size_t(input_freq * time);
+        input_size = channels * sizeof(int16_t) * input_frames;
+        input_vaddr = malloc(input_size);
+        int16_t* in = (int16_t*)input_vaddr;
+        for (size_t i=0 ; i<input_frames ; i++) {
+            double t = double(i) / input_freq;
+            double y = sin(M_PI * k * t * t);
+            int16_t yi = floor(y * 32767.0 + 0.5);
+            for (size_t j=0 ; j<(size_t)channels ; j++) {
+                in[i*channels + j] = yi / (1+j);
+            }
+        }
+    }
+
+    // ----------------------------------------------------------
+
+    class Provider: public AudioBufferProvider {
+        int16_t* mAddr;
+        size_t mNumFrames;
+    public:
+        Provider(const void* addr, size_t size, int channels) {
+            mAddr = (int16_t*) addr;
+            mNumFrames = size / (channels*sizeof(int16_t));
+        }
+        virtual status_t getNextBuffer(Buffer* buffer,
+                int64_t pts = kInvalidPTS) {
+            buffer->frameCount = mNumFrames;
+            buffer->i16 = mAddr;
+            return NO_ERROR;
+        }
+        virtual void releaseBuffer(Buffer* buffer) {
+        }
+    } provider(input_vaddr, input_size, channels);
+
+    size_t input_frames = input_size / (channels * sizeof(int16_t));
+    size_t output_size = 2 * 4 * ((int64_t) input_frames * output_freq) / input_freq;
+    output_size &= ~7; // always stereo, 32-bits
+
+    void* output_vaddr = malloc(output_size);
+
+    if (profiling) {
+        AudioResampler* resampler = AudioResampler::create(16, channels,
+                output_freq, quality);
+
+        size_t out_frames = output_size/8;
+        resampler->setSampleRate(input_freq);
+        resampler->setVolume(0x1000, 0x1000);
+
+        memset(output_vaddr, 0, output_size);
+        timespec start, end;
+        clock_gettime(CLOCK_MONOTONIC_HR, &start);
+        resampler->resample((int*) output_vaddr, out_frames, &provider);
+        resampler->resample((int*) output_vaddr, out_frames, &provider);
+        resampler->resample((int*) output_vaddr, out_frames, &provider);
+        resampler->resample((int*) output_vaddr, out_frames, &provider);
+        clock_gettime(CLOCK_MONOTONIC_HR, &end);
+        int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
+        int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
+        int64_t time = (end_ns - start_ns)/4;
+        printf("%f Mspl/s\n", out_frames/(time/1e9)/1e6);
+
+        delete resampler;
+    }
+
+    AudioResampler* resampler = AudioResampler::create(16, channels,
+            output_freq, quality);
+    size_t out_frames = output_size/8;
+    resampler->setSampleRate(input_freq);
+    resampler->setVolume(0x1000, 0x1000);
+
+    memset(output_vaddr, 0, output_size);
+    resampler->resample((int*) output_vaddr, out_frames, &provider);
+
+    // down-mix (we just truncate and keep the left channel)
+    int32_t* out = (int32_t*) output_vaddr;
+    int16_t* convert = (int16_t*) malloc(out_frames * channels * sizeof(int16_t));
+    for (size_t i = 0; i < out_frames; i++) {
+        for (int j=0 ; j<channels ; j++) {
+            int32_t s = out[i * 2 + j] >> 12;
+            if (s > 32767)       s =  32767;
+            else if (s < -32768) s = -32768;
+            convert[i * channels + j] = int16_t(s);
+        }
+    }
+
+    // write output to disk
+    int output_fd = open(file_out, O_WRONLY | O_CREAT | O_TRUNC,
+            S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH);
+    if (output_fd < 0) {
+        fprintf(stderr, "open: %s\n", strerror(errno));
+        return -1;
+    }
+
+    if (writeHeader) {
+        HeaderWav wav(out_frames * channels * sizeof(int16_t), channels, output_freq, 16);
+        write(output_fd, &wav, sizeof(wav));
+    }
+
+    write(output_fd, convert, out_frames * channels * sizeof(int16_t));
+    close(output_fd);
+
+    return 0;
+}
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 e59a240..9627416 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);
@@ -1062,7 +1057,7 @@
     return OK;
 }
 
