diff --git a/camera/CameraParameters.cpp b/camera/CameraParameters.cpp
index e5e4e90..3dbf75e 100644
--- a/camera/CameraParameters.cpp
+++ b/camera/CameraParameters.cpp
@@ -526,7 +526,7 @@
         !strcmp(format, PIXEL_FORMAT_RGBA8888) ?
             HAL_PIXEL_FORMAT_RGBA_8888 :    // RGB8888
         !strcmp(format, PIXEL_FORMAT_BAYER_RGGB) ?
-            HAL_PIXEL_FORMAT_RAW_SENSOR :   // Raw sensor data
+            HAL_PIXEL_FORMAT_RAW16 :   // Raw sensor data
         -1;
 }
 
diff --git a/camera/ICameraService.cpp b/camera/ICameraService.cpp
index fc3e437..a75cb48 100644
--- a/camera/ICameraService.cpp
+++ b/camera/ICameraService.cpp
@@ -209,6 +209,20 @@
         return status;
     }
 
+    virtual status_t setTorchMode(const String16& cameraId, bool enabled,
+            const sp<IBinder>& clientBinder)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(ICameraService::getInterfaceDescriptor());
+        data.writeString16(cameraId);
+        data.writeInt32(enabled ? 1 : 0);
+        data.writeStrongBinder(clientBinder);
+        remote()->transact(BnCameraService::SET_TORCH_MODE, data, &reply);
+
+        if (readExceptionCode(reply)) return -EPROTO;
+        return reply.readInt32();
+    }
+
     // connect to camera service (pro client)
     virtual status_t connectPro(const sp<IProCameraCallbacks>& cameraCb, int cameraId,
                                 const String16 &clientPackageName, int clientUid,
@@ -490,6 +504,16 @@
             }
             return NO_ERROR;
         } break;
+        case SET_TORCH_MODE: {
+            CHECK_INTERFACE(ICameraService, data, reply);
+            String16 cameraId = data.readString16();
+            bool enabled = data.readInt32() != 0 ? true : false;
+            const sp<IBinder> clientBinder = data.readStrongBinder();
+            status_t status = setTorchMode(cameraId, enabled, clientBinder);
+            reply->writeNoException();
+            reply->writeInt32(status);
+            return NO_ERROR;
+        } break;
         default:
             return BBinder::onTransact(code, data, reply, flags);
     }
diff --git a/camera/ICameraServiceListener.cpp b/camera/ICameraServiceListener.cpp
index b2f1729..90a8bc2 100644
--- a/camera/ICameraServiceListener.cpp
+++ b/camera/ICameraServiceListener.cpp
@@ -29,6 +29,7 @@
 namespace {
     enum {
         STATUS_CHANGED = IBinder::FIRST_CALL_TRANSACTION,
+        TORCH_STATUS_CHANGED,
     };
 }; // namespace anonymous
 
@@ -54,8 +55,21 @@
                            data,
                            &reply,
                            IBinder::FLAG_ONEWAY);
+    }
 
-        reply.readExceptionCode();
+    virtual void onTorchStatusChanged(TorchStatus status, const String16 &cameraId)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(
+                              ICameraServiceListener::getInterfaceDescriptor());
+
+        data.writeInt32(static_cast<int32_t>(status));
+        data.writeString16(cameraId);
+
+        remote()->transact(TORCH_STATUS_CHANGED,
+                           data,
+                           &reply,
+                           IBinder::FLAG_ONEWAY);
     }
 };
 
@@ -75,7 +89,16 @@
             int32_t cameraId = data.readInt32();
 
             onStatusChanged(status, cameraId);
-            reply->writeNoException();
+
+            return NO_ERROR;
+        } break;
+        case TORCH_STATUS_CHANGED: {
+            CHECK_INTERFACE(ICameraServiceListener, data, reply);
+
+            TorchStatus status = static_cast<TorchStatus>(data.readInt32());
+            String16 cameraId = data.readString16();
+
+            onTorchStatusChanged(status, cameraId);
 
             return NO_ERROR;
         } break;
diff --git a/camera/tests/ProCameraTests.cpp b/camera/tests/ProCameraTests.cpp
index 1f5867a..24b2327 100644
--- a/camera/tests/ProCameraTests.cpp
+++ b/camera/tests/ProCameraTests.cpp
@@ -89,6 +89,12 @@
         mCondition.broadcast();
     }
 
+    void onTorchStatusChanged(TorchStatus status, const String16& cameraId) {
+        dout << "On torch status changed: 0x" << std::hex
+             << (unsigned int) status << " cameraId " << cameraId.string()
+             << std::endl;
+    }
+
     status_t waitForStatusChange(Status& newStatus) {
         Mutex::Autolock al(mMutex);
 
@@ -469,7 +475,7 @@
         CMP_STR(NV16, YCbCr_422_SP);
         CMP_STR(NV21, YCrCb_420_SP);
         CMP_STR(YUY2, YCbCr_422_I);
-        CMP_STR(RAW,  RAW_SENSOR);
+        CMP_STR(RAW,  RAW16);
         CMP_STR(RGBA, RGBA_8888);
 
         std::cerr << "Unknown format string " << str << std::endl;
diff --git a/cmds/screenrecord/screenrecord.cpp b/cmds/screenrecord/screenrecord.cpp
index 02df1d2..36a7e73 100644
--- a/cmds/screenrecord/screenrecord.cpp
+++ b/cmds/screenrecord/screenrecord.cpp
@@ -23,7 +23,10 @@
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
+#include <sys/stat.h>
+#include <sys/types.h>
 #include <sys/wait.h>
+
 #include <termios.h>
 #include <unistd.h>
 
@@ -637,7 +640,13 @@
         case FORMAT_MP4: {
             // Configure muxer.  We have to wait for the CSD blob from the encoder
             // before we can start it.
-            muxer = new MediaMuxer(fileName, MediaMuxer::OUTPUT_FORMAT_MPEG_4);
+            int fd = open(fileName, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+            if (fd < 0) {
+                fprintf(stderr, "ERROR: couldn't open file\n");
+                abort();
+            }
+            muxer = new MediaMuxer(fd, MediaMuxer::OUTPUT_FORMAT_MPEG_4);
+            close(fd);
             if (gRotate) {
                 muxer->setOrientationHint(90);  // TODO: does this do anything?
             }
diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk
index 561ce02..0e3bc68 100644
--- a/cmds/stagefright/Android.mk
+++ b/cmds/stagefright/Android.mk
@@ -169,6 +169,48 @@
 
 include $(CLEAR_VARS)
 
+LOCAL_SRC_FILES:= \
+	filters/argbtorgba.rs \
+	filters/nightvision.rs \
+	filters/saturation.rs \
+	mediafilter.cpp \
+
+LOCAL_SHARED_LIBRARIES := \
+	libstagefright \
+	liblog \
+	libutils \
+	libbinder \
+	libstagefright_foundation \
+	libmedia \
+	libgui \
+	libcutils \
+	libui \
+	libRScpp \
+
+LOCAL_C_INCLUDES:= \
+	$(TOP)/frameworks/av/media/libstagefright \
+	$(TOP)/frameworks/native/include/media/openmax \
+	$(TOP)/frameworks/rs/cpp \
+	$(TOP)/frameworks/rs \
+
+intermediates := $(call intermediates-dir-for,STATIC_LIBRARIES,libRS,TARGET,)
+LOCAL_C_INCLUDES += $(intermediates)
+
+LOCAL_STATIC_LIBRARIES:= \
+	libstagefright_mediafilter
+
+LOCAL_CFLAGS += -Wno-multichar
+
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_MODULE:= mediafilter
+
+include $(BUILD_EXECUTABLE)
+
+################################################################################
+
+include $(CLEAR_VARS)
+
 LOCAL_SRC_FILES:=               \
         muxer.cpp            \
 
diff --git a/cmds/stagefright/audioloop.cpp b/cmds/stagefright/audioloop.cpp
index 96073f1..7b0de24 100644
--- a/cmds/stagefright/audioloop.cpp
+++ b/cmds/stagefright/audioloop.cpp
@@ -14,6 +14,10 @@
  * limitations under the License.
  */
 
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+
 #include <binder/ProcessState.h>
 #include <media/mediarecorder.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -109,7 +113,12 @@
 
     if (fileOut != NULL) {
         // target file specified, write encoded AMR output
-        sp<AMRWriter> writer = new AMRWriter(fileOut);
+        int fd = open(fileOut, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+        if (fd < 0) {
+            return 1;
+        }
+        sp<AMRWriter> writer = new AMRWriter(fd);
+        close(fd);
         writer->addSource(encoder);
         writer->start();
         sleep(duration);
diff --git a/cmds/stagefright/codec.cpp b/cmds/stagefright/codec.cpp
index fd02bcc..d987250 100644
--- a/cmds/stagefright/codec.cpp
+++ b/cmds/stagefright/codec.cpp
@@ -45,9 +45,10 @@
     fprintf(stderr, "usage: %s [-a] use audio\n"
                     "\t\t[-v] use video\n"
                     "\t\t[-p] playback\n"
-                    "\t\t[-S] allocate buffers from a surface\n",
+                    "\t\t[-S] allocate buffers from a surface\n"
+                    "\t\t[-R] render output to surface (enables -S)\n"
+                    "\t\t[-T] use render timestamps (enables -R)\n",
                     me);
-
     exit(1);
 }
 
@@ -71,7 +72,9 @@
         const char *path,
         bool useAudio,
         bool useVideo,
-        const android::sp<android::Surface> &surface) {
+        const android::sp<android::Surface> &surface,
+        bool renderSurface,
+        bool useTimestamp) {
     using namespace android;
 
     static int64_t kTimeout = 500ll;
@@ -136,6 +139,7 @@
     CHECK(!stateByTrack.isEmpty());
 
     int64_t startTimeUs = ALooper::GetNowUs();
+    int64_t startTimeRender = -1;
 
     for (size_t i = 0; i < stateByTrack.size(); ++i) {
         CodecState *state = &stateByTrack.editValueAt(i);
@@ -260,7 +264,23 @@
                 ++state->mNumBuffersDecoded;
                 state->mNumBytesDecoded += size;
 
-                err = state->mCodec->releaseOutputBuffer(index);
+                if (surface == NULL || !renderSurface) {
+                    err = state->mCodec->releaseOutputBuffer(index);
+                } else if (useTimestamp) {
+                    if (startTimeRender == -1) {
+                        // begin rendering 2 vsyncs (~33ms) after first decode
+                        startTimeRender =
+                                systemTime(SYSTEM_TIME_MONOTONIC) + 33000000
+                                - (presentationTimeUs * 1000);
+                    }
+                    presentationTimeUs =
+                            (presentationTimeUs * 1000) + startTimeRender;
+                    err = state->mCodec->renderOutputBufferAndRelease(
+                            index, presentationTimeUs);
+                } else {
+                    err = state->mCodec->renderOutputBufferAndRelease(index);
+                }
+
                 CHECK_EQ(err, (status_t)OK);
 
                 if (flags & MediaCodec::BUFFER_FLAG_EOS) {
@@ -320,34 +340,42 @@
     bool useVideo = false;
     bool playback = false;
     bool useSurface = false;
+    bool renderSurface = false;
+    bool useTimestamp = false;
 
     int res;
-    while ((res = getopt(argc, argv, "havpSD")) >= 0) {
+    while ((res = getopt(argc, argv, "havpSDRT")) >= 0) {
         switch (res) {
             case 'a':
             {
                 useAudio = true;
                 break;
             }
-
             case 'v':
             {
                 useVideo = true;
                 break;
             }
-
             case 'p':
             {
                 playback = true;
                 break;
             }
-
+            case 'T':
+            {
+                useTimestamp = true;
+            }
+            // fall through
+            case 'R':
+            {
+                renderSurface = true;
+            }
+            // fall through
             case 'S':
             {
                 useSurface = true;
                 break;
             }
-
             case '?':
             case 'h':
             default:
@@ -422,7 +450,8 @@
         player->stop();
         player->reset();
     } else {
-        decode(looper, argv[0], useAudio, useVideo, surface);
+        decode(looper, argv[0], useAudio, useVideo, surface, renderSurface,
+                useTimestamp);
     }
 
     if (playback || (useSurface && useVideo)) {
diff --git a/include/media/nbaio/roundup.h b/cmds/stagefright/filters/argbtorgba.rs
similarity index 65%
rename from include/media/nbaio/roundup.h
rename to cmds/stagefright/filters/argbtorgba.rs
index 4c3cc25..229ff8c 100644
--- a/include/media/nbaio/roundup.h
+++ b/cmds/stagefright/filters/argbtorgba.rs
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2012 The Android Open Source Project
+ * Copyright (C) 2014 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.
@@ -14,18 +14,13 @@
  * limitations under the License.
  */
 
-#ifndef ROUNDUP_H
-#define ROUNDUP_H
+#pragma version(1)
+#pragma rs java_package_name(com.android.rs.cppbasic)
+#pragma rs_fp_relaxed
 
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-// Round up to the next highest power of 2
-unsigned roundup(unsigned v);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif  // ROUNDUP_H
+void root(const uchar4 *v_in, uchar4 *v_out) {
+    v_out->x = v_in->y;
+    v_out->y = v_in->z;
+    v_out->z = v_in->w;
+    v_out->w = v_in->x;
+}
\ No newline at end of file
diff --git a/cmds/stagefright/filters/nightvision.rs b/cmds/stagefright/filters/nightvision.rs
new file mode 100644
index 0000000..f61413c
--- /dev/null
+++ b/cmds/stagefright/filters/nightvision.rs
@@ -0,0 +1,38 @@
+/*
+ * Copyright (C) 2014 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.
+ */
+
+#pragma version(1)
+#pragma rs java_package_name(com.android.rs.cppbasic)
+#pragma rs_fp_relaxed
+
+const static float3 gMonoMult = {0.299f, 0.587f, 0.114f};
+const static float3 gNightVisionMult = {0.5f, 1.f, 0.5f};
+
+// calculates luminance of pixel, then biases color balance toward green
+void root(const uchar4 *v_in, uchar4 *v_out) {
+    v_out->x = v_in->x; // don't modify A
+
+    // get RGB, scale 0-255 uchar to 0-1.0 float
+    float3 rgb = {v_in->y * 0.003921569f, v_in->z * 0.003921569f,
+            v_in->w * 0.003921569f};
+
+    // apply filter
+    float3 result = dot(rgb, gMonoMult) * gNightVisionMult;
+
+    v_out->y = (uchar)clamp((result.r * 255.f + 0.5f), 0.f, 255.f);
+    v_out->z = (uchar)clamp((result.g * 255.f + 0.5f), 0.f, 255.f);
+    v_out->w = (uchar)clamp((result.b * 255.f + 0.5f), 0.f, 255.f);
+}
diff --git a/cmds/stagefright/filters/saturation.rs b/cmds/stagefright/filters/saturation.rs
new file mode 100644
index 0000000..1de9dd8
--- /dev/null
+++ b/cmds/stagefright/filters/saturation.rs
@@ -0,0 +1,40 @@
+/*
+ * Copyright (C) 2014 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.
+ */
+
+#pragma version(1)
+#pragma rs java_package_name(com.android.rs.cppbasic)
+#pragma rs_fp_relaxed
+
+const static float3 gMonoMult = {0.299f, 0.587f, 0.114f};
+
+// global variables (parameters accessible to application code)
+float gSaturation = 1.0f;
+
+void root(const uchar4 *v_in, uchar4 *v_out) {
+    v_out->x = v_in->x; // don't modify A
+
+    // get RGB, scale 0-255 uchar to 0-1.0 float
+    float3 rgb = {v_in->y * 0.003921569f, v_in->z * 0.003921569f,
+            v_in->w * 0.003921569f};
+
+    // apply saturation filter
+    float3 result = dot(rgb, gMonoMult);
+    result = mix(result, rgb, gSaturation);
+
+    v_out->y = (uchar)clamp((result.r * 255.f + 0.5f), 0.f, 255.f);
+    v_out->z = (uchar)clamp((result.g * 255.f + 0.5f), 0.f, 255.f);
+    v_out->w = (uchar)clamp((result.b * 255.f + 0.5f), 0.f, 255.f);
+}
diff --git a/cmds/stagefright/mediafilter.cpp b/cmds/stagefright/mediafilter.cpp
new file mode 100644
index 0000000..f77b38b
--- /dev/null
+++ b/cmds/stagefright/mediafilter.cpp
@@ -0,0 +1,785 @@
+/*
+ * Copyright (C) 2014 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 "mediafilterTest"
+
+#include <inttypes.h>
+
+#include <binder/ProcessState.h>
+#include <filters/ColorConvert.h>
+#include <gui/ISurfaceComposer.h>
+#include <gui/SurfaceComposerClient.h>
+#include <gui/Surface.h>
+#include <media/ICrypto.h>
+#include <media/IMediaHTTPService.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/MediaCodec.h>
+#include <media/stagefright/NuMediaExtractor.h>
+#include <media/stagefright/RenderScriptWrapper.h>
+#include <OMX_IVCommon.h>
+#include <ui/DisplayInfo.h>
+
+#include "RenderScript.h"
+#include "ScriptC_argbtorgba.h"
+#include "ScriptC_nightvision.h"
+#include "ScriptC_saturation.h"
+
+// test parameters
+static const bool kTestFlush = true;        // Note: true will drop 1 out of
+static const int kFlushAfterFrames = 25;    // kFlushAfterFrames output frames
+static const int64_t kTimeout = 500ll;
+
+// built-in filter parameters
+static const int32_t kInvert = false;   // ZeroFilter param
+static const float kBlurRadius = 15.0f; // IntrinsicBlurFilter param
+static const float kSaturation = 0.0f;  // SaturationFilter param
+
+static void usage(const char *me) {
+    fprintf(stderr, "usage: [flags] %s\n"
+                    "\t[-b] use IntrinsicBlurFilter\n"
+                    "\t[-c] use argb to rgba conversion RSFilter\n"
+                    "\t[-n] use night vision RSFilter\n"
+                    "\t[-r] use saturation RSFilter\n"
+                    "\t[-s] use SaturationFilter\n"
+                    "\t[-z] use ZeroFilter (copy filter)\n"
+                    "\t[-R] render output to surface (enables -S)\n"
+                    "\t[-S] allocate buffers from a surface\n"
+                    "\t[-T] use render timestamps (enables -R)\n",
+                    me);
+    exit(1);
+}
+
+namespace android {
+
+struct SaturationRSFilter : RenderScriptWrapper::RSFilterCallback {
+    void init(RSC::sp<RSC::RS> context) {
+        mScript = new ScriptC_saturation(context);
+        mScript->set_gSaturation(3.f);
+    }
+
+    virtual status_t processBuffers(
+            RSC::Allocation *inBuffer, RSC::Allocation *outBuffer) {
+        mScript->forEach_root(inBuffer, outBuffer);
+
+        return OK;
+    }
+
+    status_t handleSetParameters(const sp<AMessage> &msg) {
+        return OK;
+    }
+
+private:
+    RSC::sp<ScriptC_saturation> mScript;
+};
+
+struct NightVisionRSFilter : RenderScriptWrapper::RSFilterCallback {
+    void init(RSC::sp<RSC::RS> context) {
+        mScript = new ScriptC_nightvision(context);
+    }
+
+    virtual status_t processBuffers(
+            RSC::Allocation *inBuffer, RSC::Allocation *outBuffer) {
+        mScript->forEach_root(inBuffer, outBuffer);
+
+        return OK;
+    }
+
+    status_t handleSetParameters(const sp<AMessage> &msg) {
+        return OK;
+    }
+
+private:
+    RSC::sp<ScriptC_nightvision> mScript;
+};
+
+struct ARGBToRGBARSFilter : RenderScriptWrapper::RSFilterCallback {
+    void init(RSC::sp<RSC::RS> context) {
+        mScript = new ScriptC_argbtorgba(context);
+    }
+
+    virtual status_t processBuffers(
+            RSC::Allocation *inBuffer, RSC::Allocation *outBuffer) {
+        mScript->forEach_root(inBuffer, outBuffer);
+
+        return OK;
+    }
+
+    status_t handleSetParameters(const sp<AMessage> &msg) {
+        return OK;
+    }
+
+private:
+    RSC::sp<ScriptC_argbtorgba> mScript;
+};
+
+struct CodecState {
+    sp<MediaCodec> mCodec;
+    Vector<sp<ABuffer> > mInBuffers;
+    Vector<sp<ABuffer> > mOutBuffers;
+    bool mSignalledInputEOS;
+    bool mSawOutputEOS;
+    int64_t mNumBuffersDecoded;
+};
+
+struct DecodedFrame {
+    size_t index;
+    size_t offset;
+    size_t size;
+    int64_t presentationTimeUs;
+    uint32_t flags;
+};
+
+enum FilterType {
+    FILTERTYPE_ZERO,
+    FILTERTYPE_INTRINSIC_BLUR,
+    FILTERTYPE_SATURATION,
+    FILTERTYPE_RS_SATURATION,
+    FILTERTYPE_RS_NIGHT_VISION,
+    FILTERTYPE_RS_ARGB_TO_RGBA,
+};
+
+size_t inputFramesSinceFlush = 0;
+void tryCopyDecodedBuffer(
+        List<DecodedFrame> *decodedFrameIndices,
+        CodecState *filterState,
+        CodecState *vidState) {
+    if (decodedFrameIndices->empty()) {
+        return;
+    }
+
+    size_t filterIndex;
+    status_t err = filterState->mCodec->dequeueInputBuffer(
+            &filterIndex, kTimeout);
+    if (err != OK) {
+        return;
+    }
+
+    ++inputFramesSinceFlush;
+
+    DecodedFrame frame = *decodedFrameIndices->begin();
+
+    // only consume a buffer if we are not going to flush, since we expect
+    // the dequeue -> flush -> queue operation to cause an error and
+    // not produce an output frame
+    if (!kTestFlush || inputFramesSinceFlush < kFlushAfterFrames) {
+        decodedFrameIndices->erase(decodedFrameIndices->begin());
+    }
+    size_t outIndex = frame.index;
+
+    const sp<ABuffer> &srcBuffer =
+        vidState->mOutBuffers.itemAt(outIndex);
+    const sp<ABuffer> &destBuffer =
+        filterState->mInBuffers.itemAt(filterIndex);
+
+    sp<AMessage> srcFormat, destFormat;
+    vidState->mCodec->getOutputFormat(&srcFormat);
+    filterState->mCodec->getInputFormat(&destFormat);
+
+    int32_t srcWidth, srcHeight, srcStride, srcSliceHeight;
+    int32_t srcColorFormat, destColorFormat;
+    int32_t destWidth, destHeight, destStride, destSliceHeight;
+    CHECK(srcFormat->findInt32("stride", &srcStride)
+            && srcFormat->findInt32("slice-height", &srcSliceHeight)
+            && srcFormat->findInt32("width", &srcWidth)
+            && srcFormat->findInt32("height", & srcHeight)
+            && srcFormat->findInt32("color-format", &srcColorFormat));
+    CHECK(destFormat->findInt32("stride", &destStride)
+            && destFormat->findInt32("slice-height", &destSliceHeight)
+            && destFormat->findInt32("width", &destWidth)
+            && destFormat->findInt32("height", & destHeight)
+            && destFormat->findInt32("color-format", &destColorFormat));
+
+    CHECK(srcWidth <= destStride && srcHeight <= destSliceHeight);
+
+    convertYUV420spToARGB(
+            srcBuffer->data(),
+            srcBuffer->data() + srcStride * srcSliceHeight,
+            srcWidth,
+            srcHeight,
+            destBuffer->data());
+
+    // copy timestamp
+    int64_t timeUs;
+    CHECK(srcBuffer->meta()->findInt64("timeUs", &timeUs));
+    destBuffer->meta()->setInt64("timeUs", timeUs);
+
+    if (kTestFlush && inputFramesSinceFlush >= kFlushAfterFrames) {
+        inputFramesSinceFlush = 0;
+
+        // check that queueing a buffer that was dequeued before flush
+        // fails with expected error EACCES
+        filterState->mCodec->flush();
+
+        err = filterState->mCodec->queueInputBuffer(
+                filterIndex, 0 /* offset */, destBuffer->size(),
+                timeUs, frame.flags);
+
+        if (err == OK) {
+            ALOGE("FAIL: queue after flush returned OK");
+        } else if (err != -EACCES) {
+            ALOGE("queueInputBuffer after flush returned %d, "
+                    "expected -EACCES (-13)", err);
+        }
+    } else {
+        err = filterState->mCodec->queueInputBuffer(
+                filterIndex, 0 /* offset */, destBuffer->size(),
+                timeUs, frame.flags);
+        CHECK(err == OK);
+
+        err = vidState->mCodec->releaseOutputBuffer(outIndex);
+        CHECK(err == OK);
+    }
+}
+
+size_t outputFramesSinceFlush = 0;
+void tryDrainOutputBuffer(
+        CodecState *filterState,
+        const sp<Surface> &surface, bool renderSurface,
+        bool useTimestamp, int64_t *startTimeRender) {
+    size_t index;
+    size_t offset;
+    size_t size;
+    int64_t presentationTimeUs;
+    uint32_t flags;
+    status_t err = filterState->mCodec->dequeueOutputBuffer(
+            &index, &offset, &size, &presentationTimeUs, &flags,
+            kTimeout);
+
+    if (err != OK) {
+        return;
+    }
+
+    ++outputFramesSinceFlush;
+
+    if (kTestFlush && outputFramesSinceFlush >= kFlushAfterFrames) {
+        filterState->mCodec->flush();
+    }
+
+    if (surface == NULL || !renderSurface) {
+        err = filterState->mCodec->releaseOutputBuffer(index);
+    } else if (useTimestamp) {
+        if (*startTimeRender == -1) {
+            // begin rendering 2 vsyncs after first decode
+            *startTimeRender = systemTime(SYSTEM_TIME_MONOTONIC)
+                    + 33000000 - (presentationTimeUs * 1000);
+        }
+        presentationTimeUs =
+                (presentationTimeUs * 1000) + *startTimeRender;
+        err = filterState->mCodec->renderOutputBufferAndRelease(
+                index, presentationTimeUs);
+    } else {
+        err = filterState->mCodec->renderOutputBufferAndRelease(index);
+    }
+
+    if (kTestFlush && outputFramesSinceFlush >= kFlushAfterFrames) {
+        outputFramesSinceFlush = 0;
+
+        // releasing the buffer dequeued before flush should cause an error
+        // if so, the frame will also be skipped in output stream
+        if (err == OK) {
+            ALOGE("FAIL: release after flush returned OK");
+        } else if (err != -EACCES) {
+            ALOGE("releaseOutputBuffer after flush returned %d, "
+                    "expected -EACCES (-13)", err);
+        }
+    } else {
+        CHECK(err == OK);
+    }
+
+    if (flags & MediaCodec::BUFFER_FLAG_EOS) {
+        ALOGV("reached EOS on output.");
+        filterState->mSawOutputEOS = true;
+    }
+}
+
+static int decode(
+        const sp<ALooper> &looper,
+        const char *path,
+        const sp<Surface> &surface,
+        bool renderSurface,
+        bool useTimestamp,
+        FilterType filterType) {
+
+    static int64_t kTimeout = 500ll;
+
+    sp<NuMediaExtractor> extractor = new NuMediaExtractor;
+    if (extractor->setDataSource(NULL /* httpService */, path) != OK) {
+        fprintf(stderr, "unable to instantiate extractor.\n");
+        return 1;
+    }
+
+    KeyedVector<size_t, CodecState> stateByTrack;
+
+    CodecState *vidState = NULL;
+    for (size_t i = 0; i < extractor->countTracks(); ++i) {
+        sp<AMessage> format;
+        status_t err = extractor->getTrackFormat(i, &format);
+        CHECK(err == OK);
+
+        AString mime;
+        CHECK(format->findString("mime", &mime));
+        bool isVideo = !strncasecmp(mime.c_str(), "video/", 6);
+        if (!isVideo) {
+            continue;
+        }
+
+        ALOGV("selecting track %zu", i);
+
+        err = extractor->selectTrack(i);
+        CHECK(err == OK);
+
+        CodecState *state =
+            &stateByTrack.editValueAt(stateByTrack.add(i, CodecState()));
+
+        vidState = state;
+
+        state->mNumBuffersDecoded = 0;
+
+        state->mCodec = MediaCodec::CreateByType(
+                looper, mime.c_str(), false /* encoder */);
+
+        CHECK(state->mCodec != NULL);
+
+        err = state->mCodec->configure(
+                format, NULL /* surface */, NULL /* crypto */, 0 /* flags */);
+
+        CHECK(err == OK);
+
+        state->mSignalledInputEOS = false;
+        state->mSawOutputEOS = false;
+
+        break;
+    }
+    CHECK(!stateByTrack.isEmpty());
+    CHECK(vidState != NULL);
+    sp<AMessage> vidFormat;
+    vidState->mCodec->getOutputFormat(&vidFormat);
+
+    // set filter to use ARGB8888
+    vidFormat->setInt32("color-format", OMX_COLOR_Format32bitARGB8888);
+    // set app cache directory path
+    vidFormat->setString("cacheDir", "/system/bin");
+
+    // create RenderScript context for RSFilters
+    RSC::sp<RSC::RS> context = new RSC::RS();
+    context->init("/system/bin");
+
+    sp<RenderScriptWrapper::RSFilterCallback> rsFilter;
+
+    // create renderscript wrapper for RSFilters
+    sp<RenderScriptWrapper> rsWrapper = new RenderScriptWrapper;
+    rsWrapper->mContext = context.get();
+
+    CodecState *filterState = new CodecState();
+    filterState->mNumBuffersDecoded = 0;
+
+    sp<AMessage> params = new AMessage();
+
+    switch (filterType) {
+        case FILTERTYPE_ZERO:
+        {
+            filterState->mCodec = MediaCodec::CreateByComponentName(
+                    looper, "android.filter.zerofilter");
+            params->setInt32("invert", kInvert);
+            break;
+        }
+        case FILTERTYPE_INTRINSIC_BLUR:
+        {
+            filterState->mCodec = MediaCodec::CreateByComponentName(
+                    looper, "android.filter.intrinsicblur");
+            params->setFloat("blur-radius", kBlurRadius);
+            break;
+        }
+        case FILTERTYPE_SATURATION:
+        {
+            filterState->mCodec = MediaCodec::CreateByComponentName(
+                    looper, "android.filter.saturation");
+            params->setFloat("saturation", kSaturation);
+            break;
+        }
+        case FILTERTYPE_RS_SATURATION:
+        {
+            SaturationRSFilter *satFilter = new SaturationRSFilter;
+            satFilter->init(context);
+            rsFilter = satFilter;
+            rsWrapper->mCallback = rsFilter;
+            vidFormat->setObject("rs-wrapper", rsWrapper);
+
+            filterState->mCodec = MediaCodec::CreateByComponentName(
+                    looper, "android.filter.RenderScript");
+            break;
+        }
+        case FILTERTYPE_RS_NIGHT_VISION:
+        {
+            NightVisionRSFilter *nightVisionFilter = new NightVisionRSFilter;
+            nightVisionFilter->init(context);
+            rsFilter = nightVisionFilter;
+            rsWrapper->mCallback = rsFilter;
+            vidFormat->setObject("rs-wrapper", rsWrapper);
+
+            filterState->mCodec = MediaCodec::CreateByComponentName(
+                    looper, "android.filter.RenderScript");
+            break;
+        }
+        case FILTERTYPE_RS_ARGB_TO_RGBA:
+        {
+            ARGBToRGBARSFilter *argbToRgbaFilter = new ARGBToRGBARSFilter;
+            argbToRgbaFilter->init(context);
+            rsFilter = argbToRgbaFilter;
+            rsWrapper->mCallback = rsFilter;
+            vidFormat->setObject("rs-wrapper", rsWrapper);
+
+            filterState->mCodec = MediaCodec::CreateByComponentName(
+                    looper, "android.filter.RenderScript");
+            break;
+        }
+        default:
+        {
+            LOG_ALWAYS_FATAL("mediacodec.cpp error: unrecognized FilterType");
+            break;
+        }
+    }
+    CHECK(filterState->mCodec != NULL);
+
+    status_t err = filterState->mCodec->configure(
+            vidFormat /* format */, surface, NULL /* crypto */, 0 /* flags */);
+    CHECK(err == OK);
+
+    filterState->mSignalledInputEOS = false;
+    filterState->mSawOutputEOS = false;
+
+    int64_t startTimeUs = ALooper::GetNowUs();
+    int64_t startTimeRender = -1;
+
+    for (size_t i = 0; i < stateByTrack.size(); ++i) {
+        CodecState *state = &stateByTrack.editValueAt(i);
+
+        sp<MediaCodec> codec = state->mCodec;
+
+        CHECK_EQ((status_t)OK, codec->start());
+
+        CHECK_EQ((status_t)OK, codec->getInputBuffers(&state->mInBuffers));
+        CHECK_EQ((status_t)OK, codec->getOutputBuffers(&state->mOutBuffers));
+
+        ALOGV("got %zu input and %zu output buffers",
+                state->mInBuffers.size(), state->mOutBuffers.size());
+    }
+
+    CHECK_EQ((status_t)OK, filterState->mCodec->setParameters(params));
+
+    if (kTestFlush) {
+        status_t flushErr = filterState->mCodec->flush();
+        if (flushErr == OK) {
+            ALOGE("FAIL: Flush before start returned OK");
+        } else {
+            ALOGV("Flush before start returned status %d, usually ENOSYS (-38)",
+                    flushErr);
+        }
+    }
+
+    CHECK_EQ((status_t)OK, filterState->mCodec->start());
+    CHECK_EQ((status_t)OK, filterState->mCodec->getInputBuffers(
+            &filterState->mInBuffers));
+    CHECK_EQ((status_t)OK, filterState->mCodec->getOutputBuffers(
+            &filterState->mOutBuffers));
+
+    if (kTestFlush) {
+        status_t flushErr = filterState->mCodec->flush();
+        if (flushErr != OK) {
+            ALOGE("FAIL: Flush after start returned %d, expect OK (0)",
+                    flushErr);
+        } else {
+            ALOGV("Flush immediately after start OK");
+        }
+    }
+
+    List<DecodedFrame> decodedFrameIndices;
+
+    // loop until decoder reaches EOS
+    bool sawInputEOS = false;
+    bool sawOutputEOSOnAllTracks = false;
+    while (!sawOutputEOSOnAllTracks) {
+        if (!sawInputEOS) {
+            size_t trackIndex;
+            status_t err = extractor->getSampleTrackIndex(&trackIndex);
+
+            if (err != OK) {
+                ALOGV("saw input eos");
+                sawInputEOS = true;
+            } else {
+                CodecState *state = &stateByTrack.editValueFor(trackIndex);
+
+                size_t index;
+                err = state->mCodec->dequeueInputBuffer(&index, kTimeout);
+
+                if (err == OK) {
+                    ALOGV("filling input buffer %zu", index);
+
+                    const sp<ABuffer> &buffer = state->mInBuffers.itemAt(index);
+
+                    err = extractor->readSampleData(buffer);
+                    CHECK(err == OK);
+
+                    int64_t timeUs;
+                    err = extractor->getSampleTime(&timeUs);
+                    CHECK(err == OK);
+
+                    uint32_t bufferFlags = 0;
+
+                    err = state->mCodec->queueInputBuffer(
+                            index, 0 /* offset */, buffer->size(),
+                            timeUs, bufferFlags);
+
+                    CHECK(err == OK);
+
+                    extractor->advance();
+                } else {
+                    CHECK_EQ(err, -EAGAIN);
+                }
+            }
+        } else {
+            for (size_t i = 0; i < stateByTrack.size(); ++i) {
+                CodecState *state = &stateByTrack.editValueAt(i);
+
+                if (!state->mSignalledInputEOS) {
+                    size_t index;
+                    status_t err =
+                        state->mCodec->dequeueInputBuffer(&index, kTimeout);
+
+                    if (err == OK) {
+                        ALOGV("signalling input EOS on track %zu", i);
+
+                        err = state->mCodec->queueInputBuffer(
+                                index, 0 /* offset */, 0 /* size */,
+                                0ll /* timeUs */, MediaCodec::BUFFER_FLAG_EOS);
+
+                        CHECK(err == OK);
+
+                        state->mSignalledInputEOS = true;
+                    } else {
+                        CHECK_EQ(err, -EAGAIN);
+                    }
+                }
+            }
+        }
+
+        sawOutputEOSOnAllTracks = true;
+        for (size_t i = 0; i < stateByTrack.size(); ++i) {
+            CodecState *state = &stateByTrack.editValueAt(i);
+
+            if (state->mSawOutputEOS) {
+                continue;
+            } else {
+                sawOutputEOSOnAllTracks = false;
+            }
+
+            DecodedFrame frame;
+            status_t err = state->mCodec->dequeueOutputBuffer(
+                    &frame.index, &frame.offset, &frame.size,
+                    &frame.presentationTimeUs, &frame.flags, kTimeout);
+
+            if (err == OK) {
+                ALOGV("draining decoded buffer %zu, time = %lld us",
+                        frame.index, frame.presentationTimeUs);
+
+                ++(state->mNumBuffersDecoded);
+
+                decodedFrameIndices.push_back(frame);
+
+                if (frame.flags & MediaCodec::BUFFER_FLAG_EOS) {
+                    ALOGV("reached EOS on decoder output.");
+                    state->mSawOutputEOS = true;
+                }
+
+            } else if (err == INFO_OUTPUT_BUFFERS_CHANGED) {
+                ALOGV("INFO_OUTPUT_BUFFERS_CHANGED");
+                CHECK_EQ((status_t)OK, state->mCodec->getOutputBuffers(
+                        &state->mOutBuffers));
+
+                ALOGV("got %zu output buffers", state->mOutBuffers.size());
+            } else if (err == INFO_FORMAT_CHANGED) {
+                sp<AMessage> format;
+                CHECK_EQ((status_t)OK, state->mCodec->getOutputFormat(&format));
+
+                ALOGV("INFO_FORMAT_CHANGED: %s",
+                        format->debugString().c_str());
+            } else {
+                CHECK_EQ(err, -EAGAIN);
+            }
+
+            tryCopyDecodedBuffer(&decodedFrameIndices, filterState, vidState);
+
+            tryDrainOutputBuffer(
+                    filterState, surface, renderSurface,
+                    useTimestamp, &startTimeRender);
+        }
+    }
+
+    // after EOS on decoder, let filter reach EOS
+    while (!filterState->mSawOutputEOS) {
+        tryCopyDecodedBuffer(&decodedFrameIndices, filterState, vidState);
+
+        tryDrainOutputBuffer(
+                filterState, surface, renderSurface,
+                useTimestamp, &startTimeRender);
+    }
+
+    int64_t elapsedTimeUs = ALooper::GetNowUs() - startTimeUs;
+
+    for (size_t i = 0; i < stateByTrack.size(); ++i) {
+        CodecState *state = &stateByTrack.editValueAt(i);
+
+        CHECK_EQ((status_t)OK, state->mCodec->release());
+
+        printf("track %zu: %" PRId64 " frames decoded and filtered, "
+                "%.2f fps.\n", i, state->mNumBuffersDecoded,
+                state->mNumBuffersDecoded * 1E6 / elapsedTimeUs);
+    }
+
+    return 0;
+}
+
+}  // namespace android
+
+int main(int argc, char **argv) {
+    using namespace android;
+
+    const char *me = argv[0];
+
+    bool useSurface = false;
+    bool renderSurface = false;
+    bool useTimestamp = false;
+    FilterType filterType = FILTERTYPE_ZERO;
+
+    int res;
+    while ((res = getopt(argc, argv, "bcnrszTRSh")) >= 0) {
+        switch (res) {
+            case 'b':
+            {
+                filterType = FILTERTYPE_INTRINSIC_BLUR;
+                break;
+            }
+            case 'c':
+            {
+                filterType = FILTERTYPE_RS_ARGB_TO_RGBA;
+                break;
+            }
+            case 'n':
+            {
+                filterType = FILTERTYPE_RS_NIGHT_VISION;
+                break;
+            }
+            case 'r':
+            {
+                filterType = FILTERTYPE_RS_SATURATION;
+                break;
+            }
+            case 's':
+            {
+                filterType = FILTERTYPE_SATURATION;
+                break;
+            }
+            case 'z':
+            {
+                filterType = FILTERTYPE_ZERO;
+                break;
+            }
+            case 'T':
+            {
+                useTimestamp = true;
+            }
+            // fall through
+            case 'R':
+            {
+                renderSurface = true;
+            }
+            // fall through
+            case 'S':
+            {
+                useSurface = true;
+                break;
+            }
+            case '?':
+            case 'h':
+            default:
+            {
+                usage(me);
+                break;
+            }
+        }
+    }
+
+    argc -= optind;
+    argv += optind;
+
+    if (argc != 1) {
+        usage(me);
+    }
+
+    ProcessState::self()->startThreadPool();
+
+    DataSource::RegisterDefaultSniffers();
+
+    android::sp<ALooper> looper = new ALooper;
+    looper->start();
+
+    android::sp<SurfaceComposerClient> composerClient;
+    android::sp<SurfaceControl> control;
+    android::sp<Surface> surface;
+
+    if (useSurface) {
+        composerClient = new SurfaceComposerClient;
+        CHECK_EQ((status_t)OK, composerClient->initCheck());
+
+        android::sp<IBinder> display(SurfaceComposerClient::getBuiltInDisplay(
+                ISurfaceComposer::eDisplayIdMain));
+        DisplayInfo info;
+        SurfaceComposerClient::getDisplayInfo(display, &info);
+        ssize_t displayWidth = info.w;
+        ssize_t displayHeight = info.h;
+
+        ALOGV("display is %zd x %zd", displayWidth, displayHeight);
+
+        control = composerClient->createSurface(
+                String8("A Surface"), displayWidth, displayHeight,
+                PIXEL_FORMAT_RGBA_8888, 0);
+
+        CHECK(control != NULL);
+        CHECK(control->isValid());
+
+        SurfaceComposerClient::openGlobalTransaction();
+        CHECK_EQ((status_t)OK, control->setLayer(INT_MAX));
+        CHECK_EQ((status_t)OK, control->show());
+        SurfaceComposerClient::closeGlobalTransaction();
+
+        surface = control->getSurface();
+        CHECK(surface != NULL);
+    }
+
+    decode(looper, argv[0], surface, renderSurface, useTimestamp, filterType);
+
+    if (useSurface) {
+        composerClient->dispose();
+    }
+
+    looper->stop();
+
+    return 0;
+}
diff --git a/cmds/stagefright/muxer.cpp b/cmds/stagefright/muxer.cpp
index f4a33e8..461b56c 100644
--- a/cmds/stagefright/muxer.cpp
+++ b/cmds/stagefright/muxer.cpp
@@ -17,6 +17,9 @@
 //#define LOG_NDEBUG 0
 #define LOG_TAG "muxer"
 #include <inttypes.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
 #include <utils/Log.h>
 
 #include <binder/ProcessState.h>
@@ -72,8 +75,15 @@
     ALOGV("input file %s, output file %s", path, outputFileName);
     ALOGV("useAudio %d, useVideo %d", useAudio, useVideo);
 
-    sp<MediaMuxer> muxer = new MediaMuxer(outputFileName,
+    int fd = open(outputFileName, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+
+    if (fd < 0) {
+        ALOGE("couldn't open file");
+        return fd;
+    }
+    sp<MediaMuxer> muxer = new MediaMuxer(fd,
                                           MediaMuxer::OUTPUT_FORMAT_MPEG_4);
+    close(fd);
 
     size_t trackCount = extractor->countTracks();
     // Map the extractor's track index to the muxer's track index.
diff --git a/cmds/stagefright/recordvideo.cpp b/cmds/stagefright/recordvideo.cpp
index 9f547c7..2ad40bd 100644
--- a/cmds/stagefright/recordvideo.cpp
+++ b/cmds/stagefright/recordvideo.cpp
@@ -17,6 +17,10 @@
 #include "SineSource.h"
 
 #include <inttypes.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+
 #include <binder/ProcessState.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/AudioPlayer.h>
@@ -300,7 +304,13 @@
                 client.interface(), enc_meta, true /* createEncoder */, source,
                 0, preferSoftwareCodec ? OMXCodec::kPreferSoftwareCodecs : 0);
 
-    sp<MPEG4Writer> writer = new MPEG4Writer(fileName);
+    int fd = open(fileName, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+    if (fd < 0) {
+        fprintf(stderr, "couldn't open file");
+        return 1;
+    }
+    sp<MPEG4Writer> writer = new MPEG4Writer(fd);
+    close(fd);
     writer->addSource(encoder);
     int64_t start = systemTime();
     CHECK_EQ((status_t)OK, writer->start());
diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index 81edcb4..318b56d 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -19,6 +19,8 @@
 #include <stdlib.h>
 #include <string.h>
 #include <sys/time.h>
+#include <sys/types.h>
+#include <sys/stat.h>
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "stagefright"
@@ -506,8 +508,13 @@
     sp<MPEG4Writer> writer =
         new MPEG4Writer(gWriteMP4Filename.string());
 #else
+    int fd = open(gWriteMP4Filename.string(), O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+    if (fd < 0) {
+        fprintf(stderr, "couldn't open file");
+        return;
+    }
     sp<MPEG2TSWriter> writer =
-        new MPEG2TSWriter(gWriteMP4Filename.string());
+        new MPEG2TSWriter(fd);
 #endif
 
     // at most one minute.
diff --git a/include/camera/ICameraService.h b/include/camera/ICameraService.h
index f7f06bb..cc41efe 100644
--- a/include/camera/ICameraService.h
+++ b/include/camera/ICameraService.h
@@ -53,6 +53,7 @@
         GET_LEGACY_PARAMETERS,
         SUPPORTS_CAMERA_API,
         CONNECT_LEGACY,
+        SET_TORCH_MODE,
     };
 
     enum {
@@ -142,6 +143,12 @@
             int clientUid,
             /*out*/
             sp<ICamera>& device) = 0;
+
+    /**
+     * Turn on or off a camera's torch mode.
+     */
+    virtual status_t setTorchMode(const String16& cameraId, bool enabled,
+            const sp<IBinder>& clientBinder) = 0;
 };
 
 // ----------------------------------------------------------------------------
diff --git a/include/camera/ICameraServiceListener.h b/include/camera/ICameraServiceListener.h
index 0a0e43a..9e8b912 100644
--- a/include/camera/ICameraServiceListener.h
+++ b/include/camera/ICameraServiceListener.h
@@ -66,9 +66,33 @@
         STATUS_UNKNOWN          = 0xFFFFFFFF,
     };
 
+    /**
+     * The torch mode status of a camera.
+     *
+     * Initial status will be transmitted with onTorchStatusChanged immediately
+     * after this listener is added to the service listener list.
+     */
+    enum TorchStatus {
+        // The camera's torch mode has become available to use via
+        // setTorchMode().
+        TORCH_STATUS_AVAILABLE      = TORCH_MODE_STATUS_AVAILABLE,
+        // The camera's torch mode has become not available to use via
+        // setTorchMode().
+        TORCH_STATUS_NOT_AVAILABLE  = TORCH_MODE_STATUS_RESOURCE_BUSY,
+        // The camera's torch mode has been turned off by setTorchMode().
+        TORCH_STATUS_OFF            = TORCH_MODE_STATUS_OFF,
+        // The camera's torch mode has been turned on by setTorchMode().
+        TORCH_STATUS_ON             = 0x80000000,
+
+        // Use to initialize variables only
+        TORCH_STATUS_UNKNOWN        = 0xFFFFFFFF,
+    };
+
     DECLARE_META_INTERFACE(CameraServiceListener);
 
     virtual void onStatusChanged(Status status, int32_t cameraId) = 0;
+
+    virtual void onTorchStatusChanged(TorchStatus status, const String16& cameraId) = 0;
 };
 
 // ----------------------------------------------------------------------------
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 843a354..2ab3dd6 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -201,7 +201,7 @@
     // 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);
+                                             const char *device_address, const char *device_name);
     static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
                                                                 const char *device_address);
     static status_t setPhoneState(audio_mode_t state);
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index fd51b8f..2e1ed6c 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -79,9 +79,7 @@
         size_t      size;         // input/output in bytes == frameCount * frameSize
                                   // on input it is unused
                                   // on output is the number of bytes actually filled
-                                  // FIXME this is redundant with respect to frameCount,
-                                  // and TRANSFER_OBTAIN mode is broken for 8-bit data
-                                  // since we don't define the frame format
+                                  // FIXME this is redundant with respect to frameCount.
 
         union {
             void*       raw;
@@ -154,9 +152,9 @@
      * streamType:         Select the type of audio stream this track is attached to
      *                     (e.g. AUDIO_STREAM_MUSIC).
      * sampleRate:         Data source sampling rate in Hz.
-     * format:             Audio format.  For mixed tracks, any PCM format supported by server is OK
-     *                     or AUDIO_FORMAT_PCM_8_BIT which is handled on client side.  For direct
-     *                     and offloaded tracks, the possible format(s) depends on the output sink.
+     * format:             Audio format. For mixed tracks, any PCM format supported by server is OK.
+     *                     For direct and offloaded tracks, the possible format(s) depends on the
+     *                     output sink.
      * channelMask:        Channel mask, such that audio_is_output_channel(channelMask) is true.
      * frameCount:         Minimum size of track PCM buffer in frames. This defines the
      *                     application's contribution to the
@@ -193,7 +191,6 @@
 
     /* Creates an audio track and registers it with AudioFlinger.
      * With this constructor, the track is configured for static buffer mode.
-     * The format must not be 8-bit linear PCM.
      * Data to be rendered is passed in a shared memory buffer
      * identified by the argument sharedBuffer, which must be non-0.
      * The memory should be initialized to the desired data before calling start().
@@ -614,6 +611,7 @@
 
                 void        pause();    // suspend thread from execution at next loop boundary
                 void        resume();   // allow thread to execute, if not requested to exit
+                void        wake();     // wake to handle changed notification conditions.
 
     private:
                 void        pauseInternal(nsecs_t ns = 0LL);
@@ -628,7 +626,9 @@
         bool                mPaused;    // whether thread is requested to pause at next loop entry
         bool                mPausedInt; // whether thread internally requests pause
         nsecs_t             mPausedNs;  // if mPausedInt then associated timeout, otherwise ignored
-        bool                mIgnoreNextPausedInt;   // whether to ignore next mPausedInt request
+        bool                mIgnoreNextPausedInt;   // skip any internal pause and go immediately
+                                        // to processAudioBuffer() as state may have changed
+                                        // since pause time calculated.
     };
 
             // body of AudioTrackThread::threadLoop()
@@ -680,7 +680,7 @@
 
     float                   mVolume[2];
     float                   mSendLevel;
-    mutable uint32_t        mSampleRate;            // mutable because getSampleRate() can update it.
+    mutable uint32_t        mSampleRate;            // mutable because getSampleRate() can update it
     size_t                  mFrameCount;            // corresponds to current IAudioTrack, value is
                                                     // reported back by AudioFlinger to the client
     size_t                  mReqFrameCount;         // frame count to request the first or next time
@@ -698,10 +698,7 @@
     const audio_offload_info_t* mOffloadInfo;
     audio_attributes_t      mAttributes;
 
-    // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data.  For 8-bit PCM data, it's
-    // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer.
-    size_t                  mFrameSize;             // app-level frame size
-    size_t                  mFrameSizeAF;           // AudioFlinger frame size
+    size_t                  mFrameSize;             // frame size in bytes
 
     status_t                mStatus;
 
@@ -732,13 +729,20 @@
     bool                    mRefreshRemaining;      // processAudioBuffer() should refresh
                                                     // mRemainingFrames and mRetryOnPartialBuffer
 
+                                                    // used for static track cbf and restoration
+    int32_t                 mLoopCount;             // last setLoop loopCount; zero means disabled
+    uint32_t                mLoopStart;             // last setLoop loopStart
+    uint32_t                mLoopEnd;               // last setLoop loopEnd
+    int32_t                 mLoopCountNotified;     // the last loopCount notified by callback.
+                                                    // mLoopCountNotified counts down, matching
+                                                    // the remaining loop count for static track
+                                                    // playback.
+
     // These are private to processAudioBuffer(), and are not protected by a lock
     uint32_t                mRemainingFrames;       // number of frames to request in obtainBuffer()
     bool                    mRetryOnPartialBuffer;  // sleep and retry after partial obtainBuffer()
     uint32_t                mObservedSequence;      // last observed value of mSequence
 
-    uint32_t                mLoopPeriod;            // in frames, zero means looping is disabled
-
     uint32_t                mMarkerPosition;        // in wrapping (overflow) frame units
     bool                    mMarkerReached;
     uint32_t                mNewPosition;           // in frames
diff --git a/include/media/EffectsFactoryApi.h b/include/media/EffectsFactoryApi.h
index b1ed7b0..64a3212 100644
--- a/include/media/EffectsFactoryApi.h
+++ b/include/media/EffectsFactoryApi.h
@@ -171,6 +171,8 @@
 ////////////////////////////////////////////////////////////////////////////////
 int EffectIsNullUuid(const effect_uuid_t *pEffectUuid);
 
+int EffectDumpEffects(int fd);
+
 #if __cplusplus
 }  // extern "C"
 #endif
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
index c98c475..fecc6f1 100644
--- a/include/media/IAudioPolicyService.h
+++ b/include/media/IAudioPolicyService.h
@@ -44,7 +44,8 @@
     //
     virtual status_t setDeviceConnectionState(audio_devices_t device,
                                               audio_policy_dev_state_t state,
-                                              const char *device_address) = 0;
+                                              const char *device_address,
+                                              const char *device_name) = 0;
     virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
                                                                   const char *device_address) = 0;
     virtual status_t setPhoneState(audio_mode_t state) = 0;
diff --git a/include/media/IMediaPlayerService.h b/include/media/IMediaPlayerService.h
index 67b599a..49a3d61 100644
--- a/include/media/IMediaPlayerService.h
+++ b/include/media/IMediaPlayerService.h
@@ -49,7 +49,8 @@
 
     virtual sp<IMediaRecorder> createMediaRecorder() = 0;
     virtual sp<IMediaMetadataRetriever> createMetadataRetriever() = 0;
-    virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0) = 0;
+    virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0)
+            = 0;
 
     virtual sp<IOMX>            getOMX() = 0;
     virtual sp<ICrypto>         makeCrypto() = 0;
diff --git a/include/media/IMediaRecorder.h b/include/media/IMediaRecorder.h
index 3e67550..509c06b 100644
--- a/include/media/IMediaRecorder.h
+++ b/include/media/IMediaRecorder.h
@@ -41,7 +41,6 @@
     virtual status_t setOutputFormat(int of) = 0;
     virtual status_t setVideoEncoder(int ve) = 0;
     virtual status_t setAudioEncoder(int ae) = 0;
-    virtual status_t setOutputFile(const char* path) = 0;
     virtual status_t setOutputFile(int fd, int64_t offset, int64_t length) = 0;
     virtual status_t setVideoSize(int width, int height) = 0;
     virtual status_t setVideoFrameRate(int frames_per_second) = 0;
diff --git a/include/media/IOMX.h b/include/media/IOMX.h
index 627f23b..6def65b 100644
--- a/include/media/IOMX.h
+++ b/include/media/IOMX.h
@@ -147,6 +147,7 @@
         INTERNAL_OPTION_SUSPEND,  // data is a bool
         INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY,  // data is an int64_t
         INTERNAL_OPTION_MAX_TIMESTAMP_GAP, // data is int64_t
+        INTERNAL_OPTION_MAX_FPS, // data is float
         INTERNAL_OPTION_START_TIME, // data is an int64_t
         INTERNAL_OPTION_TIME_LAPSE, // data is an int64_t[2]
     };
diff --git a/include/media/MediaRecorderBase.h b/include/media/MediaRecorderBase.h
index d7ac302..f55063e 100644
--- a/include/media/MediaRecorderBase.h
+++ b/include/media/MediaRecorderBase.h
@@ -43,7 +43,6 @@
     virtual status_t setCamera(const sp<ICamera>& camera,
                                const sp<ICameraRecordingProxy>& proxy) = 0;
     virtual status_t setPreviewSurface(const sp<IGraphicBufferProducer>& surface) = 0;
-    virtual status_t setOutputFile(const char *path) = 0;
     virtual status_t setOutputFile(int fd, int64_t offset, int64_t length) = 0;
     virtual status_t setOutputFileAuxiliary(int fd) {return INVALID_OPERATION;}
     virtual status_t setParameters(const String8& params) = 0;
diff --git a/include/media/SingleStateQueue.h b/include/media/SingleStateQueue.h
index 04c5fd0..d423962 100644
--- a/include/media/SingleStateQueue.h
+++ b/include/media/SingleStateQueue.h
@@ -21,6 +21,7 @@
 // Non-blocking single-reader / single-writer multi-word atomic load / store
 
 #include <stdint.h>
+#include <cutils/atomic.h>
 
 namespace android {
 
@@ -31,6 +32,12 @@
     class Mutator;
     class Observer;
 
+    enum SSQ_STATUS {
+        SSQ_PENDING, /* = 0 */
+        SSQ_READ,
+        SSQ_DONE,
+    };
+
     struct Shared {
         // needs to be part of a union so don't define constructor or destructor
 
@@ -41,28 +48,56 @@
         void                init() { mAck = 0; mSequence = 0; }
 
         volatile int32_t    mAck;
-#if 0
-        int                 mPad[7];
-        // cache line boundary
-#endif
         volatile int32_t    mSequence;
         T                   mValue;
     };
 
     class Mutator {
     public:
-        Mutator(Shared *shared);
-        /*virtual*/ ~Mutator() { }
+        Mutator(Shared *shared)
+            : mSequence(0), mShared(shared)
+        {
+            // exactly one of Mutator and Observer must initialize, currently it is Observer
+            // shared->init();
+        }
 
         // push new value onto state queue, overwriting previous value;
         // returns a sequence number which can be used with ack()
-        int32_t push(const T& value);
+        int32_t push(const T& value)
+        {
+            Shared *shared = mShared;
+            int32_t sequence = mSequence;
+            sequence++;
+            android_atomic_acquire_store(sequence, &shared->mSequence);
+            shared->mValue = value;
+            sequence++;
+            android_atomic_release_store(sequence, &shared->mSequence);
+            mSequence = sequence;
+            // consider signalling a futex here, if we know that observer is waiting
+            return sequence;
+        }
 
-        // return true if most recent push has been observed
-        bool ack();
+        // returns the status of the last state push.  This may be a stale value.
+        //
+        // SSQ_PENDING, or 0, means it has not been observed
+        // SSQ_READ means it has been read
+        // SSQ_DONE means it has been acted upon, after Observer::done() is called
+        enum SSQ_STATUS ack() const
+        {
+            // in the case of SSQ_DONE, prevent any subtle data-races of subsequent reads
+            // being performed (out-of-order) before the ack read, should the caller be
+            // depending on sequentiality of reads.
+            const int32_t ack = android_atomic_acquire_load(&mShared->mAck);
+            return ack - mSequence & ~1 ? SSQ_PENDING /* seq differ */ :
+                    ack & 1 ? SSQ_DONE : SSQ_READ;
+        }
 
         // return true if a push with specified sequence number or later has been observed
-        bool ack(int32_t sequence);
+        bool ack(int32_t sequence) const
+        {
+            // this relies on 2's complement rollover to detect an ancient sequence number
+            return mShared->mAck - sequence >= 0;
+        }
 
     private:
         int32_t     mSequence;
@@ -71,11 +106,54 @@
 
     class Observer {
     public:
-        Observer(Shared *shared);
-        /*virtual*/ ~Observer() { }
+        Observer(Shared *shared)
+            : mSequence(0), mSeed(1), mShared(shared)
+        {
+            // exactly one of Mutator and Observer must initialize, currently it is Observer
+            shared->init();
+        }
 
         // return true if value has changed
-        bool poll(T& value);
+        bool poll(T& value)
+        {
+            Shared *shared = mShared;
+            int32_t before = shared->mSequence;
+            if (before == mSequence) {
+                return false;
+            }
+            for (int tries = 0; ; ) {
+                const int MAX_TRIES = 5;
+                if (before & 1) {
+                    if (++tries >= MAX_TRIES) {
+                        return false;
+                    }
+                    before = shared->mSequence;
+                } else {
+                    android_memory_barrier();
+                    T temp = shared->mValue;
+                    int32_t after = android_atomic_release_load(&shared->mSequence);
+                    if (after == before) {
+                        value = temp;
+                        shared->mAck = before;
+                        mSequence = before; // mSequence is even after poll success
+                        return true;
+                    }
+                    if (++tries >= MAX_TRIES) {
+                        return false;
+                    }
+                    before = after;
+                }
+            }
+        }
+
+        // (optional) used to indicate to the Mutator that the state that has been polled
+        // has also been acted upon.
+        void done()
+        {
+            const int32_t ack = mShared->mAck + 1;
+            // ensure all previous writes have been performed.
+            android_atomic_release_store(ack, &mShared->mAck); // mSequence is odd after "done"
+        }
 
     private:
         int32_t     mSequence;
diff --git a/include/media/StringArray.h b/include/media/StringArray.h
index ae47085..48d98bf 100644
--- a/include/media/StringArray.h
+++ b/include/media/StringArray.h
@@ -16,7 +16,7 @@
 
 //
 // Sortable array of strings.  STL-ish, but STL-free.
-//  
+//
 #ifndef _LIBS_MEDIA_STRING_ARRAY_H
 #define _LIBS_MEDIA_STRING_ARRAY_H
 
diff --git a/include/media/mediarecorder.h b/include/media/mediarecorder.h
index b0a62a7..74a6469 100644
--- a/include/media/mediarecorder.h
+++ b/include/media/mediarecorder.h
@@ -221,7 +221,6 @@
     status_t    setOutputFormat(int of);
     status_t    setVideoEncoder(int ve);
     status_t    setAudioEncoder(int ae);
-    status_t    setOutputFile(const char* path);
     status_t    setOutputFile(int fd, int64_t offset, int64_t length);
     status_t    setVideoSize(int width, int height);
     status_t    setVideoFrameRate(int frames_per_second);
diff --git a/include/media/nbaio/NBAIO.h b/include/media/nbaio/NBAIO.h
index d422576..d9bbc8d 100644
--- a/include/media/nbaio/NBAIO.h
+++ b/include/media/nbaio/NBAIO.h
@@ -231,7 +231,8 @@
     virtual status_t getTimestamp(AudioTimestamp& timestamp) { return INVALID_OPERATION; }
 
 protected:
-    NBAIO_Sink(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesWritten(0) { }
+    NBAIO_Sink(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesWritten(0)
+            { }
     virtual ~NBAIO_Sink() { }
 
     // Implementations are free to ignore these if they don't need them
@@ -322,7 +323,8 @@
     virtual void    onTimestamp(const AudioTimestamp& timestamp) { }
 
 protected:
-    NBAIO_Source(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesRead(0) { }
+    NBAIO_Source(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesRead(0)
+            { }
     virtual ~NBAIO_Source() { }
 
     // Implementations are free to ignore these if they don't need them
diff --git a/include/media/nbaio/NBLog.h b/include/media/nbaio/NBLog.h
index bcbbc04..1297b51 100644
--- a/include/media/nbaio/NBLog.h
+++ b/include/media/nbaio/NBLog.h
@@ -21,7 +21,7 @@
 
 #include <binder/IMemory.h>
 #include <utils/Mutex.h>
-#include <media/nbaio/roundup.h>
+#include <audio_utils/roundup.h>
 
 namespace android {
 
diff --git a/include/media/stagefright/AACWriter.h b/include/media/stagefright/AACWriter.h
index d22707a..86417a5 100644
--- a/include/media/stagefright/AACWriter.h
+++ b/include/media/stagefright/AACWriter.h
@@ -27,7 +27,6 @@
 struct MetaData;
 
 struct AACWriter : public MediaWriter {
-    AACWriter(const char *filename);
     AACWriter(int fd);
 
     status_t initCheck() const;
diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h
index 595ace8..442c861 100644
--- a/include/media/stagefright/ACodec.h
+++ b/include/media/stagefright/ACodec.h
@@ -214,6 +214,7 @@
 
     int64_t mRepeatFrameDelayUs;
     int64_t mMaxPtsGapUs;
+    float mMaxFps;
 
     int64_t mTimePerFrameUs;
     int64_t mTimePerCaptureUs;
diff --git a/include/media/stagefright/AMRWriter.h b/include/media/stagefright/AMRWriter.h
index 392f968..bac878b 100644
--- a/include/media/stagefright/AMRWriter.h
+++ b/include/media/stagefright/AMRWriter.h
@@ -29,7 +29,6 @@
 struct MetaData;
 
 struct AMRWriter : public MediaWriter {
-    AMRWriter(const char *filename);
     AMRWriter(int fd);
 
     status_t initCheck() const;
diff --git a/include/media/stagefright/MPEG2TSWriter.h b/include/media/stagefright/MPEG2TSWriter.h
index 2e2922e..3d7960b 100644
--- a/include/media/stagefright/MPEG2TSWriter.h
+++ b/include/media/stagefright/MPEG2TSWriter.h
@@ -29,7 +29,6 @@
 
 struct MPEG2TSWriter : public MediaWriter {
     MPEG2TSWriter(int fd);
-    MPEG2TSWriter(const char *filename);
 
     MPEG2TSWriter(
             void *cookie,
diff --git a/include/media/stagefright/MPEG4Writer.h b/include/media/stagefright/MPEG4Writer.h
index 26ce5f9..899b324 100644
--- a/include/media/stagefright/MPEG4Writer.h
+++ b/include/media/stagefright/MPEG4Writer.h
@@ -32,7 +32,6 @@
 
 class MPEG4Writer : public MediaWriter {
 public:
-    MPEG4Writer(const char *filename);
     MPEG4Writer(int fd);
 
     // Limitations
diff --git a/include/media/stagefright/MediaFilter.h b/include/media/stagefright/MediaFilter.h
new file mode 100644
index 0000000..7b3f700
--- /dev/null
+++ b/include/media/stagefright/MediaFilter.h
@@ -0,0 +1,167 @@
+/*
+ * Copyright (C) 2014 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 MEDIA_FILTER_H_
+#define MEDIA_FILTER_H_
+
+#include <media/stagefright/CodecBase.h>
+
+namespace android {
+
+struct ABuffer;
+struct GraphicBufferListener;
+struct MemoryDealer;
+struct SimpleFilter;
+
+struct MediaFilter : public CodecBase {
+    MediaFilter();
+
+    virtual void setNotificationMessage(const sp<AMessage> &msg);
+
+    virtual void initiateAllocateComponent(const sp<AMessage> &msg);
+    virtual void initiateConfigureComponent(const sp<AMessage> &msg);
+    virtual void initiateCreateInputSurface();
+    virtual void initiateStart();
+    virtual void initiateShutdown(bool keepComponentAllocated = false);
+
+    virtual void signalFlush();
+    virtual void signalResume();
+
+    virtual void signalRequestIDRFrame();
+    virtual void signalSetParameters(const sp<AMessage> &msg);
+    virtual void signalEndOfInputStream();
+
+    virtual void onMessageReceived(const sp<AMessage> &msg);
+
+    struct PortDescription : public CodecBase::PortDescription {
+        virtual size_t countBuffers();
+        virtual IOMX::buffer_id bufferIDAt(size_t index) const;
+        virtual sp<ABuffer> bufferAt(size_t index) const;
+
+    protected:
+        PortDescription();
+
+    private:
+        friend struct MediaFilter;
+
+        Vector<IOMX::buffer_id> mBufferIDs;
+        Vector<sp<ABuffer> > mBuffers;
+
+        void addBuffer(IOMX::buffer_id id, const sp<ABuffer> &buffer);
+
+        DISALLOW_EVIL_CONSTRUCTORS(PortDescription);
+    };
+
+protected:
+    virtual ~MediaFilter();
+
+private:
+    struct BufferInfo {
+        enum Status {
+            OWNED_BY_US,
+            OWNED_BY_UPSTREAM,
+        };
+
+        IOMX::buffer_id mBufferID;
+        int32_t mGeneration;
+        int32_t mOutputFlags;
+        Status mStatus;
+
+        sp<ABuffer> mData;
+    };
+
+    enum State {
+      UNINITIALIZED,
+      INITIALIZED,
+      CONFIGURED,
+      STARTED,
+    };
+
+    enum {
+        kWhatInputBufferFilled       = 'inpF',
+        kWhatOutputBufferDrained     = 'outD',
+        kWhatShutdown                = 'shut',
+        kWhatFlush                   = 'flus',
+        kWhatResume                  = 'resm',
+        kWhatAllocateComponent       = 'allo',
+        kWhatConfigureComponent      = 'conf',
+        kWhatCreateInputSurface      = 'cisf',
+        kWhatSignalEndOfInputStream  = 'eois',
+        kWhatStart                   = 'star',
+        kWhatSetParameters           = 'setP',
+        kWhatProcessBuffers          = 'proc',
+    };
+
+    enum {
+        kPortIndexInput  = 0,
+        kPortIndexOutput = 1
+    };
+
+    // member variables
+    AString mComponentName;
+    State mState;
+    status_t mInputEOSResult;
+    int32_t mWidth, mHeight;
+    int32_t mStride, mSliceHeight;
+    int32_t mColorFormatIn, mColorFormatOut;
+    size_t mMaxInputSize, mMaxOutputSize;
+    int32_t mGeneration;
+    sp<AMessage> mNotify;
+    sp<AMessage> mInputFormat;
+    sp<AMessage> mOutputFormat;
+
+    sp<MemoryDealer> mDealer[2];
+    Vector<BufferInfo> mBuffers[2];
+    Vector<BufferInfo*> mAvailableInputBuffers;
+    Vector<BufferInfo*> mAvailableOutputBuffers;
+    bool mPortEOS[2];
+
+    sp<SimpleFilter> mFilter;
+    sp<GraphicBufferListener> mGraphicBufferListener;
+
+    // helper functions
+    void signalProcessBuffers();
+    void signalError(status_t error);
+
+    status_t allocateBuffersOnPort(OMX_U32 portIndex);
+    BufferInfo *findBufferByID(
+            uint32_t portIndex, IOMX::buffer_id bufferID,
+            ssize_t *index = NULL);
+    void postFillThisBuffer(BufferInfo *info);
+    void postDrainThisBuffer(BufferInfo *info);
+    void postEOS();
+    void sendFormatChange();
+    void requestFillEmptyInput();
+    void processBuffers();
+
+    void onAllocateComponent(const sp<AMessage> &msg);
+    void onConfigureComponent(const sp<AMessage> &msg);
+    void onStart();
+    void onInputBufferFilled(const sp<AMessage> &msg);
+    void onOutputBufferDrained(const sp<AMessage> &msg);
+    void onShutdown(const sp<AMessage> &msg);
+    void onFlush();
+    void onSetParameters(const sp<AMessage> &msg);
+    void onCreateInputSurface();
+    void onInputFrameAvailable();
+    void onSignalEndOfInputStream();
+
+    DISALLOW_EVIL_CONSTRUCTORS(MediaFilter);
+};
+
+}  // namespace android
+
+#endif  // MEDIA_FILTER_H_
diff --git a/include/media/stagefright/MediaMuxer.h b/include/media/stagefright/MediaMuxer.h
index 9da98d9..e6538d1 100644
--- a/include/media/stagefright/MediaMuxer.h
+++ b/include/media/stagefright/MediaMuxer.h
@@ -50,9 +50,6 @@
         OUTPUT_FORMAT_LIST_END // must be last - used to validate format type
     };
 
-    // Construct the muxer with the output file path.
-    MediaMuxer(const char *path, OutputFormat format);
-
     // Construct the muxer with the file descriptor. Note that the MediaMuxer
     // will close this file at stop().
     MediaMuxer(int fd, OutputFormat format);
diff --git a/include/media/stagefright/RenderScriptWrapper.h b/include/media/stagefright/RenderScriptWrapper.h
new file mode 100644
index 0000000..b42649e
--- /dev/null
+++ b/include/media/stagefright/RenderScriptWrapper.h
@@ -0,0 +1,42 @@
+/*
+ * Copyright (C) 2014 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 RENDERSCRIPT_WRAPPER_H_
+#define RENDERSCRIPT_WRAPPER_H_
+
+#include <RenderScript.h>
+
+namespace android {
+
+struct RenderScriptWrapper : public RefBase {
+public:
+    struct RSFilterCallback : public RefBase {
+    public:
+        // called by RSFilter to process each input buffer
+        virtual status_t processBuffers(
+                RSC::Allocation* inBuffer,
+                RSC::Allocation* outBuffer) = 0;
+
+        virtual status_t handleSetParameters(const sp<AMessage> &msg) = 0;
+    };
+
+    sp<RSFilterCallback> mCallback;
+    RSC::sp<RSC::RS> mContext;
+};
+
+}   // namespace android
+
+#endif  // RENDERSCRIPT_WRAPPER_H_
diff --git a/include/ndk/NdkMediaCodec.h b/include/ndk/NdkMediaCodec.h
index c07f4c9..4f6a1ef 100644
--- a/include/ndk/NdkMediaCodec.h
+++ b/include/ndk/NdkMediaCodec.h
@@ -142,7 +142,8 @@
 /**
  * Get the index of the next available buffer of processed data.
  */
-ssize_t AMediaCodec_dequeueOutputBuffer(AMediaCodec*, AMediaCodecBufferInfo *info, int64_t timeoutUs);
+ssize_t AMediaCodec_dequeueOutputBuffer(AMediaCodec*, AMediaCodecBufferInfo *info,
+        int64_t timeoutUs);
 AMediaFormat* AMediaCodec_getOutputFormat(AMediaCodec*);
 
 /**
diff --git a/include/ndk/NdkMediaExtractor.h b/include/ndk/NdkMediaExtractor.h
index 7a4e702..7324d31 100644
--- a/include/ndk/NdkMediaExtractor.h
+++ b/include/ndk/NdkMediaExtractor.h
@@ -55,12 +55,14 @@
 /**
  *  Set the file descriptor from which the extractor will read.
  */
-media_status_t AMediaExtractor_setDataSourceFd(AMediaExtractor*, int fd, off64_t offset, off64_t length);
+media_status_t AMediaExtractor_setDataSourceFd(AMediaExtractor*, int fd, off64_t offset,
+        off64_t length);
 
 /**
  * Set the URI from which the extractor will read.
  */
-media_status_t AMediaExtractor_setDataSource(AMediaExtractor*, const char *location); // TODO support headers
+media_status_t AMediaExtractor_setDataSource(AMediaExtractor*, const char *location);
+        // TODO support headers
 
 /**
  * Return the number of tracks in the previously specified media file
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 31dff36..7143f1a 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -24,9 +24,8 @@
 #include <utils/threads.h>
 #include <utils/Log.h>
 #include <utils/RefBase.h>
-#include <media/nbaio/roundup.h>
+#include <audio_utils/roundup.h>
 #include <media/SingleStateQueue.h>
-#include <private/media/StaticAudioTrackState.h>
 
 namespace android {
 
@@ -61,15 +60,57 @@
     volatile uint32_t mUnderrunFrames;  // server increments for each unavailable but desired frame
 };
 