-status_t Camera2Client::takePicture(int msgType) {
+status_t Camera2Client::takePicture(int /*msgType*/) {
     ATRACE_CALL();
     Mutex::Autolock icl(mICameraLock);
     status_t res;
@@ -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 fe4abc0..b228faf 100644
--- a/services/camera/libcameraservice/camera2/CaptureSequencer.cpp
+++ b/services/camera/libcameraservice/camera2/CaptureSequencer.cpp
@@ -128,7 +128,7 @@
 }
 
 
-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";
@@ -182,7 +182,6 @@
 };
 
 bool CaptureSequencer::threadLoop() {
-    status_t res;
 
     sp<Camera2Client> client = mClient.promote();
     if (client == 0) return false;
@@ -213,7 +212,8 @@
     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) {
@@ -350,13 +350,13 @@
 }
 
 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;
 }
@@ -378,7 +378,7 @@
 }
 
 CaptureSequencer::CaptureState CaptureSequencer::manageStandardPrecaptureWait(
-        sp<Camera2Client> &client) {
+        sp<Camera2Client> &/*client*/) {
     status_t res;
     ATRACE_CALL();
     Mutex::Autolock l(mInputMutex);
@@ -578,7 +578,7 @@
 }
 
 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");
diff --git a/tools/resampler_tools/fir.cpp b/tools/resampler_tools/fir.cpp
index 377814f..cc3d509 100644
--- a/tools/resampler_tools/fir.cpp
+++ b/tools/resampler_tools/fir.cpp
@@ -16,6 +16,9 @@
 
 #include <math.h>
 #include <stdio.h>
+#include <unistd.h>
+#include <stdlib.h>
+#include <string.h>
 
 static double sinc(double x) {
     if (fabs(x) == 0.0f) return 1.0f;
@@ -34,44 +37,82 @@
         y=x/3.75;
         y*=y;
         ans=1.0+y*(3.5156229+y*(3.0899424+y*(1.2067492
-            +y*(0.2659732+y*(0.360768e-1+y*0.45813e-2)))));
+                +y*(0.2659732+y*(0.360768e-1+y*0.45813e-2)))));
     } else {
         y=3.75/ax;
         ans=(exp(ax)/sqrt(ax))*(0.39894228+y*(0.1328592e-1
-            +y*(0.225319e-2+y*(-0.157565e-2+y*(0.916281e-2
-            +y*(-0.2057706e-1+y*(0.2635537e-1+y*(-0.1647633e-1
-            +y*0.392377e-2))))))));
+                +y*(0.225319e-2+y*(-0.157565e-2+y*(0.916281e-2
+                        +y*(-0.2057706e-1+y*(0.2635537e-1+y*(-0.1647633e-1
+                                +y*0.392377e-2))))))));
     }
     return ans;
 }
 
-static double kaiser(int k, int N, double alpha) {
+static double kaiser(int k, int N, double beta) {
     if (k < 0 || k > N)
         return 0;
-    return I0(M_PI*alpha * sqrt(1.0 - sqr((2.0*k)/N - 1.0))) / I0(M_PI*alpha);
+    return I0(beta * sqrt(1.0 - sqr((2.0*k)/N - 1.0))) / I0(beta);
+}
+
+
+static void usage(char* name) {
+    fprintf(stderr,
+            "usage: %s [-h] [-d] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings] [-f {float|fixed}] [-b beta] [-v dBFS] [-l lerp]\n"
+            "       %s [-h] [-d] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings] [-f {float|fixed}] [-b beta] [-v dBFS] -p M/N\n"
+            "    -h    this help message\n"
+            "    -d    debug, print comma-separated coefficient table\n"
+            "    -p    generate poly-phase filter coefficients, with sample increment M/N\n"
+            "    -s    sample rate (48000)\n"
+            "    -c    cut-off frequency (20478)\n"
+            "    -n    number of zero-crossings on one side (8)\n"
+            "    -l    number of lerping bits (4)\n"
+            "    -f    output format, can be fixed-point or floating-point (fixed)\n"
+            "    -b    kaiser window parameter beta (7.865 [-80dB])\n"
+            "    -v    attenuation in dBFS (0)\n",
+            name, name
+    );
+    exit(0);
 }
 
 int main(int argc, char** argv)
 {
     // nc is the number of bits to store the coefficients
-    int nc = 32;
+    const int nc = 32;
 
-    // ni is the minimum number of bits needed for interpolation
-    // (not used for generating the coefficients)
-    const int ni = nc / 2;
+    bool polyphase = false;
+    unsigned int polyM = 160;
+    unsigned int polyN = 147;
+    bool debug = false;
+    double Fs = 48000;
+    double Fc = 20478;
+    double atten = 1;
+    int format = 0;
 