+// Represents a single state of an AudioTrack that was created in static mode (shared memory buffer
+// supplied by the client).  This state needs to be communicated from the client to server.  As this
+// state is too large to be updated atomically without a mutex, and mutexes aren't allowed here, the
+// state is wrapped by a SingleStateQueue.
+struct StaticAudioTrackState {
+    // Do not define constructors, destructors, or virtual methods as this is part of a
+    // union in shared memory and they will not get called properly.
+
+    // These fields should both be size_t, but since they are located in shared memory we
+    // force to 32-bit.  The client and server may have different typedefs for size_t.
+
+    // The state has a sequence counter to indicate whether changes are made to loop or position.
+    // The sequence counter also currently indicates whether loop or position is first depending
+    // on which is greater; it jumps by max(mLoopSequence, mPositionSequence) + 1.
+
+    uint32_t    mLoopStart;
+    uint32_t    mLoopEnd;
+    int32_t     mLoopCount;
+    uint32_t    mLoopSequence; // a sequence counter to indicate changes to loop
+    uint32_t    mPosition;
+    uint32_t    mPositionSequence; // a sequence counter to indicate changes to position
+};
+
 typedef SingleStateQueue<StaticAudioTrackState> StaticAudioTrackSingleStateQueue;
 
+struct StaticAudioTrackPosLoop {
+    // Do not define constructors, destructors, or virtual methods as this is part of a
+    // union in shared memory and will not get called properly.
+
+    // These fields should both be size_t, but since they are located in shared memory we
+    // force to 32-bit.  The client and server may have different typedefs for size_t.
+
+    // This struct information is stored in a single state queue to communicate the
+    // static AudioTrack server state to the client while data is consumed.
+    // It is smaller than StaticAudioTrackState to prevent unnecessary information from
+    // being sent.
+
+    uint32_t mBufferPosition;
+    int32_t  mLoopCount;
+};
+
+typedef SingleStateQueue<StaticAudioTrackPosLoop> StaticAudioTrackPosLoopQueue;
+
 struct AudioTrackSharedStatic {
+    // client requests to the server for loop or position changes.
     StaticAudioTrackSingleStateQueue::Shared
                     mSingleStateQueue;
-    // This field should be a size_t, but since it is located in shared memory we
-    // force to 32-bit.  The client and server may have different typedefs for size_t.
-    uint32_t        mBufferPosition;    // updated asynchronously by server,
-                                        // "for entertainment purposes only"
+    // position info updated asynchronously by server and read by client,
+    // "for entertainment purposes only"
+    StaticAudioTrackPosLoopQueue::Shared
+                    mPosLoopQueue;
 };
 
 // ----------------------------------------------------------------------------
@@ -96,7 +137,8 @@
                 uint32_t    mServer;    // Number of filled frames consumed by server (mIsOut),
                                         // or filled frames provided by server (!mIsOut).
                                         // It is updated asynchronously by server without a barrier.
-                                        // The value should be used "for entertainment purposes only",
+                                        // The value should be used
+                                        // "for entertainment purposes only",
                                         // which means don't make important decisions based on it.
 
                 uint32_t    mPad1;      // unused
@@ -313,8 +355,28 @@
     virtual void    flush();
 
 #define MIN_LOOP    16  // minimum length of each loop iteration in frames
+
+            // setLoop(), setBufferPosition(), and setBufferPositionAndLoop() set the
+            // static buffer position and looping parameters.  These commands are not
+            // synchronous (they do not wait or block); instead they take effect at the
+            // next buffer data read from the server side. However, the client side
+            // getters will read a cached version of the position and loop variables
+            // until the setting takes effect.
+            //
+            // setBufferPositionAndLoop() is equivalent to calling, in order, setLoop() and
+            // setBufferPosition().
+            //
+            // The functions should not be relied upon to do parameter or state checking.
+            // That is done at the AudioTrack level.
+
             void    setLoop(size_t loopStart, size_t loopEnd, int loopCount);
+            void    setBufferPosition(size_t position);
+            void    setBufferPositionAndLoop(size_t position, size_t loopStart, size_t loopEnd,
+                                             int loopCount);
             size_t  getBufferPosition();
+                    // getBufferPositionAndLoopCount() provides the proper snapshot of
+                    // position and loopCount together.
+            void    getBufferPositionAndLoopCount(size_t *position, int *loopCount);
 
     virtual size_t  getMisalignment() {
         return 0;
@@ -326,7 +388,9 @@
 
 private:
     StaticAudioTrackSingleStateQueue::Mutator   mMutator;
-    size_t          mBufferPosition;    // so that getBufferPosition() appears to be synchronous
+    StaticAudioTrackPosLoopQueue::Observer      mPosLoopObserver;
+                        StaticAudioTrackState   mState;   // last communicated state to server
+                        StaticAudioTrackPosLoop mPosLoop; // snapshot of position and loop.
 };
 
 // ----------------------------------------------------------------------------
@@ -447,10 +511,13 @@
     virtual uint32_t    getUnderrunFrames() const { return 0; }
 
 private:
+    status_t            updateStateWithLoop(StaticAudioTrackState *localState,
+                                            const StaticAudioTrackState &update) const;
+    status_t            updateStateWithPosition(StaticAudioTrackState *localState,
+                                                const StaticAudioTrackState &update) const;
     ssize_t             pollPosition(); // poll for state queue update, and return current position
     StaticAudioTrackSingleStateQueue::Observer  mObserver;
-    size_t              mPosition;  // server's current play position in frames, relative to 0
-
+    StaticAudioTrackPosLoopQueue::Mutator       mPosLoopMutator;
     size_t              mFramesReadySafe; // Assuming size_t read/writes are atomic on 32 / 64 bit
                                           // processors, this is a thread-safe version of
                                           // mFramesReady.
@@ -459,7 +526,8 @@
                                           // can cause a track to appear to have a large number
                                           // of frames. INT64_MAX means an infinite loop.
     bool                mFramesReadyIsCalledByMultipleThreads;
-    StaticAudioTrackState   mState;
+    StaticAudioTrackState mState;         // Server side state. Any updates from client must be
+                                          // passed by the mObserver SingleStateQueue.
 };
 
 // Proxy used by AudioFlinger for servicing AudioRecord
diff --git a/include/private/media/StaticAudioTrackState.h b/include/private/media/StaticAudioTrackState.h
deleted file mode 100644
index d483061..0000000
--- a/include/private/media/StaticAudioTrackState.h
+++ /dev/null
@@ -1,39 +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.
- */
-
-#ifndef STATIC_AUDIO_TRACK_STATE_H
-#define STATIC_AUDIO_TRACK_STATE_H
-
-namespace android {
-
-// Represents a single state of an AudioTrack that was created in static mode (shared memory buffer
-// supplied by the client).  This state needs to be communicated from the client to server.  As this
-// state is too large to be updated atomically without a mutex, and mutexes aren't allowed here, the
-// state is wrapped by a SingleStateQueue.
-struct StaticAudioTrackState {
-    // do not define constructors, destructors, or virtual methods
-
-    // These fields should both be size_t, but since they are located in shared memory we
-    // force to 32-bit.  The client and server may have different typedefs for size_t.
-    uint32_t    mLoopStart;
-    uint32_t    mLoopEnd;
-
-    int         mLoopCount;
-};
-
-}   // namespace android
-
-#endif  // STATIC_AUDIO_TRACK_STATE_H
diff --git a/media/libeffects/factory/EffectsFactory.c b/media/libeffects/factory/EffectsFactory.c
index 6d30d64..c310fe2 100644
--- a/media/libeffects/factory/EffectsFactory.c
+++ b/media/libeffects/factory/EffectsFactory.c
@@ -28,6 +28,7 @@
 
 static list_elem_t *gEffectList; // list of effect_entry_t: all currently created effects
 static list_elem_t *gLibraryList; // list of lib_entry_t: all currently loaded libraries
+static list_elem_t *gSkippedEffects; // list of effects skipped because of duplicate uuid
 // list of effect_descriptor and list of sub effects : all currently loaded
 // It does not contain effects without sub effects.
 static list_sub_elem_t *gSubEffectList;
@@ -63,10 +64,10 @@
                lib_entry_t **lib,
                effect_descriptor_t **desc);
 // To search a subeffect in the gSubEffectList
-int findSubEffect(const effect_uuid_t *uuid,
+static int findSubEffect(const effect_uuid_t *uuid,
                lib_entry_t **lib,
                effect_descriptor_t **desc);
-static void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len);
+static void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len, int indent);
 static int stringToUuid(const char *str, effect_uuid_t *uuid);
 static int uuidToString(const effect_uuid_t *uuid, char *str, size_t maxLen);
 
@@ -237,8 +238,8 @@
     }
 
 #if (LOG_NDEBUG == 0)
-    char str[256];
-    dumpEffectDescriptor(pDescriptor, str, 256);
+    char str[512];
+    dumpEffectDescriptor(pDescriptor, str, sizeof(str), 0 /* indent */);
     ALOGV("EffectQueryEffect() desc:%s", str);
 #endif
     pthread_mutex_unlock(&gLibLock);
@@ -503,15 +504,31 @@
     audio_effect_library_t *desc;
     list_elem_t *e;
     lib_entry_t *l;
+    char path[PATH_MAX];
+    char *str;
+    size_t len;
 
     node = config_find(root, PATH_TAG);
     if (node == NULL) {
         return -EINVAL;
     }
+    // audio_effects.conf always specifies 32 bit lib path: convert to 64 bit path if needed
+    strlcpy(path, node->value, PATH_MAX);
+#ifdef __LP64__
+    str = strstr(path, "/lib/");
+    if (str == NULL)
+        return -EINVAL;
+    len = str - path;
+    path[len] = '\0';
+    strlcat(path, "/lib64/", PATH_MAX);
+    strlcat(path, node->value + len + strlen("/lib/"), PATH_MAX);
+#endif
+    if (strlen(path) >= PATH_MAX - 1)
+        return -EINVAL;
 
-    hdl = dlopen(node->value, RTLD_NOW);
+    hdl = dlopen(path, RTLD_NOW);
     if (hdl == NULL) {
-        ALOGW("loadLibrary() failed to open %s", node->value);
+        ALOGW("loadLibrary() failed to open %s", path);
         goto error;
     }
 
@@ -535,7 +552,7 @@
     // add entry for library in gLibraryList
     l = malloc(sizeof(lib_entry_t));
     l->name = strndup(name, PATH_MAX);
-    l->path = strndup(node->value, PATH_MAX);
+    l->path = strndup(path, PATH_MAX);
     l->handle = hdl;
     l->desc = desc;
     l->effects = NULL;
@@ -547,7 +564,7 @@
     e->next = gLibraryList;
     gLibraryList = e;
     pthread_mutex_unlock(&gLibLock);
-    ALOGV("getLibrary() linked library %p for path %s", l, node->value);
+    ALOGV("getLibrary() linked library %p for path %s", l, path);
 
     return 0;
 
@@ -595,8 +612,8 @@
         return -EINVAL;
     }
 #if (LOG_NDEBUG==0)
-    char s[256];
-    dumpEffectDescriptor(d, s, 256);
+    char s[512];
+    dumpEffectDescriptor(d, s, sizeof(s), 0 /* indent */);
     ALOGV("addSubEffect() read descriptor %p:%s",d, s);
 #endif
     if (EFFECT_API_VERSION_MAJOR(d->apiVersion) !=
@@ -660,6 +677,13 @@
         ALOGW("loadEffect() invalid uuid %s", node->value);
         return -EINVAL;
     }
+    lib_entry_t *tmp;
+    bool skip = false;
+    if (findEffect(NULL, &uuid, &tmp, NULL) == 0) {
+        ALOGW("skipping duplicate uuid %s %s", node->value,
+                node->next ? "and its sub-effects" : "");
+        skip = true;
+    }
 
     d = malloc(sizeof(effect_descriptor_t));
     if (l->desc->get_descriptor(&uuid, d) != 0) {
@@ -670,8 +694,8 @@
         return -EINVAL;
     }
 #if (LOG_NDEBUG==0)
-    char s[256];
-    dumpEffectDescriptor(d, s, 256);
+    char s[512];
+    dumpEffectDescriptor(d, s, sizeof(s), 0 /* indent */);
     ALOGV("loadEffect() read descriptor %p:%s",d, s);
 #endif
     if (EFFECT_API_VERSION_MAJOR(d->apiVersion) !=
@@ -682,8 +706,14 @@
     }
     e = malloc(sizeof(list_elem_t));
     e->object = d;
-    e->next = l->effects;
-    l->effects = e;
+    if (skip) {
+        e->next = gSkippedEffects;
+        gSkippedEffects = e;
+        return -EINVAL;
+    } else {
+        e->next = l->effects;
+        l->effects = e;
+    }
 
     // After the UUID node in the config_tree, if node->next is valid,
     // that would be sub effect node.
@@ -876,22 +906,30 @@
     return ret;
 }
 
-void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len) {
+void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len, int indent) {
     char s[256];
+    char ss[256];
+    char idt[indent + 1];
 
-    snprintf(str, len, "\nEffect Descriptor %p:\n", desc);
-    strncat(str, "- TYPE: ", len);
-    uuidToString(&desc->uuid, s, 256);
-    snprintf(str, len, "- UUID: %s\n", s);
-    uuidToString(&desc->type, s, 256);
-    snprintf(str, len, "- TYPE: %s\n", s);
-    sprintf(s, "- apiVersion: %08X\n- flags: %08X\n",
-            desc->apiVersion, desc->flags);
-    strncat(str, s, len);
-    sprintf(s, "- name: %s\n", desc->name);
-    strncat(str, s, len);
-    sprintf(s, "- implementor: %s\n", desc->implementor);
-    strncat(str, s, len);
+    memset(idt, ' ', indent);
+    idt[indent] = 0;
+
+    str[0] = 0;
+
+    snprintf(s, sizeof(s), "%s%s / %s\n", idt, desc->name, desc->implementor);
+    strlcat(str, s, len);
+
+    uuidToString(&desc->uuid, s, sizeof(s));
+    snprintf(ss, sizeof(ss), "%s  UUID: %s\n", idt, s);
+    strlcat(str, ss, len);
+
+    uuidToString(&desc->type, s, sizeof(s));
+    snprintf(ss, sizeof(ss), "%s  TYPE: %s\n", idt, s);
+    strlcat(str, ss, len);
+
+    sprintf(s, "%s  apiVersion: %08X\n%s  flags: %08X\n", idt,
+            desc->apiVersion, idt, desc->flags);
+    strlcat(str, s, len);
 }
 
 int stringToUuid(const char *str, effect_uuid_t *uuid)
@@ -934,3 +972,40 @@
     return 0;
 }
 
+int EffectDumpEffects(int fd) {
+    char s[512];
+    list_elem_t *e = gLibraryList;
+    lib_entry_t *l = NULL;
+    effect_descriptor_t *d = NULL;
+    int found = 0;
+    int ret = 0;
+
+    while (e) {
+        l = (lib_entry_t *)e->object;
+        list_elem_t *efx = l->effects;
+        dprintf(fd, "Library %s\n", l->name);
+        if (!efx) {
+            dprintf(fd, "  (no effects)\n");
+        }
+        while (efx) {
+            d = (effect_descriptor_t *)efx->object;
+            dumpEffectDescriptor(d, s, sizeof(s), 2);
+            dprintf(fd, "%s", s);
+            efx = efx->next;
+        }
+        e = e->next;
+    }
+
+    e = gSkippedEffects;
+    if (e) {
+        dprintf(fd, "Skipped effects\n");
+        while(e) {
+            d = (effect_descriptor_t *)e->object;
+            dumpEffectDescriptor(d, s, sizeof(s), 2 /* indent */);
+            dprintf(fd, "%s", s);
+            e = e->next;
+        }
+    }
+    return ret;
+}
+
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index 6c585fb..5378bf2 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -61,15 +61,11 @@
     StringArray.cpp \
     AudioPolicy.cpp
 
-LOCAL_SRC_FILES += ../libnbaio/roundup.c
-
 LOCAL_SHARED_LIBRARIES := \
 	libui liblog libcutils libutils libbinder libsonivox libicuuc libicui18n libexpat \
         libcamera_client libstagefright_foundation \
         libgui libdl libaudioutils libnbaio
 
-LOCAL_STATIC_LIBRARIES += libinstantssq
-
 LOCAL_WHOLE_STATIC_LIBRARIES := libmedia_helper
 
 LOCAL_MODULE:= libmedia
@@ -85,12 +81,3 @@
 
 include $(BUILD_SHARED_LIBRARY)
 
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES += SingleStateQueue.cpp
-LOCAL_CFLAGS += -DSINGLE_STATE_QUEUE_INSTANTIATIONS='"SingleStateQueueInstantiations.cpp"'
-
-LOCAL_MODULE := libinstantssq
-LOCAL_MODULE_TAGS := optional
-
-include $(BUILD_STATIC_LIBRARY)
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 9cae21c..f5a5712 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -499,8 +499,8 @@
 
         OutputDescriptor *outputDesc =  new OutputDescriptor(*desc);
         gOutputs.add(ioHandle, outputDesc);
-        ALOGV("ioConfigChanged() new output samplingRate %u, format %#x channel mask %#x frameCount %zu "
-                "latency %d",
+        ALOGV("ioConfigChanged() new output samplingRate %u, format %#x channel mask %#x "
+                "frameCount %zu latency %d",
                 outputDesc->samplingRate, outputDesc->format, outputDesc->channelMask,
                 outputDesc->frameCount, outputDesc->latency);
         } break;
@@ -523,8 +523,8 @@
         if (param2 == NULL) break;
         desc = (const OutputDescriptor *)param2;
 
-        ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %#x channel mask %#x "
-                "frameCount %zu latency %d",
+        ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %#x "
+                "channel mask %#x frameCount %zu latency %d",
                 ioHandle, desc->samplingRate, desc->format,
                 desc->channelMask, desc->frameCount, desc->latency);
         OutputDescriptor *outputDesc = gOutputs.valueAt(index);
@@ -590,18 +590,22 @@
 
 status_t AudioSystem::setDeviceConnectionState(audio_devices_t device,
                                                audio_policy_dev_state_t state,
-                                               const char *device_address)
+                                               const char *device_address,
+                                               const char *device_name)
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
     const char *address = "";
+    const char *name = "";
 
     if (aps == 0) return PERMISSION_DENIED;
 
     if (device_address != NULL) {
         address = device_address;
     }
-
-    return aps->setDeviceConnectionState(device, state, address);
+    if (device_name != NULL) {
+        name = device_name;
+    }
+    return aps->setDeviceConnectionState(device, state, address, name);
 }
 
 audio_policy_dev_state_t AudioSystem::getDeviceConnectionState(audio_devices_t device,
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 735db5c..d4bacc0 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -33,11 +33,16 @@
 
 #define WAIT_PERIOD_MS                  10
 #define WAIT_STREAM_END_TIMEOUT_SEC     120
-
+static const int kMaxLoopCountNotifications = 32;
 
 namespace android {
 // ---------------------------------------------------------------------------
 
+template <typename T>
+const T &min(const T &x, const T &y) {
+    return x < y ? x : y;
+}
+
 static int64_t convertTimespecToUs(const struct timespec &tv)
 {
     return tv.tv_sec * 1000000ll + tv.tv_nsec / 1000;
@@ -317,12 +322,6 @@
     uint32_t channelCount = audio_channel_count_from_out_mask(channelMask);
     mChannelCount = channelCount;
 
-    // AudioFlinger does not currently support 8-bit data in shared memory
-    if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) {
-        ALOGE("8-bit data in shared memory is not supported");
-        return BAD_VALUE;
-    }
-
     // force direct flag if format is not linear PCM
     // or offload was requested
     if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
@@ -346,12 +345,9 @@
         } else {
             mFrameSize = sizeof(uint8_t);
         }
-        mFrameSizeAF = mFrameSize;
     } else {
         ALOG_ASSERT(audio_is_linear_pcm(format));
         mFrameSize = channelCount * audio_bytes_per_sample(format);
-        mFrameSizeAF = channelCount * audio_bytes_per_sample(
-                format == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : format);
         // createTrack will return an error if PCM format is not supported by server,
         // so no need to check for specific PCM formats here
     }
@@ -420,7 +416,10 @@
     mStatus = NO_ERROR;
     mState = STATE_STOPPED;
     mUserData = user;
-    mLoopPeriod = 0;
+    mLoopCount = 0;
+    mLoopStart = 0;
+    mLoopEnd = 0;
+    mLoopCountNotified = 0;
     mMarkerPosition = 0;
     mMarkerReached = false;
     mNewPosition = 0;
@@ -531,14 +530,12 @@
     // the playback head position will reset to 0, so if a marker is set, we need
     // to activate it again
     mMarkerReached = false;
-#if 0
-    // Force flush if a shared buffer is used otherwise audioflinger
-    // will not stop before end of buffer is reached.
-    // It may be needed to make sure that we stop playback, likely in case looping is on.
+
     if (mSharedBuffer != 0) {
-        flush_l();
+        // clear buffer position and loop count.
+        mStaticProxy->setBufferPositionAndLoop(0 /* position */,
+                0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */);
     }
-#endif
 
     sp<AudioTrackThread> t = mAudioTrackThread;
     if (t != 0) {
@@ -740,10 +737,15 @@
 
 void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
 {
-    // Setting the loop will reset next notification update period (like setPosition).
-    mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
-    mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0;
+    // We do not update the periodic notification point.
+    // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
+    mLoopCount = loopCount;
+    mLoopEnd = loopEnd;
+    mLoopStart = loopStart;
+    mLoopCountNotified = loopCount;
     mStaticProxy->setLoop(loopStart, loopEnd, loopCount);
+
+    // Waking the AudioTrackThread is not needed as this cannot be called when active.
 }
 
 status_t AudioTrack::setMarkerPosition(uint32_t marker)
@@ -757,6 +759,10 @@
     mMarkerPosition = marker;
     mMarkerReached = false;
 
+    sp<AudioTrackThread> t = mAudioTrackThread;
+    if (t != 0) {
+        t->wake();
+    }
     return NO_ERROR;
 }
 
@@ -786,6 +792,10 @@
     mNewPosition = updateAndGetPosition_l() + updatePeriod;
     mUpdatePeriod = updatePeriod;
 
+    sp<AudioTrackThread> t = mAudioTrackThread;
+    if (t != 0) {
+        t->wake();
+    }
     return NO_ERROR;
 }
 
@@ -823,12 +833,11 @@
     if (mState == STATE_ACTIVE) {
         return INVALID_OPERATION;
     }
+    // After setting the position, use full update period before notification.
     mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
-    mLoopPeriod = 0;
-    // FIXME Check whether loops and setting position are incompatible in old code.
-    // If we use setLoop for both purposes we lose the capability to set the position while looping.
-    mStaticProxy->setLoop(position, mFrameCount, 0);
+    mStaticProxy->setBufferPosition(position);
 
+    // Waking the AudioTrackThread is not needed as this cannot be called when active.
     return NO_ERROR;
 }
 
@@ -893,10 +902,18 @@
         return INVALID_OPERATION;
     }
     mNewPosition = mUpdatePeriod;
-    mLoopPeriod = 0;
-    // FIXME The new code cannot reload while keeping a loop specified.
-    // Need to check how the old code handled this, and whether it's a significant change.
-    mStaticProxy->setLoop(0, mFrameCount, 0);
+    (void) updateAndGetPosition_l();
+    mPosition = 0;
+#if 0
+    // The documentation is not clear on the behavior of reload() and the restoration
+    // of loop count. Historically we have not restored loop count, start, end,
+    // but it makes sense if one desires to repeat playing a particular sound.
+    if (mLoopCount != 0) {
+        mLoopCountNotified = mLoopCount;
+        mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount);
+    }
+#endif
+    mStaticProxy->setBufferPosition(0);
     return NO_ERROR;
 }
 
@@ -1019,12 +1036,12 @@
             mNotificationFramesAct = frameCount;
         }
     } else if (mSharedBuffer != 0) {
-
-        // Ensure that buffer alignment matches channel count
-        // 8-bit data in shared memory is not currently supported by AudioFlinger
-        size_t alignment = audio_bytes_per_sample(
-                mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat);
+        // FIXME: Ensure client side memory buffers need
+        // not have additional alignment beyond sample
+        // (e.g. 16 bit stereo accessed as 32 bit frame).
+        size_t alignment = audio_bytes_per_sample(mFormat);
         if (alignment & 1) {
+            // for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java).
             alignment = 1;
         }
         if (mChannelCount > 1) {
@@ -1042,7 +1059,7 @@
         // there's no frameCount parameter.
         // But when initializing a shared buffer AudioTrack via set(),
         // there _is_ a frameCount parameter.  We silently ignore it.
-        frameCount = mSharedBuffer->size() / mFrameSizeAF;
+        frameCount = mSharedBuffer->size() / mFrameSize;
 
     } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) {
 
@@ -1103,10 +1120,7 @@
                                 // but we will still need the original value also
     sp<IAudioTrack> track = audioFlinger->createTrack(streamType,
                                                       mSampleRate,
-                                                      // AudioFlinger only sees 16-bit PCM
-                                                      mFormat == AUDIO_FORMAT_PCM_8_BIT &&
-                                                          !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT) ?
-                                                              AUDIO_FORMAT_PCM_16_BIT : mFormat,
+                                                      mFormat,
                                                       mChannelMask,
                                                       &temp,
                                                       &trackFlags,
@@ -1230,9 +1244,9 @@
     // update proxy
     if (mSharedBuffer == 0) {
         mStaticProxy.clear();
-        mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
+        mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize);
     } else {
-        mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
+        mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize);
         mProxy = mStaticProxy;
     }
 
@@ -1352,7 +1366,7 @@
     } while ((status == DEAD_OBJECT) && (tryCounter-- > 0));
 
     audioBuffer->frameCount = buffer.mFrameCount;
-    audioBuffer->size = buffer.mFrameCount * mFrameSizeAF;
+    audioBuffer->size = buffer.mFrameCount * mFrameSize;
     audioBuffer->raw = buffer.mRaw;
     if (nonContig != NULL) {
         *nonContig = buffer.mNonContig;
@@ -1366,7 +1380,7 @@
         return;
     }
 
-    size_t stepCount = audioBuffer->size / mFrameSizeAF;
+    size_t stepCount = audioBuffer->size / mFrameSize;
     if (stepCount == 0) {
         return;
     }
@@ -1432,14 +1446,8 @@
         }
 
         size_t toWrite;
-        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
-            // Divide capacity by 2 to take expansion into account
-            toWrite = audioBuffer.size >> 1;
-            memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite);
-        } else {
-            toWrite = audioBuffer.size;
-            memcpy(audioBuffer.i8, buffer, toWrite);
-        }
+        toWrite = audioBuffer.size;
+        memcpy(audioBuffer.i8, buffer, toWrite);
         buffer = ((const char *) buffer) + toWrite;
         userSize -= toWrite;
         written += toWrite;
@@ -1559,9 +1567,8 @@
         // that the upper layers can recreate the track
         if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {
             status_t status = restoreTrack_l("processAudioBuffer");
-            mLock.unlock();
-            // Run again immediately, but with a new IAudioTrack
-            return 0;
+            // after restoration, continue below to make sure that the loop and buffer events
+            // are notified because they have been cleared from mCblk->mFlags above.
         }
     }
 
@@ -1610,7 +1617,6 @@
     }
 
     // Cache other fields that will be needed soon
-    uint32_t loopPeriod = mLoopPeriod;
     uint32_t sampleRate = mSampleRate;
     uint32_t notificationFrames = mNotificationFramesAct;
     if (mRefreshRemaining) {
@@ -1622,8 +1628,30 @@
     uint32_t sequence = mSequence;
     sp<AudioTrackClientProxy> proxy = mProxy;
 
+    // Determine the number of new loop callback(s) that will be needed, while locked.
+    int loopCountNotifications = 0;
+    uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END
+
+    if (mLoopCount > 0) {
+        int loopCount;
+        size_t bufferPosition;
+        mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
+        loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition;
+        loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications);
+        mLoopCountNotified = loopCount; // discard any excess notifications
+    } else if (mLoopCount < 0) {
+        // FIXME: We're not accurate with notification count and position with infinite looping
+        // since loopCount from server side will always return -1 (we could decrement it).
+        size_t bufferPosition = mStaticProxy->getBufferPosition();
+        loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0);
+        loopPeriod = mLoopEnd - bufferPosition;
+    } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) {
+        size_t bufferPosition = mStaticProxy->getBufferPosition();
+        loopPeriod = mFrameCount - bufferPosition;
+    }
+
     // These fields don't need to be cached, because they are assigned only by set():
-    //     mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags
+    //     mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFlags
     // mFlags is also assigned by createTrack_l(), but not the bit we care about.
 
     mLock.unlock();
@@ -1662,10 +1690,9 @@
     if (newUnderrun) {
         mCbf(EVENT_UNDERRUN, mUserData, NULL);
     }
-    // FIXME we will miss loops if loop cycle was signaled several times since last call
-    //       to processAudioBuffer()
-    if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) {
+    while (loopCountNotifications > 0) {
         mCbf(EVENT_LOOP_END, mUserData, NULL);
+        --loopCountNotifications;
     }
     if (flags & CBLK_BUFFER_END) {
         mCbf(EVENT_BUFFER_END, mUserData, NULL);
@@ -1701,10 +1728,11 @@
         minFrames = markerPosition - position;
     }
     if (loopPeriod > 0 && loopPeriod < minFrames) {
+        // loopPeriod is already adjusted for actual position.
         minFrames = loopPeriod;
     }
-    if (updatePeriod > 0 && updatePeriod < minFrames) {
-        minFrames = updatePeriod;
+    if (updatePeriod > 0) {
+        minFrames = min(minFrames, uint32_t(newPosition - position));
     }
 
     // If > 0, poll periodically to recover from a stuck server.  A good value is 2.
@@ -1767,13 +1795,6 @@
             }
         }
 
-        // Divide buffer size by 2 to take into account the expansion
-        // due to 8 to 16 bit conversion: the callback must fill only half
-        // of the destination buffer
-        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
-            audioBuffer.size >>= 1;
-        }
-
         size_t reqSize = audioBuffer.size;
         mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
         size_t writtenSize = audioBuffer.size;
@@ -1793,13 +1814,7 @@
             return WAIT_PERIOD_MS * 1000000LL;
         }
 
-        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
-            memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize);
-            audioBuffer.size <<= 1;
-        }
-
-        size_t releasedFrames = audioBuffer.size / mFrameSizeAF;
+        size_t releasedFrames = audioBuffer.size / mFrameSize;
         audioBuffer.frameCount = releasedFrames;
         mRemainingFrames -= releasedFrames;
         if (misalignment >= releasedFrames) {
@@ -1856,7 +1871,11 @@
     }
 
     // save the old static buffer position
-    size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0;
+    size_t bufferPosition = 0;
+    int loopCount = 0;
+    if (mStaticProxy != 0) {
+        mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
+    }
 
     // If a new IAudioTrack is successfully created, createTrack_l() will modify the
     // following member variables: mAudioTrack, mCblkMemory and mCblk.
@@ -1865,30 +1884,26 @@
     result = createTrack_l();
 
     // take the frames that will be lost by track recreation into account in saved position
+    // For streaming tracks, this is the amount we obtained from the user/client
+    // (not the number actually consumed at the server - those are already lost).
     (void) updateAndGetPosition_l();
-    mPosition = mReleased;
+    if (mStaticProxy != 0) {
+        mPosition = mReleased;
+    }
 
     if (result == NO_ERROR) {
-        // continue playback from last known position, but
-        // don't attempt to restore loop after invalidation; it's difficult and not worthwhile
-        if (mStaticProxy != NULL) {
-            mLoopPeriod = 0;
-            mStaticProxy->setLoop(bufferPosition, mFrameCount, 0);
-        }
-        // FIXME How do we simulate the fact that all frames present in the buffer at the time of
-        //       track destruction have been played? This is critical for SoundPool implementation
-        //       This must be broken, and needs to be tested/debugged.
-#if 0
-        // restore write index and set other indexes to reflect empty buffer status
-        if (!strcmp(from, "start")) {
-            // 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) {
-                // restart playback even if buffer is not completely filled.
-                android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags);
+        // Continue playback from last known position and restore loop.
+        if (mStaticProxy != 0) {
+            if (loopCount != 0) {
+                mStaticProxy->setBufferPositionAndLoop(bufferPosition,
+                        mLoopStart, mLoopEnd, loopCount);
+            } else {
+                mStaticProxy->setBufferPosition(bufferPosition);
+                if (bufferPosition == mFrameCount) {
+                    ALOGD("restoring track at end of static buffer");
+                }
             }
         }
-#endif
         if (mState == STATE_ACTIVE) {
             result = mAudioTrack->start();
         }
@@ -2148,8 +2163,8 @@
     case NS_NEVER:
         return false;
     case NS_WHENEVER:
-        // FIXME increase poll interval, or make event-driven
-        ns = 1000000000LL;
+        // Event driven: call wake() when callback notifications conditions change.
+        ns = INT64_MAX;
         // fall through
     default:
         LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns);
@@ -2182,6 +2197,17 @@
     }
 }
 
+void AudioTrack::AudioTrackThread::wake()
+{
+    AutoMutex _l(mMyLock);
+    if (!mPaused && mPausedInt && mPausedNs > 0) {
+        // audio track is active and internally paused with timeout.
+        mIgnoreNextPausedInt = true;
+        mPausedInt = false;
+        mMyCond.signal();
+    }
+}
+
 void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns)
 {
     AutoMutex _l(mMyLock);
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index ff24475..08241e2 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -28,7 +28,21 @@
 // used to clamp a value to size_t.  TODO: move to another file.
 template <typename T>
 size_t clampToSize(T x) {
-    return x > SIZE_MAX ? SIZE_MAX : x < 0 ? 0 : (size_t) x;
+    return sizeof(T) > sizeof(size_t) && x > (T) SIZE_MAX ? SIZE_MAX : x < 0 ? 0 : (size_t) x;
+}
+
+// incrementSequence is used to determine the next sequence value
+// for the loop and position sequence counters.  It should return
+// a value between "other" + 1 and "other" + INT32_MAX, the choice of
+// which needs to be the "least recently used" sequence value for "self".
+// In general, this means (new_self) returned is max(self, other) + 1.
+
+static uint32_t incrementSequence(uint32_t self, uint32_t other) {
+    int32_t diff = self - other;
+    if (diff >= 0 && diff < INT32_MAX) {
+        return self + 1; // we're already ahead of other.
+    }
+    return other + 1; // we're behind, so move just ahead of other.
 }
 
 audio_track_cblk_t::audio_track_cblk_t()
@@ -485,8 +499,11 @@
 StaticAudioTrackClientProxy::StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers,
         size_t frameCount, size_t frameSize)
     : AudioTrackClientProxy(cblk, buffers, frameCount, frameSize),
-      mMutator(&cblk->u.mStatic.mSingleStateQueue), mBufferPosition(0)
+      mMutator(&cblk->u.mStatic.mSingleStateQueue),
+      mPosLoopObserver(&cblk->u.mStatic.mPosLoopQueue)
 {
+    memset(&mState, 0, sizeof(mState));
+    memset(&mPosLoop, 0, sizeof(mPosLoop));
 }
 
 void StaticAudioTrackClientProxy::flush()
@@ -501,30 +518,72 @@
         // FIXME Should return an error status
         return;
     }
-    StaticAudioTrackState newState;
-    newState.mLoopStart = (uint32_t) loopStart;
-    newState.mLoopEnd = (uint32_t) loopEnd;
-    newState.mLoopCount = loopCount;
-    size_t bufferPosition;
-    if (loopCount == 0 || (bufferPosition = getBufferPosition()) >= loopEnd) {
-        bufferPosition = loopStart;
+    mState.mLoopStart = (uint32_t) loopStart;
+    mState.mLoopEnd = (uint32_t) loopEnd;
+    mState.mLoopCount = loopCount;
+    mState.mLoopSequence = incrementSequence(mState.mLoopSequence, mState.mPositionSequence);
+    // set patch-up variables until the mState is acknowledged by the ServerProxy.
+    // observed buffer position and loop count will freeze until then to give the
+    // illusion of a synchronous change.
+    getBufferPositionAndLoopCount(NULL, NULL);
+    // preserve behavior to restart at mState.mLoopStart if position exceeds mState.mLoopEnd.
+    if (mState.mLoopCount != 0 && mPosLoop.mBufferPosition >= mState.mLoopEnd) {
+        mPosLoop.mBufferPosition = mState.mLoopStart;
     }
-    mBufferPosition = bufferPosition; // snapshot buffer position until loop is acknowledged.
-    (void) mMutator.push(newState);
+    mPosLoop.mLoopCount = mState.mLoopCount;
+    (void) mMutator.push(mState);
+}
+
+void StaticAudioTrackClientProxy::setBufferPosition(size_t position)
+{
+    // This can only happen on a 64-bit client
+    if (position > UINT32_MAX) {
+        // FIXME Should return an error status
+        return;
+    }
+    mState.mPosition = (uint32_t) position;
+    mState.mPositionSequence = incrementSequence(mState.mPositionSequence, mState.mLoopSequence);
+    // set patch-up variables until the mState is acknowledged by the ServerProxy.
+    // observed buffer position and loop count will freeze until then to give the
+    // illusion of a synchronous change.
+    if (mState.mLoopCount > 0) {  // only check if loop count is changing
+        getBufferPositionAndLoopCount(NULL, NULL); // get last position
+    }
+    mPosLoop.mBufferPosition = position;
+    if (position >= mState.mLoopEnd) {
+        // no ongoing loop is possible if position is greater than loopEnd.
+        mPosLoop.mLoopCount = 0;
+    }
+    (void) mMutator.push(mState);
+}
+
+void StaticAudioTrackClientProxy::setBufferPositionAndLoop(size_t position, size_t loopStart,
+        size_t loopEnd, int loopCount)
+{
+    setLoop(loopStart, loopEnd, loopCount);
+    setBufferPosition(position);
 }
 
 size_t StaticAudioTrackClientProxy::getBufferPosition()
 {
-    size_t bufferPosition;
-    if (mMutator.ack()) {
-        bufferPosition = (size_t) mCblk->u.mStatic.mBufferPosition;
-        if (bufferPosition > mFrameCount) {
-            bufferPosition = mFrameCount;
-        }
-    } else {
-        bufferPosition = mBufferPosition;
+    getBufferPositionAndLoopCount(NULL, NULL);
+    return mPosLoop.mBufferPosition;
+}
+
+void StaticAudioTrackClientProxy::getBufferPositionAndLoopCount(
+        size_t *position, int *loopCount)
+{
+    if (mMutator.ack() == StaticAudioTrackSingleStateQueue::SSQ_DONE) {
+         if (mPosLoopObserver.poll(mPosLoop)) {
+             ; // a valid mPosLoop should be available if ackDone is true.
+         }
     }
-    return bufferPosition;
+    if (position != NULL) {
+        *position = mPosLoop.mBufferPosition;
+    }
+    if (loopCount != NULL) {
+        *loopCount = mPosLoop.mLoopCount;
+    }
 }
 
 // ---------------------------------------------------------------------------
@@ -560,7 +619,8 @@
             ssize_t filled = rear - newFront;
             // Rather than shutting down on a corrupt flush, just treat it as a full flush
             if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
-                ALOGE("mFlush %#x -> %#x, front %#x, rear %#x, mask %#x, newFront %#x, filled %d=%#x",
+                ALOGE("mFlush %#x -> %#x, front %#x, rear %#x, mask %#x, newFront %#x, "
+                        "filled %d=%#x",
                         mFlush, flush, front, rear, mask, newFront, filled, filled);
                 newFront = rear;
             }
@@ -739,13 +799,12 @@
 StaticAudioTrackServerProxy::StaticAudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers,
         size_t frameCount, size_t frameSize)
     : AudioTrackServerProxy(cblk, buffers, frameCount, frameSize),
-      mObserver(&cblk->u.mStatic.mSingleStateQueue), mPosition(0),
+      mObserver(&cblk->u.mStatic.mSingleStateQueue),
+      mPosLoopMutator(&cblk->u.mStatic.mPosLoopQueue),
       mFramesReadySafe(frameCount), mFramesReady(frameCount),
       mFramesReadyIsCalledByMultipleThreads(false)
 {
-    mState.mLoopStart = 0;
-    mState.mLoopEnd = 0;
-    mState.mLoopCount = 0;
+    memset(&mState, 0, sizeof(mState));
 }
 
 void StaticAudioTrackServerProxy::framesReadyIsCalledByMultipleThreads()
@@ -762,55 +821,97 @@
     return mFramesReadySafe;
 }
 
-ssize_t StaticAudioTrackServerProxy::pollPosition()
+status_t StaticAudioTrackServerProxy::updateStateWithLoop(
+        StaticAudioTrackState *localState, const StaticAudioTrackState &update) const
 {
-    size_t position = mPosition;
-    StaticAudioTrackState state;
-    if (mObserver.poll(state)) {
+    if (localState->mLoopSequence != update.mLoopSequence) {
         bool valid = false;
-        size_t loopStart = state.mLoopStart;
-        size_t loopEnd = state.mLoopEnd;
-        if (state.mLoopCount == 0) {
-            if (loopStart > mFrameCount) {
-                loopStart = mFrameCount;
-            }
-            // ignore loopEnd
-            mPosition = position = loopStart;
-            mFramesReady = mFrameCount - mPosition;
-            mState.mLoopCount = 0;
+        const size_t loopStart = update.mLoopStart;
+        const size_t loopEnd = update.mLoopEnd;
+        size_t position = localState->mPosition;
+        if (update.mLoopCount == 0) {
             valid = true;
-        } else if (state.mLoopCount >= -1) {
+        } else if (update.mLoopCount >= -1) {
             if (loopStart < loopEnd && loopEnd <= mFrameCount &&
                     loopEnd - loopStart >= MIN_LOOP) {
                 // If the current position is greater than the end of the loop
                 // we "wrap" to the loop start. This might cause an audible pop.
                 if (position >= loopEnd) {
-                    mPosition = position = loopStart;
+                    position = loopStart;
                 }
-                if (state.mLoopCount == -1) {
-                    mFramesReady = INT64_MAX;
-                } else {
-                    // mFramesReady is 64 bits to handle the effective number of frames
-                    // that the static audio track contains, including loops.
-                    // TODO: Later consider fixing overflow, but does not seem needed now
-                    // as will not overflow if loopStart and loopEnd are Java "ints".
-                    mFramesReady = int64_t(state.mLoopCount) * (loopEnd - loopStart)
-                            + mFrameCount - mPosition;
-                }
-                mState = state;
                 valid = true;
             }
         }
-        if (!valid || mPosition > mFrameCount) {
+        if (!valid || position > mFrameCount) {
+            return NO_INIT;
+        }
+        localState->mPosition = position;
+        localState->mLoopCount = update.mLoopCount;
+        localState->mLoopEnd = loopEnd;
+        localState->mLoopStart = loopStart;
+        localState->mLoopSequence = update.mLoopSequence;
+    }
+    return OK;
+}
+
+status_t StaticAudioTrackServerProxy::updateStateWithPosition(
+        StaticAudioTrackState *localState, const StaticAudioTrackState &update) const
+{
+    if (localState->mPositionSequence != update.mPositionSequence) {
+        if (update.mPosition > mFrameCount) {
+            return NO_INIT;
+        } else if (localState->mLoopCount != 0 && update.mPosition >= localState->mLoopEnd) {
+            localState->mLoopCount = 0; // disable loop count if position is beyond loop end.
+        }
+        localState->mPosition = update.mPosition;
+        localState->mPositionSequence = update.mPositionSequence;
+    }
+    return OK;
+}
+
+ssize_t StaticAudioTrackServerProxy::pollPosition()
+{
+    StaticAudioTrackState state;
+    if (mObserver.poll(state)) {
+        StaticAudioTrackState trystate = mState;
+        bool result;
+        const int32_t diffSeq = state.mLoopSequence - state.mPositionSequence;
+
+        if (diffSeq < 0) {
+            result = updateStateWithLoop(&trystate, state) == OK &&
+                    updateStateWithPosition(&trystate, state) == OK;
+        } else {
+            result = updateStateWithPosition(&trystate, state) == OK &&
+                    updateStateWithLoop(&trystate, state) == OK;
+        }
+        if (!result) {
+            mObserver.done();
+            // caution: no update occurs so server state will be inconsistent with client state.
             ALOGE("%s client pushed an invalid state, shutting down", __func__);
             mIsShutdown = true;
             return (ssize_t) NO_INIT;
         }
+        mState = trystate;
+        if (mState.mLoopCount == -1) {
+            mFramesReady = INT64_MAX;
+        } else if (mState.mLoopCount == 0) {
+            mFramesReady = mFrameCount - mState.mPosition;
+        } else if (mState.mLoopCount > 0) {
+            // TODO: Later consider fixing overflow, but does not seem needed now
+            // as will not overflow if loopStart and loopEnd are Java "ints".
+            mFramesReady = int64_t(mState.mLoopCount) * (mState.mLoopEnd - mState.mLoopStart)
+                    + mFrameCount - mState.mPosition;
+        }
         mFramesReadySafe = clampToSize(mFramesReady);
         // This may overflow, but client is not supposed to rely on it
-        mCblk->u.mStatic.mBufferPosition = (uint32_t) position;
+        StaticAudioTrackPosLoop posLoop;
+
+        posLoop.mLoopCount = (int32_t) mState.mLoopCount;
+        posLoop.mBufferPosition = (uint32_t) mState.mPosition;
+        mPosLoopMutator.push(posLoop);
+        mObserver.done(); // safe to read mStatic variables.
     }
-    return (ssize_t) position;
+    return (ssize_t) mState.mPosition;
 }
 
 status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush __unused)
@@ -849,7 +950,7 @@
     }
     // As mFramesReady is the total remaining frames in the static audio track,
     // it is always larger or equal to avail.
-    LOG_ALWAYS_FATAL_IF(mFramesReady < avail);
+    LOG_ALWAYS_FATAL_IF(mFramesReady < (int64_t) avail);
     buffer->mNonContig = mFramesReady == INT64_MAX ? SIZE_MAX : clampToSize(mFramesReady - avail);
     mUnreleased = avail;
     return NO_ERROR;
@@ -858,7 +959,7 @@
 void StaticAudioTrackServerProxy::releaseBuffer(Buffer* buffer)
 {
     size_t stepCount = buffer->mFrameCount;
-    LOG_ALWAYS_FATAL_IF(!(stepCount <= mFramesReady));
+    LOG_ALWAYS_FATAL_IF(!((int64_t) stepCount <= mFramesReady));
     LOG_ALWAYS_FATAL_IF(!(stepCount <= mUnreleased));
     if (stepCount == 0) {
         // prevent accidental re-use of buffer
@@ -868,11 +969,12 @@
     }
     mUnreleased -= stepCount;
     audio_track_cblk_t* cblk = mCblk;
-    size_t position = mPosition;
+    size_t position = mState.mPosition;
     size_t newPosition = position + stepCount;
     int32_t setFlags = 0;
     if (!(position <= newPosition && newPosition <= mFrameCount)) {
-        ALOGW("%s newPosition %zu outside [%zu, %zu]", __func__, newPosition, position, mFrameCount);
+        ALOGW("%s newPosition %zu outside [%zu, %zu]", __func__, newPosition, position,
+                mFrameCount);
         newPosition = mFrameCount;
     } else if (mState.mLoopCount != 0 && newPosition == mState.mLoopEnd) {
         newPosition = mState.mLoopStart;
@@ -885,7 +987,7 @@
     if (newPosition == mFrameCount) {
         setFlags |= CBLK_BUFFER_END;
     }
-    mPosition = newPosition;
+    mState.mPosition = newPosition;
     if (mFramesReady != INT64_MAX) {
         mFramesReady -= stepCount;
     }
@@ -893,7 +995,10 @@
 
     cblk->mServer += stepCount;
     // This may overflow, but client is not supposed to rely on it
-    cblk->u.mStatic.mBufferPosition = (uint32_t) newPosition;
+    StaticAudioTrackPosLoop posLoop;
+    posLoop.mBufferPosition = mState.mPosition;
+    posLoop.mLoopCount = mState.mLoopCount;
+    mPosLoopMutator.push(posLoop);
     if (setFlags != 0) {
         (void) android_atomic_or(setFlags, &cblk->mFlags);
         // this would be a good place to wake a futex
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index dbc7a9e..f2ff27b 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -73,6 +73,8 @@
     REGISTER_POLICY_MIXES,
 };
 
+#define MAX_ITEMS_PER_LIST 1024
+
 class BpAudioPolicyService : public BpInterface<IAudioPolicyService>
 {
 public:
@@ -84,13 +86,15 @@
     virtual status_t setDeviceConnectionState(
                                     audio_devices_t device,
                                     audio_policy_dev_state_t state,
-                                    const char *device_address)
+                                    const char *device_address,
+                                    const char *device_name)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
         data.writeInt32(static_cast <uint32_t>(device));
         data.writeInt32(static_cast <uint32_t>(state));
         data.writeCString(device_address);
+        data.writeCString(device_name);
         remote()->transact(SET_DEVICE_CONNECTION_STATE, data, &reply);
         return static_cast <status_t> (reply.readInt32());
     }
@@ -726,9 +730,11 @@
             audio_policy_dev_state_t state =
                     static_cast <audio_policy_dev_state_t>(data.readInt32());
             const char *device_address = data.readCString();
+            const char *device_name = data.readCString();
             reply->writeInt32(static_cast<uint32_t> (setDeviceConnectionState(device,
                                                                               state,
-                                                                              device_address)));
+                                                                              device_address,
+                                                                              device_name)));
             return NO_ERROR;
         } break;
 
@@ -1054,10 +1060,18 @@
             audio_port_role_t role = (audio_port_role_t)data.readInt32();
             audio_port_type_t type = (audio_port_type_t)data.readInt32();
             unsigned int numPortsReq = data.readInt32();
+            if (numPortsReq > MAX_ITEMS_PER_LIST) {
+                numPortsReq = MAX_ITEMS_PER_LIST;
+            }
             unsigned int numPorts = numPortsReq;
-            unsigned int generation;
             struct audio_port *ports =
                     (struct audio_port *)calloc(numPortsReq, sizeof(struct audio_port));
+            if (ports == NULL) {
+                reply->writeInt32(NO_MEMORY);
+                reply->writeInt32(0);
+                return NO_ERROR;
+            }
+            unsigned int generation;
             status_t status = listAudioPorts(role, type, &numPorts, ports, &generation);
             reply->writeInt32(status);
             reply->writeInt32(numPorts);
@@ -1111,11 +1125,19 @@
         case LIST_AUDIO_PATCHES: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
             unsigned int numPatchesReq = data.readInt32();
+            if (numPatchesReq > MAX_ITEMS_PER_LIST) {
+                numPatchesReq = MAX_ITEMS_PER_LIST;
+            }
             unsigned int numPatches = numPatchesReq;
-            unsigned int generation;
             struct audio_patch *patches =
                     (struct audio_patch *)calloc(numPatchesReq,
                                                  sizeof(struct audio_patch));
+            if (patches == NULL) {
+                reply->writeInt32(NO_MEMORY);
+                reply->writeInt32(0);
+                return NO_ERROR;
+            }
+            unsigned int generation;
             status_t status = listAudioPatches(&numPatches, patches, &generation);
             reply->writeInt32(status);
             reply->writeInt32(numPatches);
diff --git a/media/libmedia/IMediaRecorder.cpp b/media/libmedia/IMediaRecorder.cpp
index a733b68..9181f86 100644
--- a/media/libmedia/IMediaRecorder.cpp
+++ b/media/libmedia/IMediaRecorder.cpp
@@ -46,7 +46,6 @@
     SET_OUTPUT_FORMAT,
     SET_VIDEO_ENCODER,
     SET_AUDIO_ENCODER,
-    SET_OUTPUT_FILE_PATH,
     SET_OUTPUT_FILE_FD,
     SET_VIDEO_SIZE,
     SET_VIDEO_FRAMERATE,
@@ -158,16 +157,6 @@
         return reply.readInt32();
     }
 
-    status_t setOutputFile(const char* path)
-    {
-        ALOGV("setOutputFile(%s)", path);
-        Parcel data, reply;
-        data.writeInterfaceToken(IMediaRecorder::getInterfaceDescriptor());
-        data.writeCString(path);
-        remote()->transact(SET_OUTPUT_FILE_PATH, data, &reply);
-        return reply.readInt32();
-    }
-
     status_t setOutputFile(int fd, int64_t offset, int64_t length) {
         ALOGV("setOutputFile(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
         Parcel data, reply;
@@ -300,7 +289,8 @@
 // ----------------------------------------------------------------------
 
 status_t BnMediaRecorder::onTransact(
-                                     uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+                                     uint32_t code, const Parcel& data, Parcel* reply,
+                                     uint32_t flags)
 {
     switch (code) {
         case RELEASE: {
@@ -390,13 +380,6 @@
             return NO_ERROR;
 
         } break;
-        case SET_OUTPUT_FILE_PATH: {
-            ALOGV("SET_OUTPUT_FILE_PATH");
-            CHECK_INTERFACE(IMediaRecorder, data, reply);
-            const char* path = data.readCString();
-            reply->writeInt32(setOutputFile(path));
-            return NO_ERROR;
-        } break;
         case SET_OUTPUT_FILE_FD: {
             ALOGV("SET_OUTPUT_FILE_FD");
             CHECK_INTERFACE(IMediaRecorder, data, reply);
@@ -445,7 +428,8 @@
         case SET_PREVIEW_SURFACE: {
             ALOGV("SET_PREVIEW_SURFACE");
             CHECK_INTERFACE(IMediaRecorder, data, reply);
-            sp<IGraphicBufferProducer> surface = interface_cast<IGraphicBufferProducer>(data.readStrongBinder());
+            sp<IGraphicBufferProducer> surface = interface_cast<IGraphicBufferProducer>(
+                    data.readStrongBinder());
             reply->writeInt32(setPreviewSurface(surface));
             return NO_ERROR;
         } break;
diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp
index 721d8d7..271be0c 100644
--- a/media/libmedia/JetPlayer.cpp
+++ b/media/libmedia/JetPlayer.cpp
@@ -408,7 +408,8 @@
     ALOGV("JetPlayer::queueSegment segmentNum=%d, libNum=%d, repeatCount=%d, transpose=%d",
         segmentNum, libNum, repeatCount, transpose);
     Mutex::Autolock lock(mMutex);
-    return JET_QueueSegment(mEasData, segmentNum, libNum, repeatCount, transpose, muteFlags, userID);
+    return JET_QueueSegment(mEasData, segmentNum, libNum, repeatCount, transpose, muteFlags,
+            userID);
 }
 
 //-------------------------------------------------------------------------------------------------
@@ -449,7 +450,8 @@
 void JetPlayer::dumpJetStatus(S_JET_STATUS* pJetStatus)
 {
     if (pJetStatus!=NULL)
-        ALOGV(">> current JET player status: userID=%d segmentRepeatCount=%d numQueuedSegments=%d paused=%d",
+        ALOGV(">> current JET player status: userID=%d segmentRepeatCount=%d numQueuedSegments=%d "
+                "paused=%d",
                 pJetStatus->currentUserID, pJetStatus->segmentRepeatCount,
                 pJetStatus->numQueuedSegments, pJetStatus->paused);
     else
diff --git a/media/libmedia/MediaProfiles.cpp b/media/libmedia/MediaProfiles.cpp
index e2e6042..47f9258 100644
--- a/media/libmedia/MediaProfiles.cpp
+++ b/media/libmedia/MediaProfiles.cpp
@@ -163,7 +163,8 @@
 }
 
 /*static*/ int
-MediaProfiles::findTagForName(const MediaProfiles::NameToTagMap *map, size_t nMappings, const char *name)
+MediaProfiles::findTagForName(const MediaProfiles::NameToTagMap *map, size_t nMappings,
+        const char *name)
 {
     int tag = -1;
     for (size_t i = 0; i < nMappings; ++i) {
@@ -295,9 +296,8 @@
     CHECK(codec != -1);
 
     MediaProfiles::AudioEncoderCap *cap =
-        new MediaProfiles::AudioEncoderCap(static_cast<audio_encoder>(codec), atoi(atts[5]), atoi(atts[7]),
-            atoi(atts[9]), atoi(atts[11]), atoi(atts[13]),
-            atoi(atts[15]));
+        new MediaProfiles::AudioEncoderCap(static_cast<audio_encoder>(codec), atoi(atts[5]),
+            atoi(atts[7]), atoi(atts[9]), atoi(atts[11]), atoi(atts[13]), atoi(atts[15]));
     logAudioEncoderCap(*cap);
     return cap;
 }
@@ -330,7 +330,8 @@
           !strcmp("fileFormat", atts[2]) &&
           !strcmp("duration",   atts[4]));
 
-    const size_t nProfileMappings = sizeof(sCamcorderQualityNameMap)/sizeof(sCamcorderQualityNameMap[0]);
+    const size_t nProfileMappings = sizeof(sCamcorderQualityNameMap)/
+            sizeof(sCamcorderQualityNameMap[0]);
     const int quality = findTagForName(sCamcorderQualityNameMap, nProfileMappings, atts[1]);
     CHECK(quality != -1);
 
@@ -722,16 +723,20 @@
 MediaProfiles::createDefaultCamcorderTimeLapseLowProfiles(
         MediaProfiles::CamcorderProfile **lowTimeLapseProfile,
         MediaProfiles::CamcorderProfile **lowSpecificTimeLapseProfile) {
-    *lowTimeLapseProfile = createDefaultCamcorderTimeLapseQcifProfile(CAMCORDER_QUALITY_TIME_LAPSE_LOW);
-    *lowSpecificTimeLapseProfile = createDefaultCamcorderTimeLapseQcifProfile(CAMCORDER_QUALITY_TIME_LAPSE_QCIF);
+    *lowTimeLapseProfile = createDefaultCamcorderTimeLapseQcifProfile(
+            CAMCORDER_QUALITY_TIME_LAPSE_LOW);
+    *lowSpecificTimeLapseProfile = createDefaultCamcorderTimeLapseQcifProfile(
+            CAMCORDER_QUALITY_TIME_LAPSE_QCIF);
 }
 
 /*static*/ void
 MediaProfiles::createDefaultCamcorderTimeLapseHighProfiles(
         MediaProfiles::CamcorderProfile **highTimeLapseProfile,
         MediaProfiles::CamcorderProfile **highSpecificTimeLapseProfile) {
-    *highTimeLapseProfile = createDefaultCamcorderTimeLapse480pProfile(CAMCORDER_QUALITY_TIME_LAPSE_HIGH);
-    *highSpecificTimeLapseProfile = createDefaultCamcorderTimeLapse480pProfile(CAMCORDER_QUALITY_TIME_LAPSE_480P);
+    *highTimeLapseProfile = createDefaultCamcorderTimeLapse480pProfile(
+            CAMCORDER_QUALITY_TIME_LAPSE_HIGH);
+    *highSpecificTimeLapseProfile = createDefaultCamcorderTimeLapse480pProfile(
+            CAMCORDER_QUALITY_TIME_LAPSE_480P);
 }
 
 /*static*/ MediaProfiles::CamcorderProfile*
@@ -809,7 +814,8 @@
 
     // high camcorder time lapse profiles.
     MediaProfiles::CamcorderProfile *highTimeLapseProfile, *highSpecificTimeLapseProfile;
-    createDefaultCamcorderTimeLapseHighProfiles(&highTimeLapseProfile, &highSpecificTimeLapseProfile);
+    createDefaultCamcorderTimeLapseHighProfiles(&highTimeLapseProfile,
+            &highSpecificTimeLapseProfile);
     profiles->mCamcorderProfiles.add(highTimeLapseProfile);
     profiles->mCamcorderProfiles.add(highSpecificTimeLapseProfile);
 
diff --git a/media/libmedia/SingleStateQueue.cpp b/media/libmedia/SingleStateQueue.cpp
deleted file mode 100644
index c241184..0000000
--- a/media/libmedia/SingleStateQueue.cpp
+++ /dev/null
@@ -1,106 +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 <new>
-#include <cutils/atomic.h>
-#include <media/SingleStateQueue.h>
-
-namespace android {
-
-template<typename T> SingleStateQueue<T>::Mutator::Mutator(Shared *shared)
-    : mSequence(0), mShared((Shared *) shared)
-{
-    // exactly one of Mutator and Observer must initialize, currently it is Observer
-    //shared->init();
-}
-
-template<typename T> int32_t SingleStateQueue<T>::Mutator::push(const T& value)
-{
-    Shared *shared = mShared;
-    int32_t sequence = mSequence;
-    sequence++;
-    android_atomic_acquire_store(sequence, &shared->mSequence);
-    shared->mValue = value;
-    sequence++;
-    android_atomic_release_store(sequence, &shared->mSequence);
-    mSequence = sequence;
-    // consider signalling a futex here, if we know that observer is waiting
-    return sequence;
-}
-
-template<typename T> bool SingleStateQueue<T>::Mutator::ack()
-{
-    return mShared->mAck - mSequence == 0;
-}
-
-template<typename T> bool SingleStateQueue<T>::Mutator::ack(int32_t sequence)
-{
-    // this relies on 2's complement rollover to detect an ancient sequence number
-    return mShared->mAck - sequence >= 0;
-}
-
-template<typename T> SingleStateQueue<T>::Observer::Observer(Shared *shared)
-    : mSequence(0), mSeed(1), mShared((Shared *) shared)
-{
-    // exactly one of Mutator and Observer must initialize, currently it is Observer
-    shared->init();
-}
-
-template<typename T> bool SingleStateQueue<T>::Observer::poll(T& value)
-{
-    Shared *shared = mShared;
-    int32_t before = shared->mSequence;
-    if (before == mSequence) {
-        return false;
-    }
-    for (int tries = 0; ; ) {
-        const int MAX_TRIES = 5;
-        if (before & 1) {
-            if (++tries >= MAX_TRIES) {
-                return false;
-            }
-            before = shared->mSequence;
-        } else {
-            android_memory_barrier();
-            T temp = shared->mValue;
-            int32_t after = android_atomic_release_load(&shared->mSequence);
-            if (after == before) {
-                value = temp;
-                shared->mAck = before;
-                mSequence = before;
-                return true;
-            }
-            if (++tries >= MAX_TRIES) {
-                return false;
-            }
-            before = after;
-        }
-    }
-}
-
-#if 0
-template<typename T> SingleStateQueue<T>::SingleStateQueue(void /*Shared*/ *shared)
-{
-    ((Shared *) shared)->init();
-}
-#endif
-
-}   // namespace android
-
-// hack for gcc
-#ifdef SINGLE_STATE_QUEUE_INSTANTIATIONS
-#include SINGLE_STATE_QUEUE_INSTANTIATIONS
-#endif
diff --git a/media/libmedia/SingleStateQueueInstantiations.cpp b/media/libmedia/SingleStateQueueInstantiations.cpp
deleted file mode 100644
index 0265c8c..0000000
--- a/media/libmedia/SingleStateQueueInstantiations.cpp
+++ /dev/null
@@ -1,28 +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 <media/SingleStateQueue.h>
-#include <private/media/StaticAudioTrackState.h>
-#include <media/AudioTimestamp.h>
-
-// FIXME hack for gcc
-
-namespace android {
-
-template class SingleStateQueue<StaticAudioTrackState>; // typedef StaticAudioTrackSingleStateQueue
-template class SingleStateQueue<AudioTimestamp>;        // typedef AudioTimestampSingleStateQueue
-
-}
diff --git a/media/libmedia/StringArray.cpp b/media/libmedia/StringArray.cpp
index 5f5b57a..477e3fd 100644
--- a/media/libmedia/StringArray.cpp
+++ b/media/libmedia/StringArray.cpp
@@ -16,7 +16,7 @@
 
 //
 // Sortable array of strings.  STL-ish, but STL-free.
-//  
+//
 
 #include <stdlib.h>
 #include <string.h>
diff --git a/media/libmedia/docs/Makefile b/media/libmedia/docs/Makefile
new file mode 100644
index 0000000..bddbc9b
--- /dev/null
+++ b/media/libmedia/docs/Makefile
@@ -0,0 +1,2 @@
+paused.png : paused.dot
+	dot -Tpng < $< > $@
diff --git a/media/libmedia/docs/paused.dot b/media/libmedia/docs/paused.dot
new file mode 100644
index 0000000..11e1777
--- /dev/null
+++ b/media/libmedia/docs/paused.dot
@@ -0,0 +1,85 @@
+digraph paused {
+initial [label="INITIAL\n\
+mIgnoreNextPausedInt = false\n\
+mPaused = false\n\
+mPausedInt = false"];
+
+resume_body [label="mIgnoreNextPausedInt = true\nif (mPaused || mPausedInt)"];
+resume_paused [label="mPaused = false\nmPausedInt = false\nsignal()"];
+resume_paused -> resume_merged;
+resume_merged [label="return"];
+
+Application -> ATstop;
+ATstop [label="AudioTrack::stop()"];
+ATstop -> pause;
+Application -> ATpause;
+ATpause [label="AudioTrack::pause()"];
+ATpause -> pause;
+ATstart -> resume;
+ATstart [label="AudioTrack::start()"];
+destructor [label="~AudioTrack()"];
+destructor -> requestExit;
+requestExit [label="AudioTrackThread::requestExit()"];
+requestExit -> resume;
+Application -> ATsetMarkerPosition
+ATsetMarkerPosition [label="AudioTrack::setMarkerPosition()\n[sets marker variables]"];
+ATsetMarkerPosition -> ATTwake
+Application -> ATsetPositionUpdatePeriod
+ATsetPositionUpdatePeriod [label="AudioTrack::setPositionUpdatePeriod()\n[sets update period variables]"];
+ATsetPositionUpdatePeriod -> ATTwake
+Application -> ATstart;
+
+resume [label="AudioTrackThread::resume()"];
+resume -> resume_body;
+
+resume_body -> resume_paused [label="true"];
+resume_body -> resume_merged [label="false"];
+
+ATTwake [label="AudioTrackThread::wake()\nif (!mPaused && mPausedInt && mPausedNs > 0)"];
+ATTwake-> ATTWake_wakeable [label="true"];
+ATTWake_wakeable [label="mIgnoreNextPausedInt = true\nmPausedInt = false\nsignal()"];
+ATTwake-> ATTWake_cannotwake [label="false"]
+ATTWake_cannotwake [label="ignore"];
+
+pause [label="mPaused = true"];
+pause -> return;
+
+threadLoop [label="AudioTrackThread::threadLoop()\nENTRY"];
+threadLoop -> threadLoop_1;
+threadLoop_1 [label="if (mPaused)"];
+threadLoop_1 -> threadLoop_1_true [label="true"];
+threadLoop_1 -> threadLoop_2 [label="false"];
+threadLoop_1_true [label="wait()\nreturn true"];
+threadLoop_2 [label="if (mIgnoreNextPausedInt)"];
+threadLoop_2 -> threadLoop_2_true [label="true"];
+threadLoop_2 -> threadLoop_3 [label="false"];
+threadLoop_2_true [label="mIgnoreNextPausedInt = false\nmPausedInt = false"];
+threadLoop_2_true -> threadLoop_3;
+threadLoop_3 [label="if (mPausedInt)"];
+threadLoop_3 -> threadLoop_3_true [label="true"];
+threadLoop_3 -> threadLoop_4 [label="false"];
+threadLoop_3_true [label="wait()\nmPausedInt = false\nreturn true"];
+threadLoop_4 [label="if (exitPending)"];
+threadLoop_4 -> threadLoop_4_true [label="true"];
+threadLoop_4 -> threadLoop_5 [label="false"];
+threadLoop_4_true [label="return false"];
+threadLoop_5 [label="ns = processAudioBuffer()"];
+threadLoop_5 -> threadLoop_6;
+threadLoop_6 [label="case ns"];
+threadLoop_6 -> threadLoop_6_0 [label="0"];
+threadLoop_6 -> threadLoop_6_NS_INACTIVE [label="NS_INACTIVE"];
+threadLoop_6 -> threadLoop_6_NS_NEVER [label="NS_NEVER"];
+threadLoop_6 -> threadLoop_6_NS_WHENEVER [label="NS_WHENEVER"];
+threadLoop_6 -> threadLoop_6_default [label="default"];
+threadLoop_6_default [label="if (ns < 0)"];
+threadLoop_6_default -> threadLoop_6_default_true [label="true"];
+threadLoop_6_default -> threadLoop_6_default_false [label="false"];
+threadLoop_6_default_true [label="FATAL"];
+threadLoop_6_default_false [label="pauseInternal(ns) [wake()-able]\nmPausedInternal = true\nmPausedNs = ns\nreturn true"];
+threadLoop_6_0 [label="return true"];
+threadLoop_6_NS_INACTIVE [label="pauseInternal()\nmPausedInternal = true\nmPausedNs = 0\nreturn true"];
+threadLoop_6_NS_NEVER [label="return false"];
+threadLoop_6_NS_WHENEVER [label="ns = 1s"];
+threadLoop_6_NS_WHENEVER -> threadLoop_6_default_false;
+
+}
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 05c89ed..432ecda 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -240,7 +240,7 @@
 // must call with lock held
 status_t MediaPlayer::prepareAsync_l()
 {
-    if ( (mPlayer != 0) && ( mCurrentState & ( MEDIA_PLAYER_INITIALIZED | MEDIA_PLAYER_STOPPED) ) ) {
+    if ( (mPlayer != 0) && ( mCurrentState & (MEDIA_PLAYER_INITIALIZED | MEDIA_PLAYER_STOPPED) ) ) {
         mPlayer->setAudioStreamType(mStreamType);
         if (mAudioAttributesParcel != NULL) {
             mPlayer->setParameter(KEY_PARAMETER_AUDIO_ATTRIBUTES, *mAudioAttributesParcel);
@@ -414,7 +414,8 @@
 status_t MediaPlayer::getDuration_l(int *msec)
 {
     ALOGV("getDuration_l");
-    bool isValidState = (mCurrentState & (MEDIA_PLAYER_PREPARED | MEDIA_PLAYER_STARTED | MEDIA_PLAYER_PAUSED | MEDIA_PLAYER_STOPPED | MEDIA_PLAYER_PLAYBACK_COMPLETE));
+    bool isValidState = (mCurrentState & (MEDIA_PLAYER_PREPARED | MEDIA_PLAYER_STARTED |
+            MEDIA_PLAYER_PAUSED | MEDIA_PLAYER_STOPPED | MEDIA_PLAYER_PLAYBACK_COMPLETE));
     if (mPlayer != 0 && isValidState) {
         int durationMs;
         status_t ret = mPlayer->getDuration(&durationMs);
@@ -443,7 +444,8 @@
 status_t MediaPlayer::seekTo_l(int msec)
 {
     ALOGV("seekTo %d", msec);
-    if ((mPlayer != 0) && ( mCurrentState & ( MEDIA_PLAYER_STARTED | MEDIA_PLAYER_PREPARED | MEDIA_PLAYER_PAUSED |  MEDIA_PLAYER_PLAYBACK_COMPLETE) ) ) {
+    if ((mPlayer != 0) && ( mCurrentState & ( MEDIA_PLAYER_STARTED | MEDIA_PLAYER_PREPARED |
+            MEDIA_PLAYER_PAUSED |  MEDIA_PLAYER_PLAYBACK_COMPLETE) ) ) {
         if ( msec < 0 ) {
             ALOGW("Attempt to seek to invalid position: %d", msec);
             msec = 0;
@@ -477,7 +479,8 @@
             return NO_ERROR;
         }
     }
-    ALOGE("Attempt to perform seekTo in wrong state: mPlayer=%p, mCurrentState=%u", mPlayer.get(), mCurrentState);
+    ALOGE("Attempt to perform seekTo in wrong state: mPlayer=%p, mCurrentState=%u", mPlayer.get(),
+            mCurrentState);
     return INVALID_OPERATION;
 }
 
diff --git a/media/libmedia/mediarecorder.cpp b/media/libmedia/mediarecorder.cpp
index 1952b86..973e156 100644
--- a/media/libmedia/mediarecorder.cpp
+++ b/media/libmedia/mediarecorder.cpp
@@ -264,32 +264,6 @@
     return ret;
 }
 
-status_t MediaRecorder::setOutputFile(const char* path)
-{
-    ALOGV("setOutputFile(%s)", path);
-    if (mMediaRecorder == NULL) {
-        ALOGE("media recorder is not initialized yet");
-        return INVALID_OPERATION;
-    }
-    if (mIsOutputFileSet) {
-        ALOGE("output file has already been set");
-        return INVALID_OPERATION;
-    }
-    if (!(mCurrentState & MEDIA_RECORDER_DATASOURCE_CONFIGURED)) {
-        ALOGE("setOutputFile called in an invalid state(%d)", mCurrentState);
-        return INVALID_OPERATION;
-    }
-
-    status_t ret = mMediaRecorder->setOutputFile(path);
-    if (OK != ret) {
-        ALOGV("setOutputFile failed: %d", ret);
-        mCurrentState = MEDIA_RECORDER_ERROR;
-        return ret;
-    }
-    mIsOutputFileSet = true;
-    return ret;
-}
-
 status_t MediaRecorder::setOutputFile(int fd, int64_t offset, int64_t length)
 {
     ALOGV("setOutputFile(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
diff --git a/media/libmediaplayerservice/MediaRecorderClient.cpp b/media/libmediaplayerservice/MediaRecorderClient.cpp
index 194abbb..4d4de9b 100644
--- a/media/libmediaplayerservice/MediaRecorderClient.cpp
+++ b/media/libmediaplayerservice/MediaRecorderClient.cpp
@@ -154,17 +154,6 @@
     return mRecorder->setAudioEncoder((audio_encoder)ae);
 }
 
-status_t MediaRecorderClient::setOutputFile(const char* path)
-{
-    ALOGV("setOutputFile(%s)", path);
-    Mutex::Autolock lock(mLock);
-    if (mRecorder == NULL) {
-        ALOGE("recorder is not initialized");
-        return NO_INIT;
-    }
-    return mRecorder->setOutputFile(path);
-}
-
 status_t MediaRecorderClient::setOutputFile(int fd, int64_t offset, int64_t length)
 {
     ALOGV("setOutputFile(%d, %lld, %lld)", fd, offset, length);
diff --git a/media/libmediaplayerservice/MediaRecorderClient.h b/media/libmediaplayerservice/MediaRecorderClient.h
index a65ec9f..a444b6c 100644
--- a/media/libmediaplayerservice/MediaRecorderClient.h
+++ b/media/libmediaplayerservice/MediaRecorderClient.h
@@ -38,7 +38,6 @@
     virtual     status_t   setOutputFormat(int of);
     virtual     status_t   setVideoEncoder(int ve);
     virtual     status_t   setAudioEncoder(int ae);
-    virtual     status_t   setOutputFile(const char* path);
     virtual     status_t   setOutputFile(int fd, int64_t offset,
                                                   int64_t length);
     virtual     status_t   setVideoSize(int width, int height);
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index 86639cb..2551040 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -206,7 +206,7 @@
 status_t StagefrightRecorder::setVideoFrameRate(int frames_per_second) {
     ALOGV("setVideoFrameRate: %d", frames_per_second);
     if ((frames_per_second <= 0 && frames_per_second != -1) ||
-        frames_per_second > 120) {
+        frames_per_second > kMaxHighSpeedFps) {
         ALOGE("Invalid video frame rate: %d", frames_per_second);
         return BAD_VALUE;
     }
@@ -241,14 +241,6 @@
     return OK;
 }
 