-    // cut off frequency ratio Fc/Fs
-    // The bigger the stop-band, the less coefficients we'll need.
-    double Fcr = 20000.0 / 48000.0;
 
-    // nzc is the number of zero-crossing on one half of the filter
-    int nzc = 8;
-    
-    // alpha parameter of the kaiser window
-    // Larger numbers reduce ripples in the rejection band but increase
-    // the width of the transition band. 
-    // the table below gives some value of alpha for a given
-    // stop-band attenuation.
+    // in order to keep the errors associated with the linear
+    // interpolation of the coefficients below the quantization error
+    // we must satisfy:
+    //   2^nz >= 2^(nc/2)
+    //
+    // for 16 bit coefficients that would be 256
+    //
+    // note that increasing nz only increases memory requirements,
+    // but doesn't increase the amount of computation to do.
+    //
+    //
+    // see:
+    // Smith, J.O. Digital Audio Resampling Home Page
+    // https://ccrma.stanford.edu/~jos/resample/, 2011-03-29
+    //
+    int nz = 4;
+
+    //         | 0.1102*(A - 8.7)                         A > 50
+    //  beta = | 0.5842*(A - 21)^0.4 + 0.07886*(A - 21)   21 <= A <= 50
+    //         | 0                                        A < 21
+    //   with A is the desired stop-band attenuation in dBFS
+    //
+    // for eg:
+    //
     //    30 dB    2.210
     //    40 dB    3.384
     //    50 dB    4.538
@@ -80,42 +121,161 @@
     //    80 dB    7.865
     //    90 dB    8.960
     //   100 dB   10.056
-    double alpha = 7.865;	// -80dB stop-band attenuation
-    
-    // 2^nz is the number coefficients per zero-crossing
-    // (int theory this should be 1<<(nc/2))
-    const int nz = 4;
+    double beta = 7.865;
 
-    // total number of coefficients
-    const int N = (1 << nz) * nzc;
+
+    // 2*nzc = (A - 8) / (2.285 * dw)
+    //      with dw the transition width = 2*pi*dF/Fs
+    //
+    int nzc = 8;
+
+    //
+    // Example:
+    // 44.1 KHz to 48 KHz resampling
+    // 100 dB rejection above 28 KHz
+    //   (the spectrum will fold around 24 KHz and we want 100 dB rejection
+    //    at the point where the folding reaches 20 KHz)
+    //  ...___|_____
+    //        |     \|
+    //        | ____/|\____
+    //        |/alias|     \
+    //  ------/------+------\---------> KHz
+    //       20     24     28
+
+    // Transition band 8 KHz, or dw = 1.0472
+    //
+    // beta = 10.056
+    // nzc  = 20
+    //
+
+    int ch;
+    while ((ch = getopt(argc, argv, ":hds:c:n:f:l:b:p:v:")) != -1) {
+        switch (ch) {
+            case 'd':
+                debug = true;
+                break;
+            case 'p':
+                if (sscanf(optarg, "%u/%u", &polyM, &polyN) != 2) {
+                    usage(argv[0]);
+                }
+                polyphase = true;
+                break;
+            case 's':
+                Fs = atof(optarg);
+                break;
+            case 'c':
+                Fc = atof(optarg);
+                break;
+            case 'n':
+                nzc = atoi(optarg);
+                break;
+            case 'l':
+                nz = atoi(optarg);
+                break;
+            case 'f':
+                if (!strcmp(optarg,"fixed")) format = 0;
+                else if (!strcmp(optarg,"float")) format = 1;
+                else usage(argv[0]);
+                break;
+            case 'b':
+                beta = atof(optarg);
+                break;
+            case 'v':
+                atten = pow(10, -fabs(atof(optarg))*0.05 );
+                break;
+            case 'h':
+            default:
+                usage(argv[0]);
+                break;
+        }
+    }
+
+    // cut off frequency ratio Fc/Fs
+    double Fcr = Fc / Fs;
+
+
+    // total number of coefficients (one side)
+    const int M = (1 << nz);
+    const int N = M * nzc;
 