-status_t StagefrightRecorder::setOutputFile(const char * /* path */) {
-    ALOGE("setOutputFile(const char*) must not be called");
-    // We don't actually support this at all, as the media_server process
-    // no longer has permissions to create files.
-
-    return -EPERM;
-}
-
 status_t StagefrightRecorder::setOutputFile(int fd, int64_t offset, int64_t length) {
     ALOGV("setOutputFile: %d, %lld, %lld", fd, offset, length);
     // These don't make any sense, do they?
diff --git a/media/libmediaplayerservice/StagefrightRecorder.h b/media/libmediaplayerservice/StagefrightRecorder.h
index 54c38d3..b5a49d3 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.h
+++ b/media/libmediaplayerservice/StagefrightRecorder.h
@@ -53,7 +53,6 @@
     virtual status_t setVideoFrameRate(int frames_per_second);
     virtual status_t setCamera(const sp<ICamera>& camera, const sp<ICameraRecordingProxy>& proxy);
     virtual status_t setPreviewSurface(const sp<IGraphicBufferProducer>& surface);
-    virtual status_t setOutputFile(const char *path);
     virtual status_t setOutputFile(int fd, int64_t offset, int64_t length);
     virtual status_t setParameters(const String8& params);
     virtual status_t setListener(const sp<IMediaRecorderClient>& listener);
@@ -127,6 +126,8 @@
     sp<IGraphicBufferProducer> mGraphicBufferProducer;
     sp<ALooper> mLooper;
 
+    static const int kMaxHighSpeedFps = 1000;
+
     status_t prepareInternal();
     status_t setupMPEG4orWEBMRecording();
     void setupMPEG4orWEBMMetaData(sp<MetaData> *meta);
diff --git a/media/libmediaplayerservice/nuplayer/Android.mk b/media/libmediaplayerservice/nuplayer/Android.mk
index 6609874..e2c72ed 100644
--- a/media/libmediaplayerservice/nuplayer/Android.mk
+++ b/media/libmediaplayerservice/nuplayer/Android.mk
@@ -4,6 +4,7 @@
 LOCAL_SRC_FILES:=                       \
         GenericSource.cpp               \
         HTTPLiveSource.cpp              \
+        MediaClock.cpp                  \
         NuPlayer.cpp                    \
         NuPlayerCCDecoder.cpp           \
         NuPlayerDecoder.cpp             \
diff --git a/media/libmediaplayerservice/nuplayer/MediaClock.cpp b/media/libmediaplayerservice/nuplayer/MediaClock.cpp
new file mode 100644
index 0000000..7bfff13
--- /dev/null
+++ b/media/libmediaplayerservice/nuplayer/MediaClock.cpp
@@ -0,0 +1,135 @@
+/*
+ * Copyright (C) 2015 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 "MediaClock"
+#include <utils/Log.h>
+
+#include "MediaClock.h"
+
+#include <media/stagefright/foundation/ALooper.h>
+
+namespace android {
+
+// Maximum time change between two updates.
+static const int64_t kMaxAnchorFluctuationUs = 1000ll;
+
+MediaClock::MediaClock()
+    : mAnchorTimeMediaUs(-1),
+      mAnchorTimeRealUs(-1),
+      mMaxTimeMediaUs(INT64_MAX),
+      mStartingTimeMediaUs(-1),
+      mPaused(false) {
+}
+
+MediaClock::~MediaClock() {
+}
+
+void MediaClock::setStartingTimeMedia(int64_t startingTimeMediaUs) {
+    Mutex::Autolock autoLock(mLock);
+    mStartingTimeMediaUs = startingTimeMediaUs;
+}
+
+void MediaClock::clearAnchor() {
+    Mutex::Autolock autoLock(mLock);
+    mAnchorTimeMediaUs = -1;
+    mAnchorTimeRealUs = -1;
+}
+
+void MediaClock::updateAnchor(
+        int64_t anchorTimeMediaUs,
+        int64_t anchorTimeRealUs,
+        int64_t maxTimeMediaUs) {
+    if (anchorTimeMediaUs < 0 || anchorTimeRealUs < 0) {
+        ALOGW("reject anchor time since it is negative.");
+        return;
+    }
+
+    int64_t nowUs = ALooper::GetNowUs();
+    int64_t nowMediaUs = anchorTimeMediaUs + nowUs - anchorTimeRealUs;
+    if (nowMediaUs < 0) {
+        ALOGW("reject anchor time since it leads to negative media time.");
+        return;
+    }
+
+    Mutex::Autolock autoLock(mLock);
+    mAnchorTimeRealUs = nowUs;
+    mAnchorTimeMediaUs = nowMediaUs;
+    mMaxTimeMediaUs = maxTimeMediaUs;
+}
+
+void MediaClock::updateMaxTimeMedia(int64_t maxTimeMediaUs) {
+    Mutex::Autolock autoLock(mLock);
+    mMaxTimeMediaUs = maxTimeMediaUs;
+}
+
+void MediaClock::pause() {
+    Mutex::Autolock autoLock(mLock);
+    if (mPaused) {
+        return;
+    }
+
+    mPaused = true;
+    if (mAnchorTimeRealUs == -1) {
+        return;
+    }
+
+    int64_t nowUs = ALooper::GetNowUs();
+    mAnchorTimeMediaUs += nowUs - mAnchorTimeRealUs;
+    if (mAnchorTimeMediaUs < 0) {
+        ALOGW("anchor time should not be negative, set to 0.");
+        mAnchorTimeMediaUs = 0;
+    }
+    mAnchorTimeRealUs = nowUs;
+}
+
+void MediaClock::resume() {
+    Mutex::Autolock autoLock(mLock);
+    if (!mPaused) {
+        return;
+    }
+
+    mPaused = false;
+    if (mAnchorTimeRealUs == -1) {
+        return;
+    }
+
+    mAnchorTimeRealUs = ALooper::GetNowUs();
+}
+
+int64_t MediaClock::getTimeMedia(int64_t realUs, bool allowPastMaxTime) {
+    Mutex::Autolock autoLock(mLock);
+    if (mAnchorTimeRealUs == -1) {
+        return -1ll;
+    }
+
+    if (mPaused) {
+        realUs = mAnchorTimeRealUs;
+    }
+    int64_t currentMediaUs = mAnchorTimeMediaUs + realUs - mAnchorTimeRealUs;
+    if (currentMediaUs > mMaxTimeMediaUs && !allowPastMaxTime) {
+        currentMediaUs = mMaxTimeMediaUs;
+    }
+    if (currentMediaUs < mStartingTimeMediaUs) {
+        currentMediaUs = mStartingTimeMediaUs;
+    }
+    if (currentMediaUs < 0) {
+        currentMediaUs = 0;
+    }
+    return currentMediaUs;
+}
+
+}  // namespace android
diff --git a/media/libmediaplayerservice/nuplayer/MediaClock.h b/media/libmediaplayerservice/nuplayer/MediaClock.h
new file mode 100644
index 0000000..d005993
--- /dev/null
+++ b/media/libmediaplayerservice/nuplayer/MediaClock.h
@@ -0,0 +1,68 @@
+/*
+ * Copyright (C) 2015 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 MEDIA_CLOCK_H_
+
+#define MEDIA_CLOCK_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/Mutex.h>
+#include <utils/RefBase.h>
+
+namespace android {
+
+struct AMessage;
+
+struct MediaClock : public RefBase {
+    MediaClock();
+
+    void setStartingTimeMedia(int64_t startingTimeMediaUs);
+
+    void clearAnchor();
+    // It's highly recommended to use timestamp of just rendered frame as
+    // anchor time, especially in paused state. Such restriction will be
+    // required when dynamic playback rate is supported in the future.
+    void updateAnchor(
+        int64_t anchorTimeMediaUs,
+        int64_t anchorTimeRealUs,
+        int64_t maxTimeMediaUs = INT64_MAX);
+
+    void updateMaxTimeMedia(int64_t maxTimeMediaUs);
+
+    void pause();
+    void resume();
+
+    int64_t getTimeMedia(int64_t realUs, bool allowPastMaxTime = false);
+
+protected:
+    virtual ~MediaClock();
+
+private:
+    Mutex mLock;
+
+    int64_t mAnchorTimeMediaUs;
+    int64_t mAnchorTimeRealUs;
+    int64_t mMaxTimeMediaUs;
+    int64_t mStartingTimeMediaUs;
+
+    bool mPaused;
+
+    DISALLOW_EVIL_CONSTRUCTORS(MediaClock);
+};
+
+}  // namespace android
+
+#endif  // MEDIA_CLOCK_H_
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp
index 9229704..cf3e8ad 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp
@@ -19,6 +19,7 @@
 #include <utils/Log.h>
 #include <inttypes.h>
 
+#include "avc_utils.h"
 #include "NuPlayerCCDecoder.h"
 
 #include <media/stagefright/foundation/ABitReader.h>
@@ -185,17 +186,38 @@
 
 // returns true if a new CC track is found
 bool NuPlayer::CCDecoder::extractFromSEI(const sp<ABuffer> &accessUnit) {
-    int64_t timeUs;
-    CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
-
     sp<ABuffer> sei;
     if (!accessUnit->meta()->findBuffer("sei", &sei) || sei == NULL) {
         return false;
     }
 
+    int64_t timeUs;
+    CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
+
     bool trackAdded = false;
 
-    NALBitReader br(sei->data() + 1, sei->size() - 1);
+    const NALPosition *nal = (NALPosition *) sei->data();
+
+    for (size_t i = 0; i < sei->size() / sizeof(NALPosition); ++i, ++nal) {
+        trackAdded |= parseSEINalUnit(
+                timeUs, accessUnit->data() + nal->nalOffset, nal->nalSize);
+    }
+
+    return trackAdded;
+}
+
+// returns true if a new CC track is found
+bool NuPlayer::CCDecoder::parseSEINalUnit(
+        int64_t timeUs, const uint8_t *nalStart, size_t nalSize) {
+    unsigned nalType = nalStart[0] & 0x1f;
+
+    // the buffer should only have SEI in it
+    if (nalType != 6) {
+        return false;
+    }
+
+    bool trackAdded = false;
+    NALBitReader br(nalStart + 1, nalSize - 1);
     // sei_message()
     while (br.atLeastNumBitsLeft(16)) { // at least 16-bit for sei_message()
         uint32_t payload_type = 0;
@@ -214,20 +236,25 @@
 
         // sei_payload()
         if (payload_type == 4) {
-            // user_data_registered_itu_t_t35()
+            bool isCC = false;
+            if (payload_size > 1 + 2 + 4 + 1) {
+                // user_data_registered_itu_t_t35()
 
-            // ATSC A/72: 6.4.2
-            uint8_t itu_t_t35_country_code = br.getBits(8);
-            uint16_t itu_t_t35_provider_code = br.getBits(16);
-            uint32_t user_identifier = br.getBits(32);
-            uint8_t user_data_type_code = br.getBits(8);
+                // ATSC A/72: 6.4.2
+                uint8_t itu_t_t35_country_code = br.getBits(8);
+                uint16_t itu_t_t35_provider_code = br.getBits(16);
+                uint32_t user_identifier = br.getBits(32);
+                uint8_t user_data_type_code = br.getBits(8);
 
-            payload_size -= 1 + 2 + 4 + 1;
+                payload_size -= 1 + 2 + 4 + 1;
 
-            if (itu_t_t35_country_code == 0xB5
-                    && itu_t_t35_provider_code == 0x0031
-                    && user_identifier == 'GA94'
-                    && user_data_type_code == 0x3) {
+                isCC = itu_t_t35_country_code == 0xB5
+                        && itu_t_t35_provider_code == 0x0031
+                        && user_identifier == 'GA94'
+                        && user_data_type_code == 0x3;
+            }
+
+            if (isCC && payload_size > 2) {
                 // MPEG_cc_data()
                 // ATSC A/53 Part 4: 6.2.3.1
                 br.skipBits(1); //process_em_data_flag
@@ -243,7 +270,7 @@
                     sp<ABuffer> ccBuf = new ABuffer(cc_count * sizeof(CCData));
                     ccBuf->setRange(0, 0);
 
-                    for (size_t i = 0; i < cc_count; i++) {
+                    for (size_t i = 0; i < cc_count && payload_size >= 3; i++) {
                         uint8_t marker = br.getBits(5);
                         CHECK_EQ(marker, 0x1f);
 
@@ -253,6 +280,8 @@
                         uint8_t cc_data_1 = br.getBits(8) & 0x7f;
                         uint8_t cc_data_2 = br.getBits(8) & 0x7f;
 
+                        payload_size -= 3;
+
                         if (cc_valid
                                 && (cc_type == 0 || cc_type == 1)) {
                             CCData cc(cc_type, cc_data_1, cc_data_2);
@@ -269,7 +298,6 @@
                             }
                         }
                     }
-                    payload_size -= cc_count * 3;
 
                     mCCMap.add(timeUs, ccBuf);
                     break;
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.h b/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.h
index 5e06f4e..77fb0fe 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.h
@@ -49,6 +49,7 @@
     bool isTrackValid(size_t index) const;
     int32_t getTrackIndex(size_t channel) const;
     bool extractFromSEI(const sp<ABuffer> &accessUnit);
+    bool parseSEINalUnit(int64_t timeUs, const uint8_t *nalStart, size_t nalSize);
     sp<ABuffer> filterCCBuf(const sp<ABuffer> &ccBuf, size_t index);
 
     DISALLOW_EVIL_CONSTRUCTORS(CCDecoder);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
index bc79fdb..abfa4d3 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
@@ -351,6 +351,14 @@
         case STATE_PREPARED:
         case STATE_STOPPED_AND_PREPARED:
         {
+            int curpos = 0;
+            if (mPositionUs > 0) {
+                curpos = (mPositionUs + 500ll) / 1000;
+            }
+            if (curpos == msec) {
+                // nothing to do, and doing something anyway could result in deadlock (b/15323063)
+                break;
+            }
             mStartupSeekTimeUs = seekTimeUs;
             // pretend that the seek completed. It will actually happen when starting playback.
             // TODO: actually perform the seek here, so the player is ready to go at the new
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index 25225a8..7f8680d 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -20,6 +20,8 @@
 
 #include "NuPlayerRenderer.h"
 
+#include "MediaClock.h"
+
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AMessage.h>
@@ -63,22 +65,18 @@
       mDrainVideoQueuePending(false),
       mAudioQueueGeneration(0),
       mVideoQueueGeneration(0),
+      mAudioDrainGeneration(0),
+      mVideoDrainGeneration(0),
       mAudioFirstAnchorTimeMediaUs(-1),
       mAnchorTimeMediaUs(-1),
-      mAnchorTimeRealUs(-1),
       mAnchorNumFramesWritten(-1),
-      mAnchorMaxMediaUs(-1),
       mVideoLateByUs(0ll),
       mHasAudio(false),
       mHasVideo(false),
-      mPauseStartedTimeRealUs(-1),
-      mFlushingAudio(false),
-      mFlushingVideo(false),
       mNotifyCompleteAudio(false),
       mNotifyCompleteVideo(false),
       mSyncQueues(false),
       mPaused(false),
-      mPausePositionMediaTimeUs(-1),
       mVideoSampleReceived(false),
       mVideoRenderingStarted(false),
       mVideoRenderingStartGeneration(0),
@@ -90,7 +88,7 @@
       mTotalBuffersQueued(0),
       mLastAudioBufferDrained(0),
       mWakeLock(new AWakeLock()) {
-
+    mMediaClock = new MediaClock;
 }
 
 NuPlayer::Renderer::~Renderer() {
@@ -106,6 +104,7 @@
         const sp<ABuffer> &buffer,
         const sp<AMessage> &notifyConsumed) {
     sp<AMessage> msg = new AMessage(kWhatQueueBuffer, id());
+    msg->setInt32("queueGeneration", getQueueGeneration(audio));
     msg->setInt32("audio", static_cast<int32_t>(audio));
     msg->setBuffer("buffer", buffer);
     msg->setMessage("notifyConsumed", notifyConsumed);
@@ -116,6 +115,7 @@
     CHECK_NE(finalResult, (status_t)OK);
 
     sp<AMessage> msg = new AMessage(kWhatQueueEOS, id());
+    msg->setInt32("queueGeneration", getQueueGeneration(audio));
     msg->setInt32("audio", static_cast<int32_t>(audio));
     msg->setInt32("finalResult", finalResult);
     msg->post();
@@ -123,20 +123,21 @@
 
 void NuPlayer::Renderer::flush(bool audio, bool notifyComplete) {
     {
-        Mutex::Autolock autoLock(mFlushLock);
+        Mutex::Autolock autoLock(mLock);
         if (audio) {
             mNotifyCompleteAudio |= notifyComplete;
-            if (mFlushingAudio) {
-                return;
-            }
-            mFlushingAudio = true;
+            ++mAudioQueueGeneration;
+            ++mAudioDrainGeneration;
         } else {
             mNotifyCompleteVideo |= notifyComplete;
-            if (mFlushingVideo) {
-                return;
-            }
-            mFlushingVideo = true;
+            ++mVideoQueueGeneration;
+            ++mVideoDrainGeneration;
         }
+
+        clearAnchorTime_l();
+        clearAudioFirstAnchorTime_l();
+        mVideoLateByUs = 0;
+        mSyncQueues = false;
     }
 
     sp<AMessage> msg = new AMessage(kWhatFlush, id());
@@ -145,17 +146,6 @@
 }
 
 void NuPlayer::Renderer::signalTimeDiscontinuity() {
-    Mutex::Autolock autoLock(mLock);
-    // CHECK(mAudioQueue.empty());
-    // CHECK(mVideoQueue.empty());
-    setAudioFirstAnchorTime(-1);
-    setAnchorTime(-1, -1);
-    setVideoLateByUs(0);
-    mSyncQueues = false;
-}
-
-void NuPlayer::Renderer::signalAudioSinkChanged() {
-    (new AMessage(kWhatAudioSinkChanged, id()))->post();
 }
 
 void NuPlayer::Renderer::signalDisableOffloadAudio() {
@@ -180,127 +170,44 @@
     msg->post();
 }
 
-// Called on any threads, except renderer's thread.
-status_t NuPlayer::Renderer::getCurrentPosition(int64_t *mediaUs) {
-    {
-        Mutex::Autolock autoLock(mLock);
-        int64_t currentPositionUs;
-        if (getCurrentPositionIfPaused_l(&currentPositionUs)) {
-            *mediaUs = currentPositionUs;
-            return OK;
-        }
-    }
-    return getCurrentPositionFromAnchor(mediaUs, ALooper::GetNowUs());
-}
-
-// Called on only renderer's thread.
-status_t NuPlayer::Renderer::getCurrentPositionOnLooper(int64_t *mediaUs) {
-    return getCurrentPositionOnLooper(mediaUs, ALooper::GetNowUs());
-}
-
-// Called on only renderer's thread.
-// Since mPaused and mPausePositionMediaTimeUs are changed only on renderer's
-// thread, no need to acquire mLock.
-status_t NuPlayer::Renderer::getCurrentPositionOnLooper(
-        int64_t *mediaUs, int64_t nowUs, bool allowPastQueuedVideo) {
-    int64_t currentPositionUs;
-    if (getCurrentPositionIfPaused_l(&currentPositionUs)) {
-        *mediaUs = currentPositionUs;
-        return OK;
-    }
-    return getCurrentPositionFromAnchor(mediaUs, nowUs, allowPastQueuedVideo);
-}
-
-// Called either with mLock acquired or on renderer's thread.
-bool NuPlayer::Renderer::getCurrentPositionIfPaused_l(int64_t *mediaUs) {
-    if (!mPaused || mPausePositionMediaTimeUs < 0ll) {
-        return false;
-    }
-    *mediaUs = mPausePositionMediaTimeUs;
-    return true;
-}
-
 // Called on any threads.
-status_t NuPlayer::Renderer::getCurrentPositionFromAnchor(
-        int64_t *mediaUs, int64_t nowUs, bool allowPastQueuedVideo) {
-    Mutex::Autolock autoLock(mTimeLock);
-    if (!mHasAudio && !mHasVideo) {
+status_t NuPlayer::Renderer::getCurrentPosition(int64_t *mediaUs) {
+    int64_t currentTimeUs = mMediaClock->getTimeMedia(ALooper::GetNowUs());
+    if (currentTimeUs == -1) {
         return NO_INIT;
     }
-
-    if (mAnchorTimeMediaUs < 0) {
-        return NO_INIT;
-    }
-
-    int64_t positionUs = (nowUs - mAnchorTimeRealUs) + mAnchorTimeMediaUs;
-
-    if (mPauseStartedTimeRealUs != -1) {
-        positionUs -= (nowUs - mPauseStartedTimeRealUs);
-    }
-
-    // limit position to the last queued media time (for video only stream
-    // position will be discrete as we don't know how long each frame lasts)
-    if (mAnchorMaxMediaUs >= 0 && !allowPastQueuedVideo) {
-        if (positionUs > mAnchorMaxMediaUs) {
-            positionUs = mAnchorMaxMediaUs;
-        }
-    }
-
-    if (positionUs < mAudioFirstAnchorTimeMediaUs) {
-        positionUs = mAudioFirstAnchorTimeMediaUs;
-    }
-
-    *mediaUs = (positionUs <= 0) ? 0 : positionUs;
+    *mediaUs = currentTimeUs;
     return OK;
 }
 
-void NuPlayer::Renderer::setHasMedia(bool audio) {
-    Mutex::Autolock autoLock(mTimeLock);
-    if (audio) {
-        mHasAudio = true;
-    } else {
-        mHasVideo = true;
-    }
+void NuPlayer::Renderer::clearAudioFirstAnchorTime_l() {
+    mAudioFirstAnchorTimeMediaUs = -1;
+    mMediaClock->setStartingTimeMedia(-1);
 }
 
-void NuPlayer::Renderer::setAudioFirstAnchorTime(int64_t mediaUs) {
-    Mutex::Autolock autoLock(mTimeLock);
-    mAudioFirstAnchorTimeMediaUs = mediaUs;
-}
-
-void NuPlayer::Renderer::setAudioFirstAnchorTimeIfNeeded(int64_t mediaUs) {
-    Mutex::Autolock autoLock(mTimeLock);
+void NuPlayer::Renderer::setAudioFirstAnchorTimeIfNeeded_l(int64_t mediaUs) {
     if (mAudioFirstAnchorTimeMediaUs == -1) {
         mAudioFirstAnchorTimeMediaUs = mediaUs;
+        mMediaClock->setStartingTimeMedia(mediaUs);
     }
 }
 
-void NuPlayer::Renderer::setAnchorTime(
-        int64_t mediaUs, int64_t realUs, int64_t numFramesWritten, bool resume) {
-    Mutex::Autolock autoLock(mTimeLock);
-    mAnchorTimeMediaUs = mediaUs;
-    mAnchorTimeRealUs = realUs;
-    mAnchorNumFramesWritten = numFramesWritten;
-    if (resume) {
-        mPauseStartedTimeRealUs = -1;
-    }
+void NuPlayer::Renderer::clearAnchorTime_l() {
+    mMediaClock->clearAnchor();
+    mAnchorTimeMediaUs = -1;
+    mAnchorNumFramesWritten = -1;
 }
 
 void NuPlayer::Renderer::setVideoLateByUs(int64_t lateUs) {
-    Mutex::Autolock autoLock(mTimeLock);
+    Mutex::Autolock autoLock(mLock);
     mVideoLateByUs = lateUs;
 }
 
 int64_t NuPlayer::Renderer::getVideoLateByUs() {
-    Mutex::Autolock autoLock(mTimeLock);
+    Mutex::Autolock autoLock(mLock);
     return mVideoLateByUs;
 }
 
-void NuPlayer::Renderer::setPauseStartedTimeRealUs(int64_t realUs) {
-    Mutex::Autolock autoLock(mTimeLock);
-    mPauseStartedTimeRealUs = realUs;
-}
-
 status_t NuPlayer::Renderer::openAudioSink(
         const sp<AMessage> &format,
         bool offloadOnly,
@@ -384,8 +291,8 @@
         case kWhatDrainAudioQueue:
         {
             int32_t generation;
-            CHECK(msg->findInt32("generation", &generation));
-            if (generation != mAudioQueueGeneration) {
+            CHECK(msg->findInt32("drainGeneration", &generation));
+            if (generation != getDrainGeneration(true /* audio */)) {
                 break;
             }
 
@@ -407,9 +314,7 @@
 
                 // Let's give it more data after about half that time
                 // has elapsed.
-                // kWhatDrainAudioQueue is used for non-offloading mode,
-                // and mLock is used only for offloading mode. Therefore,
-                // no need to acquire mLock here.
+                Mutex::Autolock autoLock(mLock);
                 postDrainAudioQueue_l(delayUs / 2);
             }
             break;
@@ -418,8 +323,8 @@
         case kWhatDrainVideoQueue:
         {
             int32_t generation;
-            CHECK(msg->findInt32("generation", &generation));
-            if (generation != mVideoQueueGeneration) {
+            CHECK(msg->findInt32("drainGeneration", &generation));
+            if (generation != getDrainGeneration(false /* audio */)) {
                 break;
             }
 
@@ -427,22 +332,20 @@
 
             onDrainVideoQueue();
 
-            Mutex::Autolock autoLock(mLock);
-            postDrainVideoQueue_l();
+            postDrainVideoQueue();
             break;
         }
 
         case kWhatPostDrainVideoQueue:
         {
             int32_t generation;
-            CHECK(msg->findInt32("generation", &generation));
-            if (generation != mVideoQueueGeneration) {
+            CHECK(msg->findInt32("drainGeneration", &generation));
+            if (generation != getDrainGeneration(false /* audio */)) {
                 break;
             }
 
             mDrainVideoQueuePending = false;
-            Mutex::Autolock autoLock(mLock);
-            postDrainVideoQueue_l();
+            postDrainVideoQueue();
             break;
         }
 
@@ -464,12 +367,6 @@
             break;
         }
 
-        case kWhatAudioSinkChanged:
-        {
-            onAudioSinkChanged();
-            break;
-        }
-
         case kWhatDisableOffloadAudio:
         {
             onDisableOffloadAudio();
@@ -511,7 +408,7 @@
         case kWhatAudioOffloadPauseTimeout:
         {
             int32_t generation;
-            CHECK(msg->findInt32("generation", &generation));
+            CHECK(msg->findInt32("drainGeneration", &generation));
             if (generation != mAudioOffloadPauseTimeoutGeneration) {
                 break;
             }
@@ -539,18 +436,18 @@
 
     mDrainAudioQueuePending = true;
     sp<AMessage> msg = new AMessage(kWhatDrainAudioQueue, id());
-    msg->setInt32("generation", mAudioQueueGeneration);
+    msg->setInt32("drainGeneration", mAudioDrainGeneration);
     msg->post(delayUs);
 }
 
-void NuPlayer::Renderer::prepareForMediaRenderingStart() {
-    mAudioRenderingStartGeneration = mAudioQueueGeneration;
-    mVideoRenderingStartGeneration = mVideoQueueGeneration;
+void NuPlayer::Renderer::prepareForMediaRenderingStart_l() {
+    mAudioRenderingStartGeneration = mAudioDrainGeneration;
+    mVideoRenderingStartGeneration = mVideoDrainGeneration;
 }
 
-void NuPlayer::Renderer::notifyIfMediaRenderingStarted() {
-    if (mVideoRenderingStartGeneration == mVideoQueueGeneration &&
-        mAudioRenderingStartGeneration == mAudioQueueGeneration) {
+void NuPlayer::Renderer::notifyIfMediaRenderingStarted_l() {
+    if (mVideoRenderingStartGeneration == mVideoDrainGeneration &&
+        mAudioRenderingStartGeneration == mAudioDrainGeneration) {
         mVideoRenderingStartGeneration = -1;
         mAudioRenderingStartGeneration = -1;
 
@@ -618,7 +515,7 @@
             int64_t mediaTimeUs;
             CHECK(entry->mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));
             ALOGV("rendering audio at media time %.2f secs", mediaTimeUs / 1E6);
-            setAudioFirstAnchorTimeIfNeeded(mediaTimeUs);
+            setAudioFirstAnchorTimeIfNeeded_l(mediaTimeUs);
         }
 
         size_t copy = entry->mBuffer->size() - entry->mOffset;
@@ -638,17 +535,18 @@
             entry = NULL;
         }
         sizeCopied += copy;
-        notifyIfMediaRenderingStarted();
+
+        notifyIfMediaRenderingStarted_l();
     }
 
     if (mAudioFirstAnchorTimeMediaUs >= 0) {
         int64_t nowUs = ALooper::GetNowUs();
-        setAnchorTime(mAudioFirstAnchorTimeMediaUs, nowUs - getPlayedOutAudioDurationUs(nowUs));
+        // we don't know how much data we are queueing for offloaded tracks.
+        mMediaClock->updateAnchor(mAudioFirstAnchorTimeMediaUs,
+                                  nowUs - getPlayedOutAudioDurationUs(nowUs),
+                                  INT64_MAX);
     }
 
-    // we don't know how much data we are queueing for offloaded tracks
-    mAnchorMaxMediaUs = -1;
-
     if (hasEOS) {
         (new AMessage(kWhatStopAudioSink, id()))->post();
     }
@@ -733,7 +631,10 @@
         size_t copiedFrames = written / mAudioSink->frameSize();
         mNumFramesWritten += copiedFrames;
 
-        notifyIfMediaRenderingStarted();
+        {
+            Mutex::Autolock autoLock(mLock);
+            notifyIfMediaRenderingStarted_l();
+        }
 
         if (written != (ssize_t)copy) {
             // A short count was received from AudioSink::write()
@@ -756,10 +657,15 @@
             break;
         }
     }
-    mAnchorMaxMediaUs =
-        mAnchorTimeMediaUs +
-                (int64_t)(max((long long)mNumFramesWritten - mAnchorNumFramesWritten, 0LL)
-                        * 1000LL * mAudioSink->msecsPerFrame());
+    int64_t maxTimeMedia;
+    {
+        Mutex::Autolock autoLock(mLock);
+        maxTimeMedia =
+            mAnchorTimeMediaUs +
+                    (int64_t)(max((long long)mNumFramesWritten - mAnchorNumFramesWritten, 0LL)
+                            * 1000LL * mAudioSink->msecsPerFrame());
+    }
+    mMediaClock->updateMaxTimeMedia(maxTimeMedia);
 
     return !mAudioQueue.empty();
 }
@@ -771,31 +677,35 @@
 }
 
 int64_t NuPlayer::Renderer::getRealTimeUs(int64_t mediaTimeUs, int64_t nowUs) {
-    int64_t currentPositionUs;
-    if (mPaused || getCurrentPositionOnLooper(
-            &currentPositionUs, nowUs, true /* allowPastQueuedVideo */) != OK) {
-        // If failed to get current position, e.g. due to audio clock is not ready, then just
-        // play out video immediately without delay.
+    int64_t currentPositionUs =
+            mMediaClock->getTimeMedia(nowUs, true /* allowPastMaxTime */);
+    if (currentPositionUs == -1) {
+        // If failed to get current position, e.g. due to audio clock is
+        // not ready, then just play out video immediately without delay.
         return nowUs;
     }
     return (mediaTimeUs - currentPositionUs) + nowUs;
 }
 
 void NuPlayer::Renderer::onNewAudioMediaTime(int64_t mediaTimeUs) {
+    Mutex::Autolock autoLock(mLock);
     // TRICKY: vorbis decoder generates multiple frames with the same
     // timestamp, so only update on the first frame with a given timestamp
     if (mediaTimeUs == mAnchorTimeMediaUs) {
         return;
     }
-    setAudioFirstAnchorTimeIfNeeded(mediaTimeUs);
+    setAudioFirstAnchorTimeIfNeeded_l(mediaTimeUs);
     int64_t nowUs = ALooper::GetNowUs();
-    setAnchorTime(
-            mediaTimeUs, nowUs + getPendingAudioPlayoutDurationUs(nowUs), mNumFramesWritten);
+    mMediaClock->updateAnchor(mediaTimeUs,
+                              nowUs + getPendingAudioPlayoutDurationUs(nowUs),
+                              mediaTimeUs);
+    mAnchorTimeMediaUs = mediaTimeUs;
 }
 
-void NuPlayer::Renderer::postDrainVideoQueue_l() {
+// Called without mLock acquired.
+void NuPlayer::Renderer::postDrainVideoQueue() {
     if (mDrainVideoQueuePending
-            || mSyncQueues
+            || getSyncQueues()
             || (mPaused && mVideoSampleReceived)) {
         return;
     }
@@ -807,7 +717,7 @@
     QueueEntry &entry = *mVideoQueue.begin();
 
     sp<AMessage> msg = new AMessage(kWhatDrainVideoQueue, id());
-    msg->setInt32("generation", mVideoQueueGeneration);
+    msg->setInt32("drainGeneration", getDrainGeneration(false /* audio */));
 
     if (entry.mBuffer == NULL) {
         // EOS doesn't carry a timestamp.
@@ -827,16 +737,19 @@
         int64_t mediaTimeUs;
         CHECK(entry.mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));
 
-        if (mAnchorTimeMediaUs < 0) {
-            setAnchorTime(mediaTimeUs, nowUs);
-            mPausePositionMediaTimeUs = mediaTimeUs;
-            mAnchorMaxMediaUs = mediaTimeUs;
-            realTimeUs = nowUs;
-        } else {
-            realTimeUs = getRealTimeUs(mediaTimeUs, nowUs);
+        {
+            Mutex::Autolock autoLock(mLock);
+            if (mAnchorTimeMediaUs < 0) {
+                mMediaClock->updateAnchor(mediaTimeUs, nowUs, mediaTimeUs);
+                mAnchorTimeMediaUs = mediaTimeUs;
+                realTimeUs = nowUs;
+            } else {
+                realTimeUs = getRealTimeUs(mediaTimeUs, nowUs);
+            }
         }
         if (!mHasAudio) {
-            mAnchorMaxMediaUs = mediaTimeUs + 100000; // smooth out videos >= 10fps
+            // smooth out videos >= 10fps
+            mMediaClock->updateMaxTimeMedia(mediaTimeUs + 100000);
         }
 
         // Heuristics to handle situation when media time changed without a
@@ -917,14 +830,15 @@
         } else {
             ALOGV("rendering video at media time %.2f secs",
                     (mFlags & FLAG_REAL_TIME ? realTimeUs :
-                    (realTimeUs + mAnchorTimeMediaUs - mAnchorTimeRealUs)) / 1E6);
+                    mMediaClock->getTimeMedia(realTimeUs)) / 1E6);
         }
     } else {
         setVideoLateByUs(0);
         if (!mVideoSampleReceived && !mHasAudio) {
             // This will ensure that the first frame after a flush won't be used as anchor
             // when renderer is in paused state, because resume can happen any time after seek.
-            setAnchorTime(-1, -1);
+            Mutex::Autolock autoLock(mLock);
+            clearAnchorTime_l();
         }
     }
 
@@ -941,7 +855,8 @@
             mVideoRenderingStarted = true;
             notifyVideoRenderingStart();
         }
-        notifyIfMediaRenderingStarted();
+        Mutex::Autolock autoLock(mLock);
+        notifyIfMediaRenderingStarted_l();
     }
 }
 
@@ -967,7 +882,15 @@
     int32_t audio;
     CHECK(msg->findInt32("audio", &audio));
 
-    setHasMedia(audio);
+    if (dropBufferIfStale(audio, msg)) {
+        return;
+    }
+
+    if (audio) {
+        mHasAudio = true;
+    } else {
+        mHasVideo = true;
+    }
 
     if (mHasVideo) {
         if (mVideoScheduler == NULL) {
@@ -976,10 +899,6 @@
         }
     }
 
-    if (dropBufferWhileFlushing(audio, msg)) {
-        return;
-    }
-
     sp<ABuffer> buffer;
     CHECK(msg->findBuffer("buffer", &buffer));
 
@@ -993,15 +912,16 @@
     entry.mFinalResult = OK;
     entry.mBufferOrdinal = ++mTotalBuffersQueued;
 
-    Mutex::Autolock autoLock(mLock);
     if (audio) {
+        Mutex::Autolock autoLock(mLock);
         mAudioQueue.push_back(entry);
         postDrainAudioQueue_l();
     } else {
         mVideoQueue.push_back(entry);
-        postDrainVideoQueue_l();
+        postDrainVideoQueue();
     }
 
+    Mutex::Autolock autoLock(mLock);
     if (!mSyncQueues || mAudioQueue.empty() || mVideoQueue.empty()) {
         return;
     }
@@ -1050,7 +970,9 @@
     }
 
     if (!mVideoQueue.empty()) {
-        postDrainVideoQueue_l();
+        mLock.unlock();
+        postDrainVideoQueue();
+        mLock.lock();
     }
 }
 
@@ -1058,7 +980,7 @@
     int32_t audio;
     CHECK(msg->findInt32("audio", &audio));
 
-    if (dropBufferWhileFlushing(audio, msg)) {
+    if (dropBufferIfStale(audio, msg)) {
         return;
     }
 
@@ -1069,19 +991,20 @@
     entry.mOffset = 0;
     entry.mFinalResult = finalResult;
 
-    Mutex::Autolock autoLock(mLock);
     if (audio) {
+        Mutex::Autolock autoLock(mLock);
         if (mAudioQueue.empty() && mSyncQueues) {
             syncQueuesDone_l();
         }
         mAudioQueue.push_back(entry);
         postDrainAudioQueue_l();
     } else {
-        if (mVideoQueue.empty() && mSyncQueues) {
+        if (mVideoQueue.empty() && getSyncQueues()) {
+            Mutex::Autolock autoLock(mLock);
             syncQueuesDone_l();
         }
         mVideoQueue.push_back(entry);
-        postDrainVideoQueue_l();
+        postDrainVideoQueue();
     }
 }
 
@@ -1090,31 +1013,25 @@
     CHECK(msg->findInt32("audio", &audio));
 
     {
-        Mutex::Autolock autoLock(mFlushLock);
+        Mutex::Autolock autoLock(mLock);
         if (audio) {
-            mFlushingAudio = false;
             notifyComplete = mNotifyCompleteAudio;
             mNotifyCompleteAudio = false;
         } else {
-            mFlushingVideo = false;
             notifyComplete = mNotifyCompleteVideo;
             mNotifyCompleteVideo = false;
         }
-    }
 
-    // If we're currently syncing the queues, i.e. dropping audio while
-    // aligning the first audio/video buffer times and only one of the
-    // two queues has data, we may starve that queue by not requesting
-    // more buffers from the decoder. If the other source then encounters
-    // a discontinuity that leads to flushing, we'll never find the
-    // corresponding discontinuity on the other queue.
-    // Therefore we'll stop syncing the queues if at least one of them
-    // is flushed.
-    {
-         Mutex::Autolock autoLock(mLock);
-         syncQueuesDone_l();
-         setPauseStartedTimeRealUs(-1);
-         setAnchorTime(-1, -1);
+        // If we're currently syncing the queues, i.e. dropping audio while
+        // aligning the first audio/video buffer times and only one of the
+        // two queues has data, we may starve that queue by not requesting
+        // more buffers from the decoder. If the other source then encounters
+        // a discontinuity that leads to flushing, we'll never find the
+        // corresponding discontinuity on the other queue.
+        // Therefore we'll stop syncing the queues if at least one of them
+        // is flushed.
+        syncQueuesDone_l();
+        clearAnchorTime_l();
     }
 
     ALOGV("flushing %s", audio ? "audio" : "video");
@@ -1123,11 +1040,11 @@
             Mutex::Autolock autoLock(mLock);
             flushQueue(&mAudioQueue);
 
-            ++mAudioQueueGeneration;
-            prepareForMediaRenderingStart();
+            ++mAudioDrainGeneration;
+            prepareForMediaRenderingStart_l();
 
             if (offloadingAudio()) {
-                setAudioFirstAnchorTime(-1);
+                clearAudioFirstAnchorTime_l();
             }
         }
 
@@ -1142,13 +1059,14 @@
         flushQueue(&mVideoQueue);
 
         mDrainVideoQueuePending = false;
-        ++mVideoQueueGeneration;
 
         if (mVideoScheduler != NULL) {
             mVideoScheduler->restart();
         }
 
-        prepareForMediaRenderingStart();
+        Mutex::Autolock autoLock(mLock);
+        ++mVideoDrainGeneration;
+        prepareForMediaRenderingStart_l();
     }
 
     mVideoSampleReceived = false;
@@ -1178,20 +1096,12 @@
     notify->post();
 }
 
-bool NuPlayer::Renderer::dropBufferWhileFlushing(
+bool NuPlayer::Renderer::dropBufferIfStale(
         bool audio, const sp<AMessage> &msg) {
-    bool flushing = false;
+    int32_t queueGeneration;
+    CHECK(msg->findInt32("queueGeneration", &queueGeneration));
 
-    {
-        Mutex::Autolock autoLock(mFlushLock);
-        if (audio) {
-            flushing = mFlushingAudio;
-        } else {
-            flushing = mFlushingVideo;
-        }
-    }
-
-    if (!flushing) {
+    if (queueGeneration == getQueueGeneration(audio)) {
         return false;
     }
 
@@ -1209,7 +1119,10 @@
     }
     CHECK(!mDrainAudioQueuePending);
     mNumFramesWritten = 0;
-    mAnchorNumFramesWritten = -1;
+    {
+        Mutex::Autolock autoLock(mLock);
+        mAnchorNumFramesWritten = -1;
+    }
     uint32_t written;
     if (mAudioSink->getFramesWritten(&written) == OK) {
         mNumFramesWritten = written;
@@ -1219,13 +1132,13 @@
 void NuPlayer::Renderer::onDisableOffloadAudio() {
     Mutex::Autolock autoLock(mLock);
     mFlags &= ~FLAG_OFFLOAD_AUDIO;
-    ++mAudioQueueGeneration;
+    ++mAudioDrainGeneration;
 }
 
 void NuPlayer::Renderer::onEnableOffloadAudio() {
     Mutex::Autolock autoLock(mLock);
     mFlags |= FLAG_OFFLOAD_AUDIO;
-    ++mAudioQueueGeneration;
+    ++mAudioDrainGeneration;
 }
 
 void NuPlayer::Renderer::onPause() {
@@ -1234,25 +1147,13 @@
         return;
     }
     int64_t currentPositionUs;
-    int64_t pausePositionMediaTimeUs;
-    if (getCurrentPositionFromAnchor(
-            &currentPositionUs, ALooper::GetNowUs()) == OK) {
-        pausePositionMediaTimeUs = currentPositionUs;
-    } else {
-        // Set paused position to -1 (unavailabe) if we don't have anchor time
-        // This could happen if client does a seekTo() immediately followed by
-        // pause(). Renderer will be flushed with anchor time cleared. We don't
-        // want to leave stale value in mPausePositionMediaTimeUs.
-        pausePositionMediaTimeUs = -1;
-    }
     {
         Mutex::Autolock autoLock(mLock);
-        mPausePositionMediaTimeUs = pausePositionMediaTimeUs;
-        ++mAudioQueueGeneration;
-        ++mVideoQueueGeneration;
-        prepareForMediaRenderingStart();
+        ++mAudioDrainGeneration;
+        ++mVideoDrainGeneration;
+        prepareForMediaRenderingStart_l();
         mPaused = true;
-        setPauseStartedTimeRealUs(ALooper::GetNowUs());
+        mMediaClock->pause();
     }
 
     mDrainAudioQueuePending = false;
@@ -1277,21 +1178,18 @@
         mAudioSink->start();
     }
 
-    Mutex::Autolock autoLock(mLock);
-    mPaused = false;
-    if (mPauseStartedTimeRealUs != -1) {
-        int64_t newAnchorRealUs =
-            mAnchorTimeRealUs + ALooper::GetNowUs() - mPauseStartedTimeRealUs;
-        setAnchorTime(
-                mAnchorTimeMediaUs, newAnchorRealUs, mAnchorNumFramesWritten, true /* resume */);
-    }
+    {
+        Mutex::Autolock autoLock(mLock);
+        mPaused = false;
+        mMediaClock->resume();
 
-    if (!mAudioQueue.empty()) {
-        postDrainAudioQueue_l();
+        if (!mAudioQueue.empty()) {
+            postDrainAudioQueue_l();
+        }
     }
 
     if (!mVideoQueue.empty()) {
-        postDrainVideoQueue_l();
+        postDrainVideoQueue();
     }
 }
 
@@ -1302,6 +1200,21 @@
     mVideoScheduler->init(fps);
 }
 
+int32_t NuPlayer::Renderer::getQueueGeneration(bool audio) {
+    Mutex::Autolock autoLock(mLock);
+    return (audio ? mAudioQueueGeneration : mVideoQueueGeneration);
+}
+
+int32_t NuPlayer::Renderer::getDrainGeneration(bool audio) {
+    Mutex::Autolock autoLock(mLock);
+    return (audio ? mAudioDrainGeneration : mVideoDrainGeneration);
+}
+
+bool NuPlayer::Renderer::getSyncQueues() {
+    Mutex::Autolock autoLock(mLock);
+    return mSyncQueues;
+}
+
 // TODO: Remove unnecessary calls to getPlayedOutAudioDurationUs()
 // as it acquires locks and may query the audio driver.
 //
@@ -1373,7 +1286,7 @@
     mAudioOffloadTornDown = true;
 
     int64_t currentPositionUs;
-    if (getCurrentPositionOnLooper(&currentPositionUs) != OK) {
+    if (getCurrentPosition(&currentPositionUs) != OK) {
         currentPositionUs = 0;
     }
 
@@ -1391,7 +1304,7 @@
     if (offloadingAudio()) {
         mWakeLock->acquire();
         sp<AMessage> msg = new AMessage(kWhatAudioOffloadPauseTimeout, id());
-        msg->setInt32("generation", mAudioOffloadPauseTimeoutGeneration);
+        msg->setInt32("drainGeneration", mAudioOffloadPauseTimeoutGeneration);
         msg->post(kOffloadPauseMaxUs);
     }
 }
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h
index 003d1d0..faf3b3f 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h
@@ -24,6 +24,7 @@
 
 struct ABuffer;
 class  AWakeLock;
+struct MediaClock;
 struct VideoFrameScheduler;
 
 struct NuPlayer::Renderer : public AHandler {
@@ -61,16 +62,8 @@
 
     void setVideoFrameRate(float fps);
 
-    // Following setters and getters are protected by mTimeLock.
     status_t getCurrentPosition(int64_t *mediaUs);
-    void setHasMedia(bool audio);
-    void setAudioFirstAnchorTime(int64_t mediaUs);
-    void setAudioFirstAnchorTimeIfNeeded(int64_t mediaUs);
-    void setAnchorTime(
-            int64_t mediaUs, int64_t realUs, int64_t numFramesWritten = -1, bool resume = false);
-    void setVideoLateByUs(int64_t lateUs);
     int64_t getVideoLateByUs();
-    void setPauseStartedTimeRealUs(int64_t realUs);
 
     status_t openAudioSink(
             const sp<AMessage> &format,
@@ -108,7 +101,6 @@
         kWhatQueueBuffer         = 'queB',
         kWhatQueueEOS            = 'qEOS',
         kWhatFlush               = 'flus',
-        kWhatAudioSinkChanged    = 'auSC',
         kWhatPause               = 'paus',
         kWhatResume              = 'resm',
         kWhatOpenAudioSink       = 'opnA',
@@ -142,26 +134,17 @@
     bool mDrainVideoQueuePending;
     int32_t mAudioQueueGeneration;
     int32_t mVideoQueueGeneration;
+    int32_t mAudioDrainGeneration;
+    int32_t mVideoDrainGeneration;
 
-    Mutex mTimeLock;
-    // |mTimeLock| protects the following 7 member vars that are related to time.
-    // Note: those members are only written on Renderer thread, so reading on Renderer thread
-    // doesn't need to be protected. Otherwise accessing those members must be protected by
-    // |mTimeLock|.
-    // TODO: move those members to a seperated media clock class.
+    sp<MediaClock> mMediaClock;
     int64_t mAudioFirstAnchorTimeMediaUs;
     int64_t mAnchorTimeMediaUs;
-    int64_t mAnchorTimeRealUs;
     int64_t mAnchorNumFramesWritten;
-    int64_t mAnchorMaxMediaUs;
     int64_t mVideoLateByUs;
     bool mHasAudio;
     bool mHasVideo;
-    int64_t mPauseStartedTimeRealUs;
 
-    Mutex mFlushLock;  // protects the following 2 member vars.
-    bool mFlushingAudio;
-    bool mFlushingVideo;
     bool mNotifyCompleteAudio;
     bool mNotifyCompleteVideo;
 
@@ -169,7 +152,6 @@
 
     // modified on only renderer's thread.
     bool mPaused;
-    int64_t mPausePositionMediaTimeUs;
 
     bool mVideoSampleReceived;
     bool mVideoRenderingStarted;
@@ -211,14 +193,19 @@
     int64_t getPlayedOutAudioDurationUs(int64_t nowUs);
     void postDrainAudioQueue_l(int64_t delayUs = 0);
 
+    void clearAnchorTime_l();
+    void clearAudioFirstAnchorTime_l();
+    void setAudioFirstAnchorTimeIfNeeded_l(int64_t mediaUs);
+    void setVideoLateByUs(int64_t lateUs);
+
     void onNewAudioMediaTime(int64_t mediaTimeUs);
     int64_t getRealTimeUs(int64_t mediaTimeUs, int64_t nowUs);
 
     void onDrainVideoQueue();
-    void postDrainVideoQueue_l();
+    void postDrainVideoQueue();
 
-    void prepareForMediaRenderingStart();
-    void notifyIfMediaRenderingStarted();
+    void prepareForMediaRenderingStart_l();
+    void notifyIfMediaRenderingStarted_l();
 
     void onQueueBuffer(const sp<AMessage> &msg);
     void onQueueEOS(const sp<AMessage> &msg);
@@ -229,6 +216,9 @@
     void onPause();
     void onResume();
     void onSetVideoFrameRate(float fps);
+    int32_t getQueueGeneration(bool audio);
+    int32_t getDrainGeneration(bool audio);
+    bool getSyncQueues();
     void onAudioOffloadTearDown(AudioOffloadTearDownReason reason);
     status_t onOpenAudioSink(
             const sp<AMessage> &format,
@@ -245,7 +235,7 @@
     void notifyAudioOffloadTearDown();
 
     void flushQueue(List<QueueEntry> *queue);
-    bool dropBufferWhileFlushing(bool audio, const sp<AMessage> &msg);
+    bool dropBufferIfStale(bool audio, const sp<AMessage> &msg);
     void syncQueuesDone_l();
 
     bool offloadingAudio() const { return (mFlags & FLAG_OFFLOAD_AUDIO) != 0; }
diff --git a/media/libnbaio/Android.mk b/media/libnbaio/Android.mk
index 9707c4a..1353f28 100644
--- a/media/libnbaio/Android.mk
+++ b/media/libnbaio/Android.mk
@@ -11,7 +11,6 @@
     MonoPipeReader.cpp              \
     Pipe.cpp                        \
     PipeReader.cpp                  \
-    roundup.c                       \
     SourceAudioBufferProvider.cpp
 
 LOCAL_SRC_FILES += NBLog.cpp
@@ -27,12 +26,13 @@
 LOCAL_MODULE := libnbaio
 
 LOCAL_SHARED_LIBRARIES := \
+    libaudioutils \
     libbinder \
     libcommon_time_client \
     libcutils \
     libutils \
     liblog
 
-LOCAL_STATIC_LIBRARIES += libinstantssq
+LOCAL_C_INCLUDES := $(call include-path-for, audio-utils)
 
 include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libnbaio/MonoPipe.cpp b/media/libnbaio/MonoPipe.cpp
index 0b65861..129e9ef 100644
--- a/media/libnbaio/MonoPipe.cpp
+++ b/media/libnbaio/MonoPipe.cpp
@@ -27,7 +27,7 @@
 #include <utils/Trace.h>
 #include <media/AudioBufferProvider.h>
 #include <media/nbaio/MonoPipe.h>
-#include <media/nbaio/roundup.h>
+#include <audio_utils/roundup.h>
 
 
 namespace android {
diff --git a/media/libnbaio/MonoPipeReader.cpp b/media/libnbaio/MonoPipeReader.cpp
index de82229..e4d3ed8 100644
--- a/media/libnbaio/MonoPipeReader.cpp
+++ b/media/libnbaio/MonoPipeReader.cpp
@@ -39,7 +39,7 @@
         return NEGOTIATE;
     }
     ssize_t ret = android_atomic_acquire_load(&mPipe->mRear) - mPipe->mFront;
-    ALOG_ASSERT((0 <= ret) && (ret <= mMaxFrames));
+    ALOG_ASSERT((0 <= ret) && ((size_t) ret <= mPipe->mMaxFrames));
     return ret;
 }
 
diff --git a/media/libnbaio/Pipe.cpp b/media/libnbaio/Pipe.cpp
index 6e0ec8c..13f211d 100644
--- a/media/libnbaio/Pipe.cpp
+++ b/media/libnbaio/Pipe.cpp
@@ -21,7 +21,7 @@
 #include <cutils/compiler.h>
 #include <utils/Log.h>
 #include <media/nbaio/Pipe.h>
-#include <media/nbaio/roundup.h>
+#include <audio_utils/roundup.h>
 
 namespace android {
 
diff --git a/media/libnbaio/roundup.c b/media/libnbaio/roundup.c
deleted file mode 100644
index 1d552d1..0000000
--- a/media/libnbaio/roundup.c
+++ /dev/null
@@ -1,32 +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 <media/nbaio/roundup.h>
-
-unsigned roundup(unsigned v)
-{
-    // __builtin_clz is undefined for zero input
-    if (v == 0) {
-        v = 1;
-    }
-    int lz = __builtin_clz((int) v);
-    unsigned rounded = ((unsigned) 0x80000000) >> lz;
-    // 0x800000001 and higher are actually rounded _down_ to prevent overflow
-    if (v > rounded && lz > 0) {
-        rounded <<= 1;
-    }
-    return rounded;
-}
diff --git a/media/libstagefright/AACWriter.cpp b/media/libstagefright/AACWriter.cpp
index 2e41d80..9d90dbd 100644
--- a/media/libstagefright/AACWriter.cpp
+++ b/media/libstagefright/AACWriter.cpp
@@ -36,33 +36,19 @@
 
 namespace android {
 
-AACWriter::AACWriter(const char *filename)
-    : mFd(-1),
-      mInitCheck(NO_INIT),
-      mStarted(false),
-      mPaused(false),
-      mResumed(false),
-      mChannelCount(-1),
-      mSampleRate(-1),
-      mAACProfile(OMX_AUDIO_AACObjectLC) {
-
-    ALOGV("AACWriter Constructor");
-
-    mFd = open(filename, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
-    if (mFd >= 0) {
-        mInitCheck = OK;
-    }
-}
-
 AACWriter::AACWriter(int fd)
     : mFd(dup(fd)),
       mInitCheck(mFd < 0? NO_INIT: OK),
       mStarted(false),
       mPaused(false),
       mResumed(false),
+      mThread(0),
+      mEstimatedSizeBytes(0),
+      mEstimatedDurationUs(0),
       mChannelCount(-1),
       mSampleRate(-1),
-      mAACProfile(OMX_AUDIO_AACObjectLC) {
+      mAACProfile(OMX_AUDIO_AACObjectLC),
+      mFrameDurationUs(0) {
 }
 
 AACWriter::~AACWriter() {
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index c8806ae..7d313e0 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -419,6 +419,7 @@
       mMetaDataBuffersToSubmit(0),
       mRepeatFrameDelayUs(-1ll),
       mMaxPtsGapUs(-1ll),
+      mMaxFps(-1),
       mTimePerFrameUs(-1ll),
       mTimePerCaptureUs(-1ll),
       mCreateInputBuffersSuspended(false),
@@ -1259,6 +1260,10 @@
             mMaxPtsGapUs = -1ll;
         }
 
+        if (!msg->findFloat("max-fps-to-encoder", &mMaxFps)) {
+            mMaxFps = -1;
+        }
+
         if (!msg->findInt64("time-lapse", &mTimePerCaptureUs)) {
             mTimePerCaptureUs = -1ll;
         }
@@ -5110,6 +5115,21 @@
         }
     }
 
+    if (err == OK && mCodec->mMaxFps > 0) {
+        err = mCodec->mOMX->setInternalOption(
+                mCodec->mNode,
+                kPortIndexInput,
+                IOMX::INTERNAL_OPTION_MAX_FPS,
+                &mCodec->mMaxFps,
+                sizeof(mCodec->mMaxFps));
+
+        if (err != OK) {
+            ALOGE("[%s] Unable to configure max fps (err %d)",
+                    mCodec->mComponentName.c_str(),
+                    err);
+        }
+    }
+
     if (err == OK && mCodec->mTimePerCaptureUs > 0ll
             && mCodec->mTimePerFrameUs > 0ll) {
         int64_t timeLapse[2];
diff --git a/media/libstagefright/AMRWriter.cpp b/media/libstagefright/AMRWriter.cpp
index 9aa7d95..f53d7f0 100644
--- a/media/libstagefright/AMRWriter.cpp
+++ b/media/libstagefright/AMRWriter.cpp
@@ -31,19 +31,6 @@
 
 namespace android {
 
-AMRWriter::AMRWriter(const char *filename)
-    : mFd(-1),
-      mInitCheck(NO_INIT),
-      mStarted(false),
-      mPaused(false),
-      mResumed(false) {
-
-    mFd = open(filename, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
-    if (mFd >= 0) {
-        mInitCheck = OK;
-    }
-}
-
 AMRWriter::AMRWriter(int fd)
     : mFd(dup(fd)),
       mInitCheck(mFd < 0? NO_INIT: OK),
diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index 2629afc..6d9bbae 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -101,6 +101,7 @@
         libstagefright_color_conversion \
         libstagefright_aacenc \
         libstagefright_matroska \
+        libstagefright_mediafilter \
         libstagefright_webm \
         libstagefright_timedtext \
         libvpx \
@@ -108,13 +109,14 @@
         libstagefright_mpeg2ts \
         libstagefright_id3 \
         libFLAC \
-        libmedia_helper
+        libmedia_helper \
 
 LOCAL_SHARED_LIBRARIES += \
         libstagefright_enc_common \
         libstagefright_avc_common \
         libstagefright_foundation \
-        libdl
+        libdl \
+        libRScpp \
 
 LOCAL_CFLAGS += -Wno-multichar
 
diff --git a/media/libstagefright/FileSource.cpp b/media/libstagefright/FileSource.cpp
index a7ca3da..f0db76b 100644
--- a/media/libstagefright/FileSource.cpp
+++ b/media/libstagefright/FileSource.cpp
@@ -14,6 +14,10 @@
  * limitations under the License.
  */
 
+//#define LOG_NDEBUG 0
+#define LOG_TAG "FileSource"
+#include <utils/Log.h>
+
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/FileSource.h>
 #include <sys/types.h>
diff --git a/media/libstagefright/MPEG2TSWriter.cpp b/media/libstagefright/MPEG2TSWriter.cpp
index 9856f92..4359fb9 100644
--- a/media/libstagefright/MPEG2TSWriter.cpp
+++ b/media/libstagefright/MPEG2TSWriter.cpp
@@ -480,19 +480,6 @@
     init();
 }
 
-MPEG2TSWriter::MPEG2TSWriter(const char *filename)
-    : mFile(fopen(filename, "wb")),
-      mWriteCookie(NULL),
-      mWriteFunc(NULL),
-      mStarted(false),
-      mNumSourcesDone(0),
-      mNumTSPacketsWritten(0),
-      mNumTSPacketsBeforeMeta(0),
-      mPATContinuityCounter(0),
-      mPMTContinuityCounter(0) {
-    init();
-}
-
 MPEG2TSWriter::MPEG2TSWriter(
         void *cookie,
         ssize_t (*write)(void *cookie, const void *data, size_t size))
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index 8bf7f63..d0f42cc 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -354,6 +354,8 @@
 
 MPEG4Extractor::MPEG4Extractor(const sp<DataSource> &source)
     : mMoofOffset(0),
+      mMoofFound(false),
+      mMdatFound(false),
       mDataSource(source),
       mInitCheck(NO_INIT),
       mHasVideo(false),
@@ -490,7 +492,9 @@
 
     off64_t offset = 0;
     status_t err;
-    while (true) {
+    bool sawMoovOrSidx = false;
+
+    while (!(sawMoovOrSidx && (mMdatFound || mMoofFound))) {
         off64_t orig_offset = offset;
         err = parseChunk(&offset, 0);
 
@@ -502,23 +506,9 @@
             ALOGE("did not advance: 0x%lld->0x%lld", orig_offset, offset);
             err = ERROR_MALFORMED;
             break;
-        } else if (err == OK) {
-            continue;
+        } else if (err == UNKNOWN_ERROR) {
+            sawMoovOrSidx = true;
         }
-
-        uint32_t hdr[2];
-        if (mDataSource->readAt(offset, hdr, 8) < 8) {
-            break;
-        }
-        uint32_t chunk_type = ntohl(hdr[1]);
-        if (chunk_type == FOURCC('m', 'o', 'o', 'f')) {
-            // store the offset of the first segment
-            mMoofOffset = offset;
-        } else if (chunk_type != FOURCC('m', 'd', 'a', 't')) {
-            // keep parsing until we get to the data
-            continue;
-        }
-        break;
     }
 
     if (mInitCheck == OK) {
@@ -864,6 +854,12 @@
         case FOURCC('s', 'c', 'h', 'i'):
         case FOURCC('e', 'd', 't', 's'):
         {
+            if (chunk_type == FOURCC('m', 'o', 'o', 'f') && !mMoofFound) {
+                // store the offset of the first segment
+                mMoofFound = true;
+                mMoofOffset = *offset;
+            }
+
             if (chunk_type == FOURCC('s', 't', 'b', 'l')) {
                 ALOGV("sampleTable chunk is %" PRIu64 " bytes long.", chunk_size);
 
@@ -1830,6 +1826,9 @@
         case FOURCC('m', 'd', 'a', 't'):
         {
             ALOGV("mdat chunk, drm: %d", mIsDrm);
+
+            mMdatFound = true;
+
             if (!mIsDrm) {
                 *offset += chunk_size;
                 break;
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index 9f20b1d..beb6f20 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -345,31 +345,6 @@
     Track &operator=(const Track &);
 };
 
-MPEG4Writer::MPEG4Writer(const char *filename)
-    : mFd(-1),
-      mInitCheck(NO_INIT),
-      mIsRealTimeRecording(true),
-      mUse4ByteNalLength(true),
-      mUse32BitOffset(true),
-      mIsFileSizeLimitExplicitlyRequested(false),
-      mPaused(false),
-      mStarted(false),
-      mWriterThreadStarted(false),
-      mOffset(0),
-      mMdatOffset(0),
-      mEstimatedMoovBoxSize(0),
-      mInterleaveDurationUs(1000000),
-      mLatitudex10000(0),
-      mLongitudex10000(0),
-      mAreGeoTagsAvailable(false),
-      mStartTimeOffsetMs(-1) {
-
-    mFd = open(filename, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
-    if (mFd >= 0) {
-        mInitCheck = OK;
-    }
-}
-
 MPEG4Writer::MPEG4Writer(int fd)
     : mFd(dup(fd)),
       mInitCheck(mFd < 0? NO_INIT: OK),
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index 6ca123a..50e6bd0 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -36,6 +36,7 @@
 #include <media/stagefright/MediaCodecList.h>
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaFilter.h>
 #include <media/stagefright/MetaData.h>
 #include <media/stagefright/NativeWindowWrapper.h>
 #include <private/android_filesystem_config.h>
@@ -189,7 +190,16 @@
     // quickly, violating the OpenMAX specs, until that is remedied
     // we need to invest in an extra looper to free the main event
     // queue.
-    mCodec = new ACodec;
+
+    if (nameIsType || !strncasecmp(name.c_str(), "omx.", 4)) {
+        mCodec = new ACodec;
+    } else if (!nameIsType
+            && !strncasecmp(name.c_str(), "android.filter.", 15)) {
+        mCodec = new MediaFilter;
+    } else {
+        return NAME_NOT_FOUND;
+    }
+
     bool needDedicatedLooper = false;
     if (nameIsType && !strncasecmp(name.c_str(), "video/", 6)) {
         needDedicatedLooper = true;
diff --git a/media/libstagefright/MediaMuxer.cpp b/media/libstagefright/MediaMuxer.cpp
index c7c6f34..b13877d 100644
--- a/media/libstagefright/MediaMuxer.cpp
+++ b/media/libstagefright/MediaMuxer.cpp
@@ -38,21 +38,6 @@
 
 namespace android {
 
-MediaMuxer::MediaMuxer(const char *path, OutputFormat format)
-    : mFormat(format),
-      mState(UNINITIALIZED) {
-    if (format == OUTPUT_FORMAT_MPEG_4) {
-        mWriter = new MPEG4Writer(path);
-    } else if (format == OUTPUT_FORMAT_WEBM) {
-        mWriter = new WebmWriter(path);
-    }
-
-    if (mWriter != NULL) {
-        mFileMeta = new MetaData;
-        mState = INITIALIZED;
-    }
-}
-
 MediaMuxer::MediaMuxer(int fd, OutputFormat format)
     : mFormat(format),
       mState(UNINITIALIZED) {
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index b3a79a0..c0be136 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -344,6 +344,28 @@
         buffer->meta()->setInt32("csd", true);
         buffer->meta()->setInt64("timeUs", 0);
         msg->setBuffer("csd-0", buffer);
+
+        if (!meta->findData(kKeyOpusCodecDelay, &type, &data, &size)) {
+            return -EINVAL;
+        }
+
+        buffer = new ABuffer(size);
+        memcpy(buffer->data(), data, size);
+
+        buffer->meta()->setInt32("csd", true);
+        buffer->meta()->setInt64("timeUs", 0);
+        msg->setBuffer("csd-1", buffer);
+
+        if (!meta->findData(kKeyOpusSeekPreRoll, &type, &data, &size)) {
+            return -EINVAL;
+        }
+
+        buffer = new ABuffer(size);
+        memcpy(buffer->data(), data, size);
+
+        buffer->meta()->setInt32("csd", true);
+        buffer->meta()->setInt64("timeUs", 0);
+        msg->setBuffer("csd-2", buffer);
     }
 
     *format = msg;
diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
index 8a95643..6e6a78a 100644
--- a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
+++ b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
@@ -38,7 +38,10 @@
             NULL /* profileLevels */, 0 /* numProfileLevels */,
             320 /* width */, 240 /* height */, callbacks, appData, component),
       mMode(codingType == OMX_VIDEO_CodingVP8 ? MODE_VP8 : MODE_VP9),
+      mEOSStatus(INPUT_DATA_AVAILABLE),
       mCtx(NULL),
+      mFrameParallelMode(false),
+      mTimeStampIdx(0),
       mImg(NULL) {
     // arbitrary from avc/hevc as vpx does not specify a min compression ratio
     const size_t kMinCompressionRatio = mMode == MODE_VP8 ? 2 : 4;
@@ -51,9 +54,7 @@
 }
 
 SoftVPX::~SoftVPX() {
-    vpx_codec_destroy((vpx_codec_ctx_t *)mCtx);
-    delete (vpx_codec_ctx_t *)mCtx;
-    mCtx = NULL;
+    destroyDecoder();
 }
 
 static int GetCPUCoreCount() {
@@ -73,12 +74,19 @@
     mCtx = new vpx_codec_ctx_t;
     vpx_codec_err_t vpx_err;
     vpx_codec_dec_cfg_t cfg;
+    vpx_codec_flags_t flags;
     memset(&cfg, 0, sizeof(vpx_codec_dec_cfg_t));
+    memset(&flags, 0, sizeof(vpx_codec_flags_t));
     cfg.threads = GetCPUCoreCount();
+
+    if (mFrameParallelMode) {
+        flags |= VPX_CODEC_USE_FRAME_THREADING;
+    }
+
     if ((vpx_err = vpx_codec_dec_init(
                 (vpx_codec_ctx_t *)mCtx,
                  mMode == MODE_VP8 ? &vpx_codec_vp8_dx_algo : &vpx_codec_vp9_dx_algo,
-                 &cfg, 0))) {
+                 &cfg, flags))) {
         ALOGE("on2 decoder failed to initialize. (%d)", vpx_err);
         return UNKNOWN_ERROR;
     }
@@ -86,86 +94,155 @@
     return OK;
 }
 
+status_t SoftVPX::destroyDecoder() {
+    vpx_codec_destroy((vpx_codec_ctx_t *)mCtx);
+    delete (vpx_codec_ctx_t *)mCtx;
+    mCtx = NULL;
+    return OK;
+}
+
+bool SoftVPX::outputBuffers(bool flushDecoder, bool display, bool eos, bool *portWillReset) {
+    List<BufferInfo *> &inQueue = getPortQueue(0);
+    List<BufferInfo *> &outQueue = getPortQueue(1);
+    BufferInfo *outInfo = NULL;
+    OMX_BUFFERHEADERTYPE *outHeader = NULL;
+    vpx_codec_iter_t iter = NULL;
+
+    if (flushDecoder && mFrameParallelMode) {
+        // Flush decoder by passing NULL data ptr and 0 size.
+        // Ideally, this should never fail.
+        if (vpx_codec_decode((vpx_codec_ctx_t *)mCtx, NULL, 0, NULL, 0)) {
+            ALOGE("Failed to flush on2 decoder.");
+            return false;
+        }
+    }
+
+    if (!display) {
+        if (!flushDecoder) {
+            ALOGE("Invalid operation.");
+            return false;
+        }
+        // Drop all the decoded frames in decoder.
+        while ((mImg = vpx_codec_get_frame((vpx_codec_ctx_t *)mCtx, &iter))) {
+        }
+        return true;
+    }
+
+    while (!outQueue.empty()) {
+        if (mImg == NULL) {
+            mImg = vpx_codec_get_frame((vpx_codec_ctx_t *)mCtx, &iter);
+            if (mImg == NULL) {
+                break;
+            }
+        }
+        uint32_t width = mImg->d_w;
+        uint32_t height = mImg->d_h;
+        outInfo = *outQueue.begin();
+        outHeader = outInfo->mHeader;
+        CHECK_EQ(mImg->fmt, IMG_FMT_I420);
+        handlePortSettingsChange(portWillReset, width, height);
+        if (*portWillReset) {
+            return true;
+        }
+
+        outHeader->nOffset = 0;
+        outHeader->nFilledLen = (width * height * 3) / 2;
+        outHeader->nFlags = 0;
+        outHeader->nTimeStamp = *(OMX_TICKS *)mImg->user_priv;
+
+        uint8_t *dst = outHeader->pBuffer;
+        const uint8_t *srcY = (const uint8_t *)mImg->planes[PLANE_Y];
+        const uint8_t *srcU = (const uint8_t *)mImg->planes[PLANE_U];
+        const uint8_t *srcV = (const uint8_t *)mImg->planes[PLANE_V];
+        size_t srcYStride = mImg->stride[PLANE_Y];
+        size_t srcUStride = mImg->stride[PLANE_U];
+        size_t srcVStride = mImg->stride[PLANE_V];
+        copyYV12FrameToOutputBuffer(dst, srcY, srcU, srcV, srcYStride, srcUStride, srcVStride);
+
+        mImg = NULL;
+        outInfo->mOwnedByUs = false;
+        outQueue.erase(outQueue.begin());
+        outInfo = NULL;
+        notifyFillBufferDone(outHeader);
+        outHeader = NULL;
+    }
+
+    if (!eos) {
+        return true;
+    }
+
+    if (!outQueue.empty()) {
+        outInfo = *outQueue.begin();
+        outQueue.erase(outQueue.begin());
+        outHeader = outInfo->mHeader;
+        outHeader->nTimeStamp = 0;
+        outHeader->nFilledLen = 0;
+        outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+        outInfo->mOwnedByUs = false;
+        notifyFillBufferDone(outHeader);
+        mEOSStatus = OUTPUT_FRAMES_FLUSHED;
+    }
+    return true;
+}
+
 void SoftVPX::onQueueFilled(OMX_U32 /* portIndex */) {
-    if (mOutputPortSettingsChange != NONE) {
+    if (mOutputPortSettingsChange != NONE || mEOSStatus == OUTPUT_FRAMES_FLUSHED) {
         return;
     }
 
     List<BufferInfo *> &inQueue = getPortQueue(0);
     List<BufferInfo *> &outQueue = getPortQueue(1);
     bool EOSseen = false;
+    vpx_codec_err_t err;
+    bool portWillReset = false;
 
-    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) {
-            EOSseen = true;
-            if (inHeader->nFilledLen == 0) {
-                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 (mImg == NULL) {
-            if (vpx_codec_decode(
-                        (vpx_codec_ctx_t *)mCtx,
-                        inHeader->pBuffer + inHeader->nOffset,
-                        inHeader->nFilledLen,
-                        NULL,
-                        0)) {
-                ALOGE("on2 decoder failed to decode frame.");
-
+    while ((mEOSStatus == INPUT_EOS_SEEN || !inQueue.empty())
+            && !outQueue.empty()) {
+        // Output the pending frames that left from last port reset or decoder flush.
+        if (mEOSStatus == INPUT_EOS_SEEN || mImg != NULL) {
+            if (!outputBuffers(
+                     mEOSStatus == INPUT_EOS_SEEN, true /* display */,
+                     mEOSStatus == INPUT_EOS_SEEN, &portWillReset)) {
+                ALOGE("on2 decoder failed to output frame.");
                 notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
                 return;
             }
-            vpx_codec_iter_t iter = NULL;
-            mImg = vpx_codec_get_frame((vpx_codec_ctx_t *)mCtx, &iter);
-        }
-
-        if (mImg != NULL) {
-            CHECK_EQ(mImg->fmt, IMG_FMT_I420);
-
-            uint32_t width = mImg->d_w;
-            uint32_t height = mImg->d_h;
-            bool portWillReset = false;
-            handlePortSettingsChange(&portWillReset, width, height);
-            if (portWillReset) {
+            if (portWillReset || mEOSStatus == OUTPUT_FRAMES_FLUSHED ||
+                    mEOSStatus == INPUT_EOS_SEEN) {
                 return;
             }
+        }
 