     // generate the right half of the filter
-    printf("const int32_t RESAMPLE_FIR_SIZE           = %d;\n", N);
-    printf("const int32_t RESAMPLE_FIR_NUM_COEF       = %d;\n", nzc);
-    printf("const int32_t RESAMPLE_FIR_COEF_BITS      = %d;\n", nc);
-    printf("const int32_t RESAMPLE_FIR_LERP_FRAC_BITS = %d;\n", ni);
-    printf("const int32_t RESAMPLE_FIR_LERP_INT_BITS  = %d;\n", nz);
-    printf("\n");
-    printf("static int16_t resampleFIR[%d] = {", N);
-    for (int i=0 ; i<N ; i++)
-    {
-        double x = (2.0 * M_PI * i * Fcr) / (1 << nz);
-        double y = kaiser(i+N, 2*N, alpha) * sinc(x);
-
-        long yi = floor(y * ((1ULL<<(nc-1))) + 0.5);
-        if (yi >= (1LL<<(nc-1))) yi = (1LL<<(nc-1))-1;        
-
-        if ((i % (1 << 4)) == 0) printf("\n    ");
-        if (nc > 16)
-        	printf("0x%08x, ", int(yi));
-        else 
-        	printf("0x%04x, ", int(yi)&0xFFFF);
+    if (!debug) {
+        printf("// cmd-line: ");
+        for (int i=1 ; i<argc ; i++) {
+            printf("%s ", argv[i]);
+        }
+        printf("\n");
+        if (!polyphase) {
+            printf("const int32_t RESAMPLE_FIR_SIZE           = %d;\n", N);
+            printf("const int32_t RESAMPLE_FIR_LERP_INT_BITS  = %d;\n", nz);
+            printf("const int32_t RESAMPLE_FIR_NUM_COEF       = %d;\n", nzc);
+        } else {
+            printf("const int32_t RESAMPLE_FIR_SIZE           = %d;\n", 2*nzc*polyN);
+            printf("const int32_t RESAMPLE_FIR_NUM_COEF       = %d;\n", 2*nzc);
+        }
+        if (!format) {
+            printf("const int32_t RESAMPLE_FIR_COEF_BITS      = %d;\n", nc);
+        }
+        printf("\n");
+        printf("static %s resampleFIR[] = {", !format ? "int32_t" : "float");
     }
-    printf("\n};\n");
-    return 0;
- }
 
-// http://www.dsptutor.freeuk.com/KaiserFilterDesign/KaiserFilterDesign.html
+    if (!polyphase) {
+        for (int i=0 ; i<=M ; i++) { // an extra set of coefs for interpolation
+            for (int j=0 ; j<nzc ; j++) {
+                int ix = j*M + i;
+                double x = (2.0 * M_PI * ix * Fcr) / (1 << nz);
+                double y = kaiser(ix+N, 2*N, beta) * sinc(x) * 2.0 * Fcr;
+                y *= atten;
+
+                if (!debug) {
+                    if (j == 0)
+                        printf("\n    ");
+                }
+
+                if (!format) {
+                    int64_t yi = floor(y * ((1ULL<<(nc-1))) + 0.5);
+                    if (yi >= (1LL<<(nc-1))) yi = (1LL<<(nc-1))-1;
+                    printf("0x%08x, ", int32_t(yi));
+                } else {
+                    printf("%.9g%s ", y, debug ? "," : "f,");
+                }
+            }
+        }
+    } else {
+        for (int j=0 ; j<polyN ; j++) {
+            // calculate the phase
+            double p = ((polyM*j) % polyN) / double(polyN);
+            if (!debug) printf("\n    ");
+            else        printf("\n");
+            // generate a FIR per phase
+            for (int i=-nzc ; i<nzc ; i++) {
+                double x = 2.0 * M_PI * Fcr * (i + p);
+                double y = kaiser(i+N, 2*N, beta) * sinc(x) * 2.0 * Fcr;;
+                y *= atten;
+                if (!format) {
+                    int64_t yi = floor(y * ((1ULL<<(nc-1))) + 0.5);
+                    if (yi >= (1LL<<(nc-1))) yi = (1LL<<(nc-1))-1;
+                    printf("0x%08x", int32_t(yi));
+                } else {
+                    printf("%.9g%s", y, debug ? "" : "f");
+                }
+
+                if (debug && (i==nzc-1)) {
+                } else {
+                    printf(", ");
+                }
+            }
+        }
+    }
+
+    if (!debug) {
+        printf("\n};");
+    }
+    printf("\n");
+    return 0;
+}
+
 // http://www.csee.umbc.edu/help/sound/AFsp-V2R1/html/audio/ResampAudio.html
 
- 
+