-            outHeader->nOffset = 0;
-            outHeader->nFilledLen = (width * height * 3) / 2;
-            outHeader->nFlags = EOSseen ? OMX_BUFFERFLAG_EOS : 0;
-            outHeader->nTimeStamp = inHeader->nTimeStamp;
+        BufferInfo *inInfo = *inQueue.begin();
+        OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+        mTimeStamps[mTimeStampIdx] = inHeader->nTimeStamp;
 
-            uint8_t *dst = outHeader->pBuffer;
-            const uint8_t *srcY = (const uint8_t *)mImg->planes[PLANE_Y];
-            const uint8_t *srcU = (const uint8_t *)mImg->planes[PLANE_U];
-            const uint8_t *srcV = (const uint8_t *)mImg->planes[PLANE_V];
-            size_t srcYStride = mImg->stride[PLANE_Y];
-            size_t srcUStride = mImg->stride[PLANE_U];
-            size_t srcVStride = mImg->stride[PLANE_V];
-            copyYV12FrameToOutputBuffer(dst, srcY, srcU, srcV, srcYStride, srcUStride, srcVStride);
+        BufferInfo *outInfo = *outQueue.begin();
+        OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+        if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+            mEOSStatus = INPUT_EOS_SEEN;
+            EOSseen = true;
+        }
 
-            mImg = NULL;
-            outInfo->mOwnedByUs = false;
-            outQueue.erase(outQueue.begin());
-            outInfo = NULL;
-            notifyFillBufferDone(outHeader);
-            outHeader = NULL;
+        if (inHeader->nFilledLen > 0 &&
+            vpx_codec_decode((vpx_codec_ctx_t *)mCtx,
+                              inHeader->pBuffer + inHeader->nOffset,
+                              inHeader->nFilledLen,
+                              &mTimeStamps[mTimeStampIdx], 0)) {
+            ALOGE("on2 decoder failed to decode frame.");
+            notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+            return;
+        }
+        mTimeStampIdx = (mTimeStampIdx + 1) % kNumBuffers;
+
+        if (!outputBuffers(
+                 EOSseen /* flushDecoder */, true /* display */, EOSseen, &portWillReset)) {
+            ALOGE("on2 decoder failed to output frame.");
+            notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+            return;
+        }
+        if (portWillReset) {
+            return;
         }
 
         inInfo->mOwnedByUs = false;
@@ -176,6 +253,30 @@
     }
 }
 
+void SoftVPX::onPortFlushCompleted(OMX_U32 portIndex) {
+    if (portIndex == kInputPortIndex) {
+        bool portWillReset = false;
+        if (!outputBuffers(
+                 true /* flushDecoder */, false /* display */, false /* eos */, &portWillReset)) {
+            ALOGE("Failed to flush decoder.");
+            notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+            return;
+        }
+        mEOSStatus = INPUT_DATA_AVAILABLE;
+    }
+}
+
+void SoftVPX::onReset() {
+    bool portWillReset = false;
+    if (!outputBuffers(
+             true /* flushDecoder */, false /* display */, false /* eos */, &portWillReset)) {
+        ALOGW("Failed to flush decoder. Try to hard reset decoder");
+        destroyDecoder();
+        initDecoder();
+    }
+    mEOSStatus = INPUT_DATA_AVAILABLE;
+}
+
 }  // namespace android
 
 android::SoftOMXComponent *createSoftOMXComponent(
diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.h b/media/libstagefright/codecs/on2/dec/SoftVPX.h
index 8f68693..8ccbae2 100644
--- a/media/libstagefright/codecs/on2/dec/SoftVPX.h
+++ b/media/libstagefright/codecs/on2/dec/SoftVPX.h
@@ -38,6 +38,8 @@
     virtual ~SoftVPX();
 
     virtual void onQueueFilled(OMX_U32 portIndex);
+    virtual void onPortFlushCompleted(OMX_U32 portIndex);
+    virtual void onReset();
 
 private:
     enum {
@@ -49,11 +51,21 @@
         MODE_VP9
     } mMode;
 
-    void *mCtx;
+    enum {
+        INPUT_DATA_AVAILABLE,  // VPX component is ready to decode data.
+        INPUT_EOS_SEEN,        // VPX component saw EOS and is flushing On2 decoder.
+        OUTPUT_FRAMES_FLUSHED  // VPX component finished flushing On2 decoder.
+    } mEOSStatus;
 
+    void *mCtx;
+    bool mFrameParallelMode;  // Frame parallel is only supported by VP9 decoder.
+    OMX_TICKS mTimeStamps[kNumBuffers];
+    uint8_t mTimeStampIdx;
     vpx_image_t *mImg;
 
     status_t initDecoder();
+    status_t destroyDecoder();
+    bool outputBuffers(bool flushDecoder, bool display, bool eos, bool *portWillReset);
 
     DISALLOW_EVIL_CONSTRUCTORS(SoftVPX);
 };
diff --git a/media/libstagefright/codecs/opus/dec/SoftOpus.cpp b/media/libstagefright/codecs/opus/dec/SoftOpus.cpp
index b8084ae..6322dc2 100644
--- a/media/libstagefright/codecs/opus/dec/SoftOpus.cpp
+++ b/media/libstagefright/codecs/opus/dec/SoftOpus.cpp
@@ -345,9 +345,15 @@
             }
 
             uint8_t channel_mapping[kMaxChannels] = {0};
-            memcpy(&channel_mapping,
-                   kDefaultOpusChannelLayout,
-                   kMaxChannelsWithDefaultLayout);
+            if (mHeader->channels <= kMaxChannelsWithDefaultLayout) {
+                memcpy(&channel_mapping,
+                       kDefaultOpusChannelLayout,
+                       kMaxChannelsWithDefaultLayout);
+            } else {
+                memcpy(&channel_mapping,
+                       mHeader->stream_map,
+                       mHeader->channels);
+            }
 
             int status = OPUS_INVALID_STATE;
             mDecoder = opus_multistream_decoder_create(kRate,
diff --git a/media/libstagefright/colorconversion/SoftwareRenderer.cpp b/media/libstagefright/colorconversion/SoftwareRenderer.cpp
index 4e75250..21da707 100644
--- a/media/libstagefright/colorconversion/SoftwareRenderer.cpp
+++ b/media/libstagefright/colorconversion/SoftwareRenderer.cpp
@@ -98,33 +98,49 @@
     mCropWidth = mCropRight - mCropLeft + 1;
     mCropHeight = mCropBottom - mCropTop + 1;
 
-    int halFormat;
-    size_t bufWidth, bufHeight;
+    // by default convert everything to RGB565
+    int halFormat = HAL_PIXEL_FORMAT_RGB_565;
+    size_t bufWidth = mCropWidth;
+    size_t bufHeight = mCropHeight;
 
-    switch (mColorFormat) {
-        case OMX_COLOR_FormatYUV420Planar:
-        case OMX_TI_COLOR_FormatYUV420PackedSemiPlanar:
-        case OMX_COLOR_FormatYUV420SemiPlanar:
-        {
-            if (!runningInEmulator()) {
+    // hardware has YUV12 and RGBA8888 support, so convert known formats
+    if (!runningInEmulator()) {
+        switch (mColorFormat) {
+            case OMX_COLOR_FormatYUV420Planar:
+            case OMX_COLOR_FormatYUV420SemiPlanar:
+            case OMX_TI_COLOR_FormatYUV420PackedSemiPlanar:
+            {
                 halFormat = HAL_PIXEL_FORMAT_YV12;
                 bufWidth = (mCropWidth + 1) & ~1;
                 bufHeight = (mCropHeight + 1) & ~1;
                 break;
             }
-
-            // fall through.
+            case OMX_COLOR_Format24bitRGB888:
+            {
+                halFormat = HAL_PIXEL_FORMAT_RGB_888;
+                bufWidth = (mCropWidth + 1) & ~1;
+                bufHeight = (mCropHeight + 1) & ~1;
+                break;
+            }
+            case OMX_COLOR_Format32bitARGB8888:
+            case OMX_COLOR_Format32BitRGBA8888:
+            {
+                halFormat = HAL_PIXEL_FORMAT_RGBA_8888;
+                bufWidth = (mCropWidth + 1) & ~1;
+                bufHeight = (mCropHeight + 1) & ~1;
+                break;
+            }
+            default:
+            {
+                break;
+            }
         }
+    }
 
-        default:
-            halFormat = HAL_PIXEL_FORMAT_RGB_565;
-            bufWidth = mCropWidth;
-            bufHeight = mCropHeight;
-
-            mConverter = new ColorConverter(
-                    mColorFormat, OMX_COLOR_Format16bitRGB565);
-            CHECK(mConverter->isValid());
-            break;
+    if (halFormat == HAL_PIXEL_FORMAT_RGB_565) {
+        mConverter = new ColorConverter(
+                mColorFormat, OMX_COLOR_Format16bitRGB565);
+        CHECK(mConverter->isValid());
     }
 
     CHECK(mNativeWindow != NULL);
@@ -201,6 +217,8 @@
     CHECK_EQ(0, mapper.lock(
                 buf->handle, GRALLOC_USAGE_SW_WRITE_OFTEN, bounds, &dst));
 
+    // TODO move the other conversions also into ColorConverter, and
+    // fix cropping issues (when mCropLeft/Top != 0 or mWidth != mCropWidth)
     if (mConverter) {
         mConverter->convert(
                 data,
@@ -211,7 +229,8 @@
                 0, 0, mCropWidth - 1, mCropHeight - 1);
     } else if (mColorFormat == OMX_COLOR_FormatYUV420Planar) {
         const uint8_t *src_y = (const uint8_t *)data;
-        const uint8_t *src_u = (const uint8_t *)data + mWidth * mHeight;
+        const uint8_t *src_u =
+                (const uint8_t *)data + mWidth * mHeight;
         const uint8_t *src_v = src_u + (mWidth / 2 * mHeight / 2);
 
         uint8_t *dst_y = (uint8_t *)dst;
@@ -239,11 +258,9 @@
         }
     } else if (mColorFormat == OMX_TI_COLOR_FormatYUV420PackedSemiPlanar
             || mColorFormat == OMX_COLOR_FormatYUV420SemiPlanar) {
-        const uint8_t *src_y =
-            (const uint8_t *)data;
-
-        const uint8_t *src_uv =
-            (const uint8_t *)data + mWidth * (mHeight - mCropTop / 2);
+        const uint8_t *src_y = (const uint8_t *)data;
+        const uint8_t *src_uv = (const uint8_t *)data
+                + mWidth * (mHeight - mCropTop / 2);
 
         uint8_t *dst_y = (uint8_t *)dst;
 
@@ -271,6 +288,38 @@
             dst_u += dst_c_stride;
             dst_v += dst_c_stride;
         }
+    } else if (mColorFormat == OMX_COLOR_Format24bitRGB888) {
+        uint8_t* srcPtr = (uint8_t*)data;
+        uint8_t* dstPtr = (uint8_t*)dst;
+
+        for (size_t y = 0; y < (size_t)mCropHeight; ++y) {
+            memcpy(dstPtr, srcPtr, mCropWidth * 3);
+            srcPtr += mWidth * 3;
+            dstPtr += buf->stride * 3;
+        }
+    } else if (mColorFormat == OMX_COLOR_Format32bitARGB8888) {
+        uint8_t *srcPtr, *dstPtr;
+
+        for (size_t y = 0; y < (size_t)mCropHeight; ++y) {
+            srcPtr = (uint8_t*)data + mWidth * 4 * y;
+            dstPtr = (uint8_t*)dst + buf->stride * 4 * y;
+            for (size_t x = 0; x < (size_t)mCropWidth; ++x) {
+                uint8_t a = *srcPtr++;
+                for (size_t i = 0; i < 3; ++i) {   // copy RGB
+                    *dstPtr++ = *srcPtr++;
+                }
+                *dstPtr++ = a;  // alpha last (ARGB to RGBA)
+            }
+        }
+    } else if (mColorFormat == OMX_COLOR_Format32BitRGBA8888) {
+        uint8_t* srcPtr = (uint8_t*)data;
+        uint8_t* dstPtr = (uint8_t*)dst;
+
+        for (size_t y = 0; y < (size_t)mCropHeight; ++y) {
+            memcpy(dstPtr, srcPtr, mCropWidth * 4);
+            srcPtr += mWidth * 4;
+            dstPtr += buf->stride * 4;
+        }
     } else {
         LOG_ALWAYS_FATAL("bad color format %#x", mColorFormat);
     }
diff --git a/media/libstagefright/filters/Android.mk b/media/libstagefright/filters/Android.mk
new file mode 100644
index 0000000..36ab444
--- /dev/null
+++ b/media/libstagefright/filters/Android.mk
@@ -0,0 +1,27 @@
+LOCAL_PATH:= $(call my-dir)
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+        ColorConvert.cpp          \
+        GraphicBufferListener.cpp \
+        IntrinsicBlurFilter.cpp   \
+        MediaFilter.cpp           \
+        RSFilter.cpp              \
+        SaturationFilter.cpp      \
+        saturationARGB.rs         \
+        SimpleFilter.cpp          \
+        ZeroFilter.cpp
+
+LOCAL_C_INCLUDES := \
+        $(TOP)/frameworks/native/include/media/openmax \
+        $(TOP)/frameworks/rs/cpp \
+        $(TOP)/frameworks/rs \
+
+intermediates := $(call intermediates-dir-for,STATIC_LIBRARIES,libRS,TARGET,)
+LOCAL_C_INCLUDES += $(intermediates)
+
+LOCAL_CFLAGS += -Wno-multichar
+
+LOCAL_MODULE:= libstagefright_mediafilter
+
+include $(BUILD_STATIC_LIBRARY)
diff --git a/media/libstagefright/filters/ColorConvert.cpp b/media/libstagefright/filters/ColorConvert.cpp
new file mode 100644
index 0000000..a5039f9
--- /dev/null
+++ b/media/libstagefright/filters/ColorConvert.cpp
@@ -0,0 +1,111 @@
+/*
+ * Copyright (C) 2014 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 "ColorConvert.h"
+
+#ifndef max
+#define max(a,b) ((a) > (b) ? (a) : (b))
+#endif
+#ifndef min
+#define min(a,b) ((a) < (b) ? (a) : (b))
+#endif
+
+namespace android {
+
+void YUVToRGB(
+        int32_t y, int32_t u, int32_t v,
+        int32_t* r, int32_t* g, int32_t* b) {
+    y -= 16;
+    u -= 128;
+    v -= 128;
+
+    *b = 1192 * y + 2066 * u;
+    *g = 1192 * y - 833 * v - 400 * u;
+    *r = 1192 * y + 1634 * v;
+
+    *r = min(262143, max(0, *r));
+    *g = min(262143, max(0, *g));
+    *b = min(262143, max(0, *b));
+
+    *r >>= 10;
+    *g >>= 10;
+    *b >>= 10;
+}
+
+void convertYUV420spToARGB(
+        uint8_t *pY, uint8_t *pUV, int32_t width, int32_t height,
+        uint8_t *dest) {
+    const int32_t bytes_per_pixel = 2;
+
+    for (int32_t i = 0; i < height; i++) {
+        for (int32_t j = 0; j < width; j++) {
+            int32_t y = *(pY + i * width + j);
+            int32_t u = *(pUV + (i/2) * width + bytes_per_pixel * (j/2));
+            int32_t v = *(pUV + (i/2) * width + bytes_per_pixel * (j/2) + 1);
+
+            int32_t r, g, b;
+            YUVToRGB(y, u, v, &r, &g, &b);
+
+            *dest++ = 0xFF;
+            *dest++ = r;
+            *dest++ = g;
+            *dest++ = b;
+        }
+    }
+}
+
+void convertYUV420spToRGB888(
+        uint8_t *pY, uint8_t *pUV, int32_t width, int32_t height,
+        uint8_t *dest) {
+    const int32_t bytes_per_pixel = 2;
+
+    for (int32_t i = 0; i < height; i++) {
+        for (int32_t j = 0; j < width; j++) {
+            int32_t y = *(pY + i * width + j);
+            int32_t u = *(pUV + (i/2) * width + bytes_per_pixel * (j/2));
+            int32_t v = *(pUV + (i/2) * width + bytes_per_pixel * (j/2) + 1);
+
+            int32_t r, g, b;
+            YUVToRGB(y, u, v, &r, &g, &b);
+
+            *dest++ = r;
+            *dest++ = g;
+            *dest++ = b;
+        }
+    }
+}
+
+// HACK - not even slightly optimized
+// TODO: remove when RGBA support is added to SoftwareRenderer
+void convertRGBAToARGB(
+        uint8_t *src, int32_t width, int32_t height, uint32_t stride,
+        uint8_t *dest) {
+    for (size_t i = 0; i < height; ++i) {
+        for (size_t j = 0; j < width; ++j) {
+            uint8_t r = *src++;
+            uint8_t g = *src++;
+            uint8_t b = *src++;
+            uint8_t a = *src++;
+            *dest++ = a;
+            *dest++ = r;
+            *dest++ = g;
+            *dest++ = b;
+        }
+        src += (stride - width) * 4;
+    }
+}
+
+}   // namespace android
diff --git a/media/libstagefright/filters/ColorConvert.h b/media/libstagefright/filters/ColorConvert.h
new file mode 100644
index 0000000..13faa02
--- /dev/null
+++ b/media/libstagefright/filters/ColorConvert.h
@@ -0,0 +1,43 @@
+/*
+ * Copyright (C) 2014 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 COLOR_CONVERT_H_
+#define COLOR_CONVERT_H_
+
+#include <inttypes.h>
+
+namespace android {
+
+void YUVToRGB(
+        int32_t y, int32_t u, int32_t v,
+        int32_t* r, int32_t* g, int32_t* b);
+
+void convertYUV420spToARGB(
+        uint8_t *pY, uint8_t *pUV, int32_t width, int32_t height,
+        uint8_t *dest);
+
+void convertYUV420spToRGB888(
+        uint8_t *pY, uint8_t *pUV, int32_t width, int32_t height,
+        uint8_t *dest);
+
+// TODO: remove when RGBA support is added to SoftwareRenderer
+void convertRGBAToARGB(
+        uint8_t *src, int32_t width, int32_t height, uint32_t stride,
+        uint8_t *dest);
+
+}   // namespace android
+
+#endif  // COLOR_CONVERT_H_
diff --git a/media/libstagefright/filters/GraphicBufferListener.cpp b/media/libstagefright/filters/GraphicBufferListener.cpp
new file mode 100644
index 0000000..fa38192
--- /dev/null
+++ b/media/libstagefright/filters/GraphicBufferListener.cpp
@@ -0,0 +1,154 @@
+/*
+ * Copyright (C) 2014 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 "GraphicBufferListener"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/MediaErrors.h>
+
+#include "GraphicBufferListener.h"
+
+namespace android {
+
+status_t GraphicBufferListener::init(
+        const sp<AMessage> &notify,
+        size_t bufferWidth, size_t bufferHeight, size_t bufferCount) {
+    mNotify = notify;
+
+    String8 name("GraphicBufferListener");
+    BufferQueue::createBufferQueue(&mProducer, &mConsumer);
+    mConsumer->setConsumerName(name);
+    mConsumer->setDefaultBufferSize(bufferWidth, bufferHeight);
+    mConsumer->setConsumerUsageBits(GRALLOC_USAGE_SW_READ_OFTEN);
+
+    status_t err = mConsumer->setMaxAcquiredBufferCount(bufferCount);
+    if (err != NO_ERROR) {
+        ALOGE("Unable to set BQ max acquired buffer count to %u: %d",
+                bufferCount, err);
+        return err;
+    }
+
+    wp<BufferQueue::ConsumerListener> listener =
+        static_cast<BufferQueue::ConsumerListener*>(this);
+    sp<BufferQueue::ProxyConsumerListener> proxy =
+        new BufferQueue::ProxyConsumerListener(listener);
+
+    err = mConsumer->consumerConnect(proxy, false);
+    if (err != NO_ERROR) {
+        ALOGE("Error connecting to BufferQueue: %s (%d)",
+                strerror(-err), err);
+        return err;
+    }
+
+    ALOGV("init() successful.");
+
+    return OK;
+}
+
+void GraphicBufferListener::onFrameAvailable(const BufferItem& /* item */) {
+    ALOGV("onFrameAvailable() called");
+
+    {
+        Mutex::Autolock autoLock(mMutex);
+        mNumFramesAvailable++;
+    }
+
+    sp<AMessage> notify = mNotify->dup();
+    mNotify->setWhat(kWhatFrameAvailable);
+    mNotify->post();
+}
+
+void GraphicBufferListener::onBuffersReleased() {
+    ALOGV("onBuffersReleased() called");
+    // nothing to do
+}
+
+void GraphicBufferListener::onSidebandStreamChanged() {
+    ALOGW("GraphicBufferListener cannot consume sideband streams.");
+    // nothing to do
+}
+
+BufferQueue::BufferItem GraphicBufferListener::getBufferItem() {
+    BufferQueue::BufferItem item;
+
+    {
+        Mutex::Autolock autoLock(mMutex);
+        if (mNumFramesAvailable <= 0) {
+            ALOGE("getBuffer() called with no frames available");
+            return item;
+        }
+        mNumFramesAvailable--;
+    }
+
+    status_t err = mConsumer->acquireBuffer(&item, 0);
+    if (err == BufferQueue::NO_BUFFER_AVAILABLE) {
+        // shouldn't happen, since we track num frames available
+        ALOGE("frame was not available");
+        item.mBuf = -1;
+        return item;
+    } else if (err != OK) {
+        ALOGE("acquireBuffer returned err=%d", err);
+        item.mBuf = -1;
+        return item;
+    }
+
+    // Wait for it to become available.
+    err = item.mFence->waitForever("GraphicBufferListener::getBufferItem");
+    if (err != OK) {
+        ALOGW("failed to wait for buffer fence: %d", err);
+        // keep going
+    }
+
+    // If this is the first time we're seeing this buffer, add it to our
+    // slot table.
+    if (item.mGraphicBuffer != NULL) {
+        ALOGV("setting mBufferSlot %d", item.mBuf);
+        mBufferSlot[item.mBuf] = item.mGraphicBuffer;
+    }
+
+    return item;
+}
+
+sp<GraphicBuffer> GraphicBufferListener::getBuffer(
+        BufferQueue::BufferItem item) {
+    sp<GraphicBuffer> buf;
+    if (item.mBuf < 0 || item.mBuf >= BufferQueue::NUM_BUFFER_SLOTS) {
+        ALOGE("getBuffer() received invalid BufferItem: mBuf==%d", item.mBuf);
+        return buf;
+    }
+
+    buf = mBufferSlot[item.mBuf];
+    CHECK(buf.get() != NULL);
+
+    return buf;
+}
+
+status_t GraphicBufferListener::releaseBuffer(
+        BufferQueue::BufferItem item) {
+    if (item.mBuf < 0 || item.mBuf >= BufferQueue::NUM_BUFFER_SLOTS) {
+        ALOGE("getBuffer() received invalid BufferItem: mBuf==%d", item.mBuf);
+        return ERROR_OUT_OF_RANGE;
+    }
+
+    mConsumer->releaseBuffer(item.mBuf, item.mFrameNumber,
+            EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, Fence::NO_FENCE);
+
+    return OK;
+}
+
+}   // namespace android
diff --git a/media/libstagefright/filters/GraphicBufferListener.h b/media/libstagefright/filters/GraphicBufferListener.h
new file mode 100644
index 0000000..b3e0ee3
--- /dev/null
+++ b/media/libstagefright/filters/GraphicBufferListener.h
@@ -0,0 +1,70 @@
+/*
+ * Copyright (C) 2014 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 GRAPHIC_BUFFER_LISTENER_H_
+#define GRAPHIC_BUFFER_LISTENER_H_
+
+#include <gui/BufferQueue.h>
+
+namespace android {
+
+struct AMessage;
+
+struct GraphicBufferListener : public BufferQueue::ConsumerListener {
+public:
+    GraphicBufferListener() {};
+
+    status_t init(
+            const sp<AMessage> &notify,
+            size_t bufferWidth, size_t bufferHeight, size_t bufferCount);
+
+    virtual void onFrameAvailable(const BufferItem& item);
+    virtual void onBuffersReleased();
+    virtual void onSidebandStreamChanged();
+
+    // Returns the handle to the producer side of the BufferQueue.  Buffers
+    // queued on this will be received by GraphicBufferListener.
+    sp<IGraphicBufferProducer> getIGraphicBufferProducer() const {
+        return mProducer;
+    }
+
+    BufferQueue::BufferItem getBufferItem();
+    sp<GraphicBuffer> getBuffer(BufferQueue::BufferItem item);
+    status_t releaseBuffer(BufferQueue::BufferItem item);
+
+    enum {
+        kWhatFrameAvailable = 'frav',
+    };
+
+private:
+    sp<AMessage> mNotify;
+    size_t mNumFramesAvailable;
+
+    mutable Mutex mMutex;
+
+    // Our BufferQueue interfaces. mProducer is passed to the producer through
+    // getIGraphicBufferProducer, and mConsumer is used internally to retrieve
+    // the buffers queued by the producer.
+    sp<IGraphicBufferProducer> mProducer;
+    sp<IGraphicBufferConsumer> mConsumer;
+
+    // Cache of GraphicBuffers from the buffer queue.
+    sp<GraphicBuffer> mBufferSlot[BufferQueue::NUM_BUFFER_SLOTS];
+};
+
+}   // namespace android
+
+#endif  // GRAPHIC_BUFFER_LISTENER_H
diff --git a/media/libstagefright/filters/IntrinsicBlurFilter.cpp b/media/libstagefright/filters/IntrinsicBlurFilter.cpp
new file mode 100644
index 0000000..cbcf699
--- /dev/null
+++ b/media/libstagefright/filters/IntrinsicBlurFilter.cpp
@@ -0,0 +1,99 @@
+/*
+ * Copyright (C) 2014 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 "IntrinsicBlurFilter"
+
+#include <utils/Log.h>
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include "IntrinsicBlurFilter.h"
+
+namespace android {
+
+status_t IntrinsicBlurFilter::configure(const sp<AMessage> &msg) {
+    status_t err = SimpleFilter::configure(msg);
+    if (err != OK) {
+        return err;
+    }
+
+    if (!msg->findString("cacheDir", &mCacheDir)) {
+        ALOGE("Failed to find cache directory in config message.");
+        return NAME_NOT_FOUND;
+    }
+
+    return OK;
+}
+
+status_t IntrinsicBlurFilter::start() {
+    // TODO: use a single RS context object for entire application
+    mRS = new RSC::RS();
+
+    if (!mRS->init(mCacheDir.c_str())) {
+        ALOGE("Failed to initialize RenderScript context.");
+        return NO_INIT;
+    }
+
+    // 32-bit elements for ARGB8888
+    RSC::sp<const RSC::Element> e = RSC::Element::U8_4(mRS);
+
+    RSC::Type::Builder tb(mRS, e);
+    tb.setX(mWidth);
+    tb.setY(mHeight);
+    RSC::sp<const RSC::Type> t = tb.create();
+
+    mAllocIn = RSC::Allocation::createTyped(mRS, t);
+    mAllocOut = RSC::Allocation::createTyped(mRS, t);
+
+    mBlur = RSC::ScriptIntrinsicBlur::create(mRS, e);
+    mBlur->setRadius(mBlurRadius);
+    mBlur->setInput(mAllocIn);
+
+    return OK;
+}
+
+void IntrinsicBlurFilter::reset() {
+    mBlur.clear();
+    mAllocOut.clear();
+    mAllocIn.clear();
+    mRS.clear();
+}
+
+status_t IntrinsicBlurFilter::setParameters(const sp<AMessage> &msg) {
+    sp<AMessage> params;
+    CHECK(msg->findMessage("params", &params));
+
+    float blurRadius;
+    if (params->findFloat("blur-radius", &blurRadius)) {
+        mBlurRadius = blurRadius;
+    }
+
+    return OK;
+}
+
+status_t IntrinsicBlurFilter::processBuffers(
+        const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) {
+    mAllocIn->copy1DRangeFrom(0, mWidth * mHeight, srcBuffer->data());
+    mBlur->forEach(mAllocOut);
+    mAllocOut->copy1DRangeTo(0, mWidth * mHeight, outBuffer->data());
+
+    return OK;
+}
+
+}   // namespace android
diff --git a/media/libstagefright/filters/IntrinsicBlurFilter.h b/media/libstagefright/filters/IntrinsicBlurFilter.h
new file mode 100644
index 0000000..4707ab7
--- /dev/null
+++ b/media/libstagefright/filters/IntrinsicBlurFilter.h
@@ -0,0 +1,50 @@
+/*
+ * Copyright (C) 2014 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 INTRINSIC_BLUR_FILTER_H_
+#define INTRINSIC_BLUR_FILTER_H_
+
+#include "RenderScript.h"
+#include "SimpleFilter.h"
+
+namespace android {
+
+struct IntrinsicBlurFilter : public SimpleFilter {
+public:
+    IntrinsicBlurFilter() : mBlurRadius(1.f) {};
+
+    virtual status_t configure(const sp<AMessage> &msg);
+    virtual status_t start();
+    virtual void reset();
+    virtual status_t setParameters(const sp<AMessage> &msg);
+    virtual status_t processBuffers(
+            const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer);
+
+protected:
+    virtual ~IntrinsicBlurFilter() {};
+
+private:
+    AString mCacheDir;
+    RSC::sp<RSC::RS> mRS;
+    RSC::sp<RSC::Allocation> mAllocIn;
+    RSC::sp<RSC::Allocation> mAllocOut;
+    RSC::sp<RSC::ScriptIntrinsicBlur> mBlur;
+    float mBlurRadius;
+};
+
+}   // namespace android
+
+#endif  // INTRINSIC_BLUR_FILTER_H_
diff --git a/media/libstagefright/filters/MediaFilter.cpp b/media/libstagefright/filters/MediaFilter.cpp
new file mode 100644
index 0000000..c5289b6
--- /dev/null
+++ b/media/libstagefright/filters/MediaFilter.cpp
@@ -0,0 +1,816 @@
+/*
+ * Copyright (C) 2014 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 "MediaFilter"
+
+#include <inttypes.h>
+#include <utils/Trace.h>
+
+#include <binder/MemoryDealer.h>
+
+#include <media/stagefright/BufferProducerWrapper.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaFilter.h>
+
+#include "ColorConvert.h"
+#include "GraphicBufferListener.h"
+#include "IntrinsicBlurFilter.h"
+#include "RSFilter.h"
+#include "SaturationFilter.h"
+#include "ZeroFilter.h"
+
+namespace android {
+
+// parameter: number of input and output buffers
+static const size_t kBufferCountActual = 4;
+
+MediaFilter::MediaFilter()
+    : mState(UNINITIALIZED),
+      mGeneration(0),
+      mGraphicBufferListener(NULL) {
+}
+
+MediaFilter::~MediaFilter() {
+}
+
+//////////////////// PUBLIC FUNCTIONS //////////////////////////////////////////
+
+void MediaFilter::setNotificationMessage(const sp<AMessage> &msg) {
+    mNotify = msg;
+}
+
+void MediaFilter::initiateAllocateComponent(const sp<AMessage> &msg) {
+    msg->setWhat(kWhatAllocateComponent);
+    msg->setTarget(id());
+    msg->post();
+}
+
+void MediaFilter::initiateConfigureComponent(const sp<AMessage> &msg) {
+    msg->setWhat(kWhatConfigureComponent);
+    msg->setTarget(id());
+    msg->post();
+}
+
+void MediaFilter::initiateCreateInputSurface() {
+    (new AMessage(kWhatCreateInputSurface, id()))->post();
+}
+
+void MediaFilter::initiateStart() {
+    (new AMessage(kWhatStart, id()))->post();
+}
+
+void MediaFilter::initiateShutdown(bool keepComponentAllocated) {
+    sp<AMessage> msg = new AMessage(kWhatShutdown, id());
+    msg->setInt32("keepComponentAllocated", keepComponentAllocated);
+    msg->post();
+}
+
+void MediaFilter::signalFlush() {
+    (new AMessage(kWhatFlush, id()))->post();
+}
+
+void MediaFilter::signalResume() {
+    (new AMessage(kWhatResume, id()))->post();
+}
+
+// nothing to do
+void MediaFilter::signalRequestIDRFrame() {
+    return;
+}
+
+void MediaFilter::signalSetParameters(const sp<AMessage> &params) {
+    sp<AMessage> msg = new AMessage(kWhatSetParameters, id());
+    msg->setMessage("params", params);
+    msg->post();
+}
+
+void MediaFilter::signalEndOfInputStream() {
+    (new AMessage(kWhatSignalEndOfInputStream, id()))->post();
+}
+
+void MediaFilter::onMessageReceived(const sp<AMessage> &msg) {
+    switch (msg->what()) {
+        case kWhatAllocateComponent:
+        {
+            onAllocateComponent(msg);
+            break;
+        }
+        case kWhatConfigureComponent:
+        {
+            onConfigureComponent(msg);
+            break;
+        }
+        case kWhatStart:
+        {
+            onStart();
+            break;
+        }
+        case kWhatProcessBuffers:
+        {
+            processBuffers();
+            break;
+        }
+        case kWhatInputBufferFilled:
+        {
+            onInputBufferFilled(msg);
+            break;
+        }
+        case kWhatOutputBufferDrained:
+        {
+            onOutputBufferDrained(msg);
+            break;
+        }
+        case kWhatShutdown:
+        {
+            onShutdown(msg);
+            break;
+        }
+        case kWhatFlush:
+        {
+            onFlush();
+            break;
+        }
+        case kWhatResume:
+        {
+            // nothing to do
+            break;
+        }
+        case kWhatSetParameters:
+        {
+            onSetParameters(msg);
+            break;
+        }
+        case kWhatCreateInputSurface:
+        {
+            onCreateInputSurface();
+            break;
+        }
+        case GraphicBufferListener::kWhatFrameAvailable:
+        {
+            onInputFrameAvailable();
+            break;
+        }
+        case kWhatSignalEndOfInputStream:
+        {
+            onSignalEndOfInputStream();
+            break;
+        }
+        default:
+        {
+            ALOGE("Message not handled:\n%s", msg->debugString().c_str());
+            break;
+        }
+    }
+}
+
+//////////////////// PORT DESCRIPTION //////////////////////////////////////////
+
+MediaFilter::PortDescription::PortDescription() {
+}
+
+void MediaFilter::PortDescription::addBuffer(
+        IOMX::buffer_id id, const sp<ABuffer> &buffer) {
+    mBufferIDs.push_back(id);
+    mBuffers.push_back(buffer);
+}
+
+size_t MediaFilter::PortDescription::countBuffers() {
+    return mBufferIDs.size();
+}
+
+IOMX::buffer_id MediaFilter::PortDescription::bufferIDAt(size_t index) const {
+    return mBufferIDs.itemAt(index);
+}
+
+sp<ABuffer> MediaFilter::PortDescription::bufferAt(size_t index) const {
+    return mBuffers.itemAt(index);
+}
+
+//////////////////// HELPER FUNCTIONS //////////////////////////////////////////
+
+void MediaFilter::signalProcessBuffers() {
+    (new AMessage(kWhatProcessBuffers, id()))->post();
+}
+
+void MediaFilter::signalError(status_t error) {
+    sp<AMessage> notify = mNotify->dup();
+    notify->setInt32("what", CodecBase::kWhatError);
+    notify->setInt32("err", error);
+    notify->post();
+}
+
+status_t MediaFilter::allocateBuffersOnPort(OMX_U32 portIndex) {
+    CHECK(portIndex == kPortIndexInput || portIndex == kPortIndexOutput);
+    const bool isInput = portIndex == kPortIndexInput;
+    const size_t bufferSize = isInput ? mMaxInputSize : mMaxOutputSize;
+
+    CHECK(mDealer[portIndex] == NULL);
+    CHECK(mBuffers[portIndex].isEmpty());
+
+    ALOGV("Allocating %zu buffers of size %zu on %s port",
+            kBufferCountActual, bufferSize,
+            isInput ? "input" : "output");
+
+    size_t totalSize = kBufferCountActual * bufferSize;
+
+    mDealer[portIndex] = new MemoryDealer(totalSize, "MediaFilter");
+
+    for (size_t i = 0; i < kBufferCountActual; ++i) {
+        sp<IMemory> mem = mDealer[portIndex]->allocate(bufferSize);
+        CHECK(mem.get() != NULL);
+
+        BufferInfo info;
+        info.mStatus = BufferInfo::OWNED_BY_US;
+        info.mBufferID = i;
+        info.mGeneration = mGeneration;
+        info.mOutputFlags = 0;
+        info.mData = new ABuffer(mem->pointer(), bufferSize);
+        info.mData->meta()->setInt64("timeUs", 0);
+
+        mBuffers[portIndex].push_back(info);
+
+        if (!isInput) {
+            mAvailableOutputBuffers.push(
+                    &mBuffers[portIndex].editItemAt(i));
+        }
+    }
+
+    sp<AMessage> notify = mNotify->dup();
+    notify->setInt32("what", CodecBase::kWhatBuffersAllocated);
+
+    notify->setInt32("portIndex", portIndex);
+
+    sp<PortDescription> desc = new PortDescription;
+
+    for (size_t i = 0; i < mBuffers[portIndex].size(); ++i) {
+        const BufferInfo &info = mBuffers[portIndex][i];
+
+        desc->addBuffer(info.mBufferID, info.mData);
+    }
+
+    notify->setObject("portDesc", desc);
+    notify->post();
+
+    return OK;
+}
+
+MediaFilter::BufferInfo* MediaFilter::findBufferByID(
+        uint32_t portIndex, IOMX::buffer_id bufferID,
+        ssize_t *index) {
+    for (size_t i = 0; i < mBuffers[portIndex].size(); ++i) {
+        BufferInfo *info = &mBuffers[portIndex].editItemAt(i);
+
+        if (info->mBufferID == bufferID) {
+            if (index != NULL) {
+                *index = i;
+            }
+            return info;
+        }
+    }
+
+    TRESPASS();
+
+    return NULL;
+}
+
+void MediaFilter::postFillThisBuffer(BufferInfo *info) {
+    ALOGV("postFillThisBuffer on buffer %d", info->mBufferID);
+    if (mPortEOS[kPortIndexInput]) {
+        return;
+    }
+
+    CHECK_EQ((int)info->mStatus, (int)BufferInfo::OWNED_BY_US);
+
+    info->mGeneration = mGeneration;
+
+    sp<AMessage> notify = mNotify->dup();
+    notify->setInt32("what", CodecBase::kWhatFillThisBuffer);
+    notify->setInt32("buffer-id", info->mBufferID);
+
+    info->mData->meta()->clear();
+    notify->setBuffer("buffer", info->mData);
+
+    sp<AMessage> reply = new AMessage(kWhatInputBufferFilled, id());
+    reply->setInt32("buffer-id", info->mBufferID);
+
+    notify->setMessage("reply", reply);
+
+    info->mStatus = BufferInfo::OWNED_BY_UPSTREAM;
+    notify->post();
+}
+
+void MediaFilter::postDrainThisBuffer(BufferInfo *info) {
+    CHECK_EQ((int)info->mStatus, (int)BufferInfo::OWNED_BY_US);
+
+    info->mGeneration = mGeneration;
+
+    sp<AMessage> notify = mNotify->dup();
+    notify->setInt32("what", CodecBase::kWhatDrainThisBuffer);
+    notify->setInt32("buffer-id", info->mBufferID);
+    notify->setInt32("flags", info->mOutputFlags);
+    notify->setBuffer("buffer", info->mData);
+
+    sp<AMessage> reply = new AMessage(kWhatOutputBufferDrained, id());
+    reply->setInt32("buffer-id", info->mBufferID);
+
+    notify->setMessage("reply", reply);
+
+    notify->post();
+
+    info->mStatus = BufferInfo::OWNED_BY_UPSTREAM;
+}
+
+void MediaFilter::postEOS() {
+    sp<AMessage> notify = mNotify->dup();
+    notify->setInt32("what", CodecBase::kWhatEOS);
+    notify->setInt32("err", ERROR_END_OF_STREAM);
+    notify->post();
+
+    ALOGV("Sent kWhatEOS.");
+}
+
+void MediaFilter::sendFormatChange() {
+    sp<AMessage> notify = mNotify->dup();
+
+    notify->setInt32("what", kWhatOutputFormatChanged);
+
+    AString mime;
+    CHECK(mOutputFormat->findString("mime", &mime));
+    notify->setString("mime", mime.c_str());
+
+    notify->setInt32("stride", mStride);
+    notify->setInt32("slice-height", mSliceHeight);
+    notify->setInt32("color-format", mColorFormatOut);
+    notify->setRect("crop", 0, 0, mStride - 1, mSliceHeight - 1);
+    notify->setInt32("width", mWidth);
+    notify->setInt32("height", mHeight);
+
+    notify->post();
+}
+
+void MediaFilter::requestFillEmptyInput() {
+    if (mPortEOS[kPortIndexInput]) {
+        return;
+    }
+
+    for (size_t i = 0; i < mBuffers[kPortIndexInput].size(); ++i) {
+        BufferInfo *info = &mBuffers[kPortIndexInput].editItemAt(i);
+
+        if (info->mStatus == BufferInfo::OWNED_BY_US) {
+            postFillThisBuffer(info);
+        }
+    }
+}
+
+void MediaFilter::processBuffers() {
+    if (mAvailableInputBuffers.empty() || mAvailableOutputBuffers.empty()) {
+        ALOGV("Skipping process (buffers unavailable)");
+        return;
+    }
+
+    if (mPortEOS[kPortIndexOutput]) {
+        // TODO notify caller of queueInput error when it is supported
+        // in MediaCodec
+        ALOGW("Tried to process a buffer after EOS.");
+        return;
+    }
+
+    BufferInfo *inputInfo = mAvailableInputBuffers[0];
+    mAvailableInputBuffers.removeAt(0);
+    BufferInfo *outputInfo = mAvailableOutputBuffers[0];
+    mAvailableOutputBuffers.removeAt(0);
+
+    status_t err;
+    err = mFilter->processBuffers(inputInfo->mData, outputInfo->mData);
+    if (err != (status_t)OK) {
+        outputInfo->mData->meta()->setInt32("err", err);
+    }
+
+    int64_t timeUs;
+    CHECK(inputInfo->mData->meta()->findInt64("timeUs", &timeUs));
+    outputInfo->mData->meta()->setInt64("timeUs", timeUs);
+    outputInfo->mOutputFlags = 0;
+    int32_t eos = 0;
+    if (inputInfo->mData->meta()->findInt32("eos", &eos) && eos != 0) {
+        outputInfo->mOutputFlags |= OMX_BUFFERFLAG_EOS;
+        mPortEOS[kPortIndexOutput] = true;
+        outputInfo->mData->meta()->setInt32("eos", eos);
+        postEOS();
+        ALOGV("Output stream saw EOS.");
+    }
+
+    ALOGV("Processed input buffer %u [%zu], output buffer %u [%zu]",
+                inputInfo->mBufferID, inputInfo->mData->size(),
+                outputInfo->mBufferID, outputInfo->mData->size());
+
+    if (mGraphicBufferListener != NULL) {
+        delete inputInfo;
+    } else {
+        postFillThisBuffer(inputInfo);
+    }
+    postDrainThisBuffer(outputInfo);
+
+    // prevent any corner case where buffers could get stuck in queue
+    signalProcessBuffers();
+}
+
+void MediaFilter::onAllocateComponent(const sp<AMessage> &msg) {
+    CHECK_EQ(mState, UNINITIALIZED);
+
+    CHECK(msg->findString("componentName", &mComponentName));
+    const char* name = mComponentName.c_str();
+    if (!strcasecmp(name, "android.filter.zerofilter")) {
+        mFilter = new ZeroFilter;
+    } else if (!strcasecmp(name, "android.filter.saturation")) {
+        mFilter = new SaturationFilter;
+    } else if (!strcasecmp(name, "android.filter.intrinsicblur")) {
+        mFilter = new IntrinsicBlurFilter;
+    } else if (!strcasecmp(name, "android.filter.RenderScript")) {
+        mFilter = new RSFilter;
+    } else {
+        ALOGE("Unrecognized filter name: %s", name);
+        signalError(NAME_NOT_FOUND);
+        return;
+    }
+
+    sp<AMessage> notify = mNotify->dup();
+    notify->setInt32("what", kWhatComponentAllocated);
+    // HACK - need "OMX.google" to use MediaCodec's software renderer
+    notify->setString("componentName", "OMX.google.MediaFilter");
+    notify->post();
+    mState = INITIALIZED;
+    ALOGV("Handled kWhatAllocateComponent.");
+}
+
+void MediaFilter::onConfigureComponent(const sp<AMessage> &msg) {
+    // TODO: generalize to allow audio filters as well as video
+
+    CHECK_EQ(mState, INITIALIZED);
+
+    // get params - at least mime, width & height
+    AString mime;
+    CHECK(msg->findString("mime", &mime));
+    if (strcasecmp(mime.c_str(), MEDIA_MIMETYPE_VIDEO_RAW)) {
+        ALOGE("Bad mime: %s", mime.c_str());
+        signalError(BAD_VALUE);
+        return;
+    }
+
+    CHECK(msg->findInt32("width", &mWidth));
+    CHECK(msg->findInt32("height", &mHeight));
+    if (!msg->findInt32("stride", &mStride)) {
+        mStride = mWidth;
+    }
+    if (!msg->findInt32("slice-height", &mSliceHeight)) {
+        mSliceHeight = mHeight;
+    }
+
+    mMaxInputSize = mWidth * mHeight * 4;   // room for ARGB8888
+    int32_t maxInputSize;
+    if (msg->findInt32("max-input-size", &maxInputSize)
+            && (size_t)maxInputSize > mMaxInputSize) {
+        mMaxInputSize = maxInputSize;
+    }
+
+    if (!msg->findInt32("color-format", &mColorFormatIn)) {
+        // default to OMX_COLOR_Format32bitARGB8888
+        mColorFormatIn = OMX_COLOR_Format32bitARGB8888;
+        msg->setInt32("color-format", mColorFormatIn);
+    }
+    mColorFormatOut = mColorFormatIn;
+
+    mMaxOutputSize = mWidth * mHeight * 4;  // room for ARGB8888
+
+    AString cacheDir;
+    if (!msg->findString("cacheDir", &cacheDir)) {
+        ALOGE("Failed to find cache directory in config message.");
+        signalError(NAME_NOT_FOUND);
+        return;
+    }
+
+    status_t err;
+    err = mFilter->configure(msg);
+    if (err != (status_t)OK) {
+        ALOGE("Failed to configure filter component, err %d", err);
+        signalError(err);
+        return;
+    }
+
+    mInputFormat = new AMessage();
+    mInputFormat->setString("mime", mime.c_str());
+    mInputFormat->setInt32("stride", mStride);
+    mInputFormat->setInt32("slice-height", mSliceHeight);
+    mInputFormat->setInt32("color-format", mColorFormatIn);
+    mInputFormat->setRect("crop", 0, 0, mStride, mSliceHeight);
+    mInputFormat->setInt32("width", mWidth);
+    mInputFormat->setInt32("height", mHeight);
+
+    mOutputFormat = new AMessage();
+    mOutputFormat->setString("mime", mime.c_str());
+    mOutputFormat->setInt32("stride", mStride);
+    mOutputFormat->setInt32("slice-height", mSliceHeight);
+    mOutputFormat->setInt32("color-format", mColorFormatOut);
+    mOutputFormat->setRect("crop", 0, 0, mStride, mSliceHeight);
+    mOutputFormat->setInt32("width", mWidth);
+    mOutputFormat->setInt32("height", mHeight);
+
+    sp<AMessage> notify = mNotify->dup();
+    notify->setInt32("what", kWhatComponentConfigured);
+    notify->setString("componentName", "MediaFilter");
+    notify->setMessage("input-format", mInputFormat);
+    notify->setMessage("output-format", mOutputFormat);
+    notify->post();
+    mState = CONFIGURED;
+    ALOGV("Handled kWhatConfigureComponent.");
+
+    sendFormatChange();
+}
+
+void MediaFilter::onStart() {
+    CHECK_EQ(mState, CONFIGURED);
+
+    allocateBuffersOnPort(kPortIndexInput);
+
+    allocateBuffersOnPort(kPortIndexOutput);
+
+    status_t err = mFilter->start();
+    if (err != (status_t)OK) {
+        ALOGE("Failed to start filter component, err %d", err);
+        signalError(err);
+        return;
+    }
+
+    mPortEOS[kPortIndexInput] = false;
+    mPortEOS[kPortIndexOutput] = false;
+    mInputEOSResult = OK;
+    mState = STARTED;
+
+    requestFillEmptyInput();
+    ALOGV("Handled kWhatStart.");
+}
+
+void MediaFilter::onInputBufferFilled(const sp<AMessage> &msg) {
+    IOMX::buffer_id bufferID;
+    CHECK(msg->findInt32("buffer-id", (int32_t*)&bufferID));
+    BufferInfo *info = findBufferByID(kPortIndexInput, bufferID);
+
+    if (mState != STARTED) {
+        // we're not running, so we'll just keep that buffer...
+        info->mStatus = BufferInfo::OWNED_BY_US;
+        return;
+    }
+
+    if (info->mGeneration != mGeneration) {
+        ALOGV("Caught a stale input buffer [ID %d]", bufferID);
+        // buffer is stale (taken before a flush/shutdown) - repost it
+        CHECK_EQ(info->mStatus, BufferInfo::OWNED_BY_US);
+        postFillThisBuffer(info);
+        return;
+    }
+
+    CHECK_EQ(info->mStatus, BufferInfo::OWNED_BY_UPSTREAM);
+    info->mStatus = BufferInfo::OWNED_BY_US;
+
+    sp<ABuffer> buffer;
+    int32_t err = OK;
+    bool eos = false;
+
+    if (!msg->findBuffer("buffer", &buffer)) {
+        // these are unfilled buffers returned by client
+        CHECK(msg->findInt32("err", &err));
+
+        if (err == OK) {
+            // buffers with no errors are returned on MediaCodec.flush
+            ALOGV("saw unfilled buffer (MediaCodec.flush)");
+            postFillThisBuffer(info);
+            return;
+        } else {
+            ALOGV("saw error %d instead of an input buffer", err);
+            eos = true;
+        }
+
+        buffer.clear();
+    }
+
+    int32_t isCSD;
+    if (buffer != NULL && buffer->meta()->findInt32("csd", &isCSD)
+            && isCSD != 0) {
+        // ignore codec-specific data buffers
+        ALOGW("MediaFilter received a codec-specific data buffer");
+        postFillThisBuffer(info);
+        return;
+    }
+
+    int32_t tmp;
+    if (buffer != NULL && buffer->meta()->findInt32("eos", &tmp) && tmp) {
+        eos = true;
+        err = ERROR_END_OF_STREAM;
+    }
+
+    mAvailableInputBuffers.push_back(info);
+    processBuffers();
+
+    if (eos) {
+        mPortEOS[kPortIndexInput] = true;
+        mInputEOSResult = err;
+    }
+
+    ALOGV("Handled kWhatInputBufferFilled. [ID %u]", bufferID);
+}
+
+void MediaFilter::onOutputBufferDrained(const sp<AMessage> &msg) {
+    IOMX::buffer_id bufferID;
+    CHECK(msg->findInt32("buffer-id", (int32_t*)&bufferID));
+    BufferInfo *info = findBufferByID(kPortIndexOutput, bufferID);
+
+    if (mState != STARTED) {
+        // we're not running, so we'll just keep that buffer...
+        info->mStatus = BufferInfo::OWNED_BY_US;
+        return;
+    }
+
+    if (info->mGeneration != mGeneration) {
+        ALOGV("Caught a stale output buffer [ID %d]", bufferID);
+        // buffer is stale (taken before a flush/shutdown) - keep it
+        CHECK_EQ(info->mStatus, BufferInfo::OWNED_BY_US);
+        return;
+    }
+
+    CHECK_EQ(info->mStatus, BufferInfo::OWNED_BY_UPSTREAM);
+    info->mStatus = BufferInfo::OWNED_BY_US;
+
+    mAvailableOutputBuffers.push_back(info);
+
+    processBuffers();
+
+    ALOGV("Handled kWhatOutputBufferDrained. [ID %u]",
+            bufferID);
+}
+
+void MediaFilter::onShutdown(const sp<AMessage> &msg) {
+    mGeneration++;
+
+    if (mState != UNINITIALIZED) {
+        mFilter->reset();
+    }
+
+    int32_t keepComponentAllocated;
+    CHECK(msg->findInt32("keepComponentAllocated", &keepComponentAllocated));
+    if (!keepComponentAllocated || mState == UNINITIALIZED) {
+        mState = UNINITIALIZED;
+    } else {
+        mState = INITIALIZED;
+    }
+
+    sp<AMessage> notify = mNotify->dup();
+    notify->setInt32("what", CodecBase::kWhatShutdownCompleted);
+    notify->post();
+}
+
+void MediaFilter::onFlush() {
+    mGeneration++;
+
+    mAvailableInputBuffers.clear();
+    for (size_t i = 0; i < mBuffers[kPortIndexInput].size(); ++i) {
+        BufferInfo *info = &mBuffers[kPortIndexInput].editItemAt(i);
+        info->mStatus = BufferInfo::OWNED_BY_US;
+    }
+    mAvailableOutputBuffers.clear();
+    for (size_t i = 0; i < mBuffers[kPortIndexOutput].size(); ++i) {
+        BufferInfo *info = &mBuffers[kPortIndexOutput].editItemAt(i);
+        info->mStatus = BufferInfo::OWNED_BY_US;
+        mAvailableOutputBuffers.push_back(info);
+    }
+
+    mPortEOS[kPortIndexInput] = false;
+    mPortEOS[kPortIndexOutput] = false;
+    mInputEOSResult = OK;
+
+    sp<AMessage> notify = mNotify->dup();
+    notify->setInt32("what", CodecBase::kWhatFlushCompleted);
+    notify->post();
+    ALOGV("Posted kWhatFlushCompleted");
+
+    // MediaCodec returns all input buffers after flush, so in
+    // onInputBufferFilled we call postFillThisBuffer on them
+}
+
+void MediaFilter::onSetParameters(const sp<AMessage> &msg) {
+    CHECK(mState != STARTED);
+
+    status_t err = mFilter->setParameters(msg);
+    if (err != (status_t)OK) {
+        ALOGE("setParameters returned err %d", err);
+    }
+}
+
+void MediaFilter::onCreateInputSurface() {
+    CHECK(mState == CONFIGURED);
+
+    mGraphicBufferListener = new GraphicBufferListener;
+
+    sp<AMessage> notify = new AMessage();
+    notify->setTarget(id());
+    status_t err = mGraphicBufferListener->init(
+            notify, mStride, mSliceHeight, kBufferCountActual);
+
+    if (err != OK) {
+        ALOGE("Failed to init mGraphicBufferListener: %d", err);
+        signalError(err);
+        return;
+    }
+
+    sp<AMessage> reply = mNotify->dup();
+    reply->setInt32("what", CodecBase::kWhatInputSurfaceCreated);
+    reply->setObject(
+            "input-surface",
+            new BufferProducerWrapper(
+                    mGraphicBufferListener->getIGraphicBufferProducer()));
+    reply->post();
+}
+
+void MediaFilter::onInputFrameAvailable() {
+    BufferQueue::BufferItem item = mGraphicBufferListener->getBufferItem();
+    sp<GraphicBuffer> buf = mGraphicBufferListener->getBuffer(item);
+
+    // get pointer to graphic buffer
+    void* bufPtr;
+    buf->lock(GraphicBuffer::USAGE_SW_READ_OFTEN, &bufPtr);
+
+    // HACK - there is no OMX_COLOR_FORMATTYPE value for RGBA, so the format
+    // conversion is hardcoded until we add this.
+    // TODO: check input format and convert only if necessary
+    // copy RGBA graphic buffer into temporary ARGB input buffer
+    BufferInfo *inputInfo = new BufferInfo;
+    inputInfo->mData = new ABuffer(buf->getWidth() * buf->getHeight() * 4);
+    ALOGV("Copying surface data into temp buffer.");
+    convertRGBAToARGB(
+            (uint8_t*)bufPtr, buf->getWidth(), buf->getHeight(),
+            buf->getStride(), inputInfo->mData->data());
+    inputInfo->mBufferID = item.mBuf;
+    inputInfo->mGeneration = mGeneration;
+    inputInfo->mOutputFlags = 0;
+    inputInfo->mStatus = BufferInfo::OWNED_BY_US;
+    inputInfo->mData->meta()->setInt64("timeUs", item.mTimestamp / 1000);
+
+    mAvailableInputBuffers.push_back(inputInfo);
+
+    mGraphicBufferListener->releaseBuffer(item);
+
+    signalProcessBuffers();
+}
+
+void MediaFilter::onSignalEndOfInputStream() {
+    // if using input surface, need to send an EOS output buffer
+    if (mGraphicBufferListener != NULL) {
+        Vector<BufferInfo> *outputBufs = &mBuffers[kPortIndexOutput];
+        BufferInfo* eosBuf;
+        bool foundBuf = false;
+        for (size_t i = 0; i < kBufferCountActual; i++) {
+            eosBuf = &outputBufs->editItemAt(i);
+            if (eosBuf->mStatus == BufferInfo::OWNED_BY_US) {
+                foundBuf = true;
+                break;
+            }
+        }
+
+        if (!foundBuf) {
+            ALOGE("onSignalEndOfInputStream failed to find an output buffer");
+            return;
+        }
+
+        eosBuf->mOutputFlags = OMX_BUFFERFLAG_EOS;
+        eosBuf->mGeneration = mGeneration;
+        eosBuf->mData->setRange(0, 0);
+        postDrainThisBuffer(eosBuf);
+        ALOGV("Posted EOS on output buffer %zu", eosBuf->mBufferID);
+    }
+
+    mPortEOS[kPortIndexOutput] = true;
+    sp<AMessage> notify = mNotify->dup();
+    notify->setInt32("what", CodecBase::kWhatSignaledInputEOS);
+    notify->post();
+
+    ALOGV("Output stream saw EOS.");
+}
+
+}   // namespace android
diff --git a/media/libstagefright/filters/RSFilter.cpp b/media/libstagefright/filters/RSFilter.cpp
new file mode 100644
index 0000000..b569945
--- /dev/null
+++ b/media/libstagefright/filters/RSFilter.cpp
@@ -0,0 +1,96 @@
+/*
+ * Copyright (C) 2014 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 "RSFilter"
+
+#include <utils/Log.h>
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include "RSFilter.h"
+
+namespace android {
+
+RSFilter::RSFilter() {
+
+}
+
+RSFilter::~RSFilter() {
+
+}
+
+status_t RSFilter::configure(const sp<AMessage> &msg) {
+    status_t err = SimpleFilter::configure(msg);
+    if (err != OK) {
+        return err;
+    }
+
+    if (!msg->findString("cacheDir", &mCacheDir)) {
+        ALOGE("Failed to find cache directory in config message.");
+        return NAME_NOT_FOUND;
+    }
+
+    sp<RenderScriptWrapper> wrapper;
+    if (!msg->findObject("rs-wrapper", (sp<RefBase>*)&wrapper)) {
+        ALOGE("Failed to find RenderScriptWrapper in config message.");
+        return NAME_NOT_FOUND;
+    }
+
+    mRS = wrapper->mContext;
+    mCallback = wrapper->mCallback;
+
+    return OK;
+}
+
+status_t RSFilter::start() {
+    // 32-bit elements for ARGB8888
+    RSC::sp<const RSC::Element> e = RSC::Element::U8_4(mRS);
+
+    RSC::Type::Builder tb(mRS, e);
+    tb.setX(mWidth);
+    tb.setY(mHeight);
+    RSC::sp<const RSC::Type> t = tb.create();
+
+    mAllocIn = RSC::Allocation::createTyped(mRS, t);
+    mAllocOut = RSC::Allocation::createTyped(mRS, t);
+
+    return OK;
+}
+
+void RSFilter::reset() {
+    mCallback.clear();
+    mAllocOut.clear();
+    mAllocIn.clear();
+    mRS.clear();
+}
+
+status_t RSFilter::setParameters(const sp<AMessage> &msg) {
+    return mCallback->handleSetParameters(msg);
+}
+
+status_t RSFilter::processBuffers(
+        const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) {
+    mAllocIn->copy1DRangeFrom(0, mWidth * mHeight, srcBuffer->data());
+    mCallback->processBuffers(mAllocIn.get(), mAllocOut.get());
+    mAllocOut->copy1DRangeTo(0, mWidth * mHeight, outBuffer->data());
+
+    return OK;
+}
+
+}   // namespace android
diff --git a/media/libstagefright/filters/RSFilter.h b/media/libstagefright/filters/RSFilter.h
new file mode 100644
index 0000000..c5b5074
--- /dev/null
+++ b/media/libstagefright/filters/RSFilter.h
@@ -0,0 +1,53 @@
+/*
+ * Copyright (C) 2014 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 RS_FILTER_H_
+#define RS_FILTER_H_
+
+#include <media/stagefright/RenderScriptWrapper.h>
+#include <RenderScript.h>
+
+#include "SimpleFilter.h"
+
+namespace android {
+
+struct AString;
+
+struct RSFilter : public SimpleFilter {
+public:
+    RSFilter();
+
+    virtual status_t configure(const sp<AMessage> &msg);
+    virtual status_t start();
+    virtual void reset();
+    virtual status_t setParameters(const sp<AMessage> &msg);
+    virtual status_t processBuffers(
+            const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer);
+
+protected:
+    virtual ~RSFilter();
+
+private:
+    AString mCacheDir;
+    sp<RenderScriptWrapper::RSFilterCallback> mCallback;
+    RSC::sp<RSC::RS> mRS;
+    RSC::sp<RSC::Allocation> mAllocIn;
+    RSC::sp<RSC::Allocation> mAllocOut;
+};
+
+}   // namespace android
+
+#endif  // RS_FILTER_H_
diff --git a/media/libstagefright/filters/SaturationFilter.cpp b/media/libstagefright/filters/SaturationFilter.cpp
new file mode 100644
index 0000000..ba5f75a
--- /dev/null
+++ b/media/libstagefright/filters/SaturationFilter.cpp
@@ -0,0 +1,99 @@
+/*
+ * Copyright (C) 2014 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 "SaturationFilter"
+
+#include <utils/Log.h>
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include "SaturationFilter.h"
+
+namespace android {
+
+status_t SaturationFilter::configure(const sp<AMessage> &msg) {
+    status_t err = SimpleFilter::configure(msg);
+    if (err != OK) {
+        return err;
+    }
+
+    if (!msg->findString("cacheDir", &mCacheDir)) {
+        ALOGE("Failed to find cache directory in config message.");
+        return NAME_NOT_FOUND;
+    }
+
+    return OK;
+}
+
+status_t SaturationFilter::start() {
+    // TODO: use a single RS context object for entire application
+    mRS = new RSC::RS();
+
+    if (!mRS->init(mCacheDir.c_str())) {
+        ALOGE("Failed to initialize RenderScript context.");
+        return NO_INIT;
+    }
+
+    // 32-bit elements for ARGB8888
+    RSC::sp<const RSC::Element> e = RSC::Element::U8_4(mRS);
+
+    RSC::Type::Builder tb(mRS, e);
+    tb.setX(mWidth);
+    tb.setY(mHeight);
+    RSC::sp<const RSC::Type> t = tb.create();
+
+    mAllocIn = RSC::Allocation::createTyped(mRS, t);
+    mAllocOut = RSC::Allocation::createTyped(mRS, t);
+
+    mScript = new ScriptC_saturationARGB(mRS);
+
+    mScript->set_gSaturation(mSaturation);
+
+    return OK;
+}
+
+void SaturationFilter::reset() {
+    mScript.clear();
+    mAllocOut.clear();
+    mAllocIn.clear();
+    mRS.clear();
+}
+
+status_t SaturationFilter::setParameters(const sp<AMessage> &msg) {
+    sp<AMessage> params;
+    CHECK(msg->findMessage("params", &params));
+
+    float saturation;
+    if (params->findFloat("saturation", &saturation)) {
+        mSaturation = saturation;
+    }
+
+    return OK;
+}
+
+status_t SaturationFilter::processBuffers(
+        const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) {
+    mAllocIn->copy1DRangeFrom(0, mWidth * mHeight, srcBuffer->data());
+    mScript->forEach_root(mAllocIn, mAllocOut);
+    mAllocOut->copy1DRangeTo(0, mWidth * mHeight, outBuffer->data());
+
+    return OK;
+}
+
+}   // namespace android
diff --git a/media/libstagefright/filters/SaturationFilter.h b/media/libstagefright/filters/SaturationFilter.h
new file mode 100644
index 0000000..0545021
--- /dev/null
+++ b/media/libstagefright/filters/SaturationFilter.h
@@ -0,0 +1,52 @@
+/*
+ * Copyright (C) 2014 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 SATURATION_FILTER_H_
+#define SATURATION_FILTER_H_
+
+#include <RenderScript.h>
+
+#include "ScriptC_saturationARGB.h"
+#include "SimpleFilter.h"
+
+namespace android {
+
+struct SaturationFilter : public SimpleFilter {
+public:
+    SaturationFilter() : mSaturation(1.f) {};
+
+    virtual status_t configure(const sp<AMessage> &msg);
+    virtual status_t start();
+    virtual void reset();
+    virtual status_t setParameters(const sp<AMessage> &msg);
+    virtual status_t processBuffers(
+            const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer);
+
+protected:
+    virtual ~SaturationFilter() {};
+
+private:
+    AString mCacheDir;
+    RSC::sp<RSC::RS> mRS;
+    RSC::sp<RSC::Allocation> mAllocIn;
+    RSC::sp<RSC::Allocation> mAllocOut;
+    RSC::sp<ScriptC_saturationARGB> mScript;
+    float mSaturation;
+};
+
+}   // namespace android
+
+#endif  // SATURATION_FILTER_H_
diff --git a/media/libstagefright/filters/SimpleFilter.cpp b/media/libstagefright/filters/SimpleFilter.cpp
new file mode 100644
index 0000000..6c1ca2c
--- /dev/null
+++ b/media/libstagefright/filters/SimpleFilter.cpp
@@ -0,0 +1,39 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include "SimpleFilter.h"
+
+namespace android {
+
+status_t SimpleFilter::configure(const sp<AMessage> &msg) {
+    CHECK(msg->findInt32("width", &mWidth));
+    CHECK(msg->findInt32("height", &mHeight));
+    if (!msg->findInt32("stride", &mStride)) {
+        mStride = mWidth;
+    }
+    if (!msg->findInt32("slice-height", &mSliceHeight)) {
+        mSliceHeight = mHeight;
+    }
+    CHECK(msg->findInt32("color-format", &mColorFormatIn));
+    mColorFormatOut = mColorFormatIn;
+
+    return OK;
+}
+
+}   // namespace android
diff --git a/media/libstagefright/filters/SimpleFilter.h b/media/libstagefright/filters/SimpleFilter.h
new file mode 100644
index 0000000..4cd37ef
--- /dev/null
+++ b/media/libstagefright/filters/SimpleFilter.h
@@ -0,0 +1,52 @@
+/*
+ * Copyright (C) 2014 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 SIMPLE_FILTER_H_
+#define SIMPLE_FILTER_H_
+
+#include <stdint.h>
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+
+struct ABuffer;
+struct AMessage;
+
+namespace android {
+
+struct SimpleFilter : public RefBase {
+public:
+    SimpleFilter() : mWidth(0), mHeight(0), mStride(0), mSliceHeight(0),
+            mColorFormatIn(0), mColorFormatOut(0) {};
+
+    virtual status_t configure(const sp<AMessage> &msg);
+
+    virtual status_t start() = 0;
+    virtual void reset() = 0;
+    virtual status_t setParameters(const sp<AMessage> &msg) = 0;
+    virtual status_t processBuffers(
+            const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) = 0;
+
+protected:
+    int32_t mWidth, mHeight;
+    int32_t mStride, mSliceHeight;
+    int32_t mColorFormatIn, mColorFormatOut;
+
+    virtual ~SimpleFilter() {};
+};
+
+}   // namespace android
+
+#endif  // SIMPLE_FILTER_H_
diff --git a/media/libstagefright/filters/ZeroFilter.cpp b/media/libstagefright/filters/ZeroFilter.cpp
new file mode 100644
index 0000000..3f1243c
--- /dev/null
+++ b/media/libstagefright/filters/ZeroFilter.cpp
@@ -0,0 +1,57 @@
+/*
+ * Copyright (C) 2014 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 "ZeroFilter"
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include "ZeroFilter.h"
+
+namespace android {
+
+status_t ZeroFilter::setParameters(const sp<AMessage> &msg) {
+    sp<AMessage> params;
+    CHECK(msg->findMessage("params", &params));
+
+    int32_t invert;
+    if (params->findInt32("invert", &invert)) {
+        mInvertData = (invert != 0);
+    }
+
+    return OK;
+}
+
+status_t ZeroFilter::processBuffers(
+        const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) {
+    // assuming identical input & output buffers, since we're a copy filter
+    if (mInvertData) {
+        uint32_t* src = (uint32_t*)srcBuffer->data();
+        uint32_t* dest = (uint32_t*)outBuffer->data();
+        for (size_t i = 0; i < srcBuffer->size() / 4; ++i) {
+            *(dest++) = *(src++) ^ 0xFFFFFFFF;
+        }
+    } else {
+        memcpy(outBuffer->data(), srcBuffer->data(), srcBuffer->size());
+    }
+    outBuffer->setRange(0, srcBuffer->size());
+
+    return OK;
+}
+
+}   // namespace android
diff --git a/media/libstagefright/filters/ZeroFilter.h b/media/libstagefright/filters/ZeroFilter.h
new file mode 100644
index 0000000..bd34dfb
--- /dev/null
+++ b/media/libstagefright/filters/ZeroFilter.h
@@ -0,0 +1,43 @@
+/*
+ * Copyright (C) 2014 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 ZERO_FILTER_H_
+#define ZERO_FILTER_H_
+
+#include "SimpleFilter.h"
+
+namespace android {
+
+struct ZeroFilter : public SimpleFilter {
+public:
+    ZeroFilter() : mInvertData(false) {};
+
+    virtual status_t start() { return OK; };
+    virtual void reset() {};
+    virtual status_t setParameters(const sp<AMessage> &msg);
+    virtual status_t processBuffers(
+            const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer);
+
+protected:
+    virtual ~ZeroFilter() {};
+
+private:
+    bool mInvertData;
+};
+
+}   // namespace android
+
+#endif  // ZERO_FILTER_H_
diff --git a/media/libstagefright/filters/saturation.rs b/media/libstagefright/filters/saturation.rs
new file mode 100644
index 0000000..2c867ac
--- /dev/null
+++ b/media/libstagefright/filters/saturation.rs
@@ -0,0 +1,40 @@
+// Sample script for RGB888 support (compare to saturationARGB.rs)
+/*
+ * Copyright (C) 2014 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.
+ */
+
+#pragma version(1)
+#pragma rs java_package_name(com.android.rs.cppbasic)
+#pragma rs_fp_relaxed
+
+const static float3 gMonoMult = {0.299f, 0.587f, 0.114f};
+
+// global variables (parameters accessible to application code)
+float gSaturation = 1.0f;
+
+void root(const uchar3 *v_in, uchar3 *v_out) {
+    // scale 0-255 uchar to 0-1.0 float
+    float3 in = {v_in->r * 0.003921569f, v_in->g * 0.003921569f,
+            v_in->b * 0.003921569f};
+
+    // apply saturation filter
+    float3 result = dot(in, gMonoMult);
+    result = mix(result, in, gSaturation);
+
+    // convert to uchar, copied from rsPackColorTo8888
+    v_out->x = (uchar)clamp((result.r * 255.f + 0.5f), 0.f, 255.f);
+    v_out->y = (uchar)clamp((result.g * 255.f + 0.5f), 0.f, 255.f);
+    v_out->z = (uchar)clamp((result.b * 255.f + 0.5f), 0.f, 255.f);
+}
diff --git a/media/libstagefright/filters/saturationARGB.rs b/media/libstagefright/filters/saturationARGB.rs
new file mode 100644
index 0000000..1de9dd8
--- /dev/null
+++ b/media/libstagefright/filters/saturationARGB.rs
@@ -0,0 +1,40 @@
+/*
+ * Copyright (C) 2014 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.
+ */
+
+#pragma version(1)
+#pragma rs java_package_name(com.android.rs.cppbasic)
+#pragma rs_fp_relaxed
+
+const static float3 gMonoMult = {0.299f, 0.587f, 0.114f};
+
+// global variables (parameters accessible to application code)
+float gSaturation = 1.0f;
+
+void root(const uchar4 *v_in, uchar4 *v_out) {
+    v_out->x = v_in->x; // don't modify A
+
+    // get RGB, scale 0-255 uchar to 0-1.0 float
+    float3 rgb = {v_in->y * 0.003921569f, v_in->z * 0.003921569f,
+            v_in->w * 0.003921569f};
+
+    // apply saturation filter
+    float3 result = dot(rgb, gMonoMult);
+    result = mix(result, rgb, gSaturation);
+
+    v_out->y = (uchar)clamp((result.r * 255.f + 0.5f), 0.f, 255.f);
+    v_out->z = (uchar)clamp((result.g * 255.f + 0.5f), 0.f, 255.f);
+    v_out->w = (uchar)clamp((result.b * 255.f + 0.5f), 0.f, 255.f);
+}
diff --git a/media/libstagefright/include/MPEG4Extractor.h b/media/libstagefright/include/MPEG4Extractor.h
index 1fe6fcf..8c16251 100644
--- a/media/libstagefright/include/MPEG4Extractor.h
+++ b/media/libstagefright/include/MPEG4Extractor.h
@@ -83,6 +83,8 @@
 
     Vector<SidxEntry> mSidxEntries;
     off64_t mMoofOffset;
+    bool mMoofFound;
+    bool mMdatFound;
 
     Vector<PsshInfo> mPssh;
 
diff --git a/media/libstagefright/include/avc_utils.h b/media/libstagefright/include/avc_utils.h
index c270bc1..dafa07e 100644
--- a/media/libstagefright/include/avc_utils.h
+++ b/media/libstagefright/include/avc_utils.h
@@ -36,6 +36,11 @@
     kAVCProfileCAVLC444Intra = 0x2c
 };
 
+struct NALPosition {
+    size_t nalOffset;
+    size_t nalSize;
+};
+
 // Optionally returns sample aspect ratio as well.
 void FindAVCDimensions(
         const sp<ABuffer> &seqParamSet,
diff --git a/media/libstagefright/mpeg2ts/ESQueue.cpp b/media/libstagefright/mpeg2ts/ESQueue.cpp
index a605595..88da275 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.cpp
+++ b/media/libstagefright/mpeg2ts/ESQueue.cpp
@@ -617,8 +617,6 @@
     // having to interpolate.
     // The final AAC frame may well extend into the next RangeInfo but
     // that's ok.
-    // TODO: the logic commented above is skipped because codec cannot take
-    // arbitrary sized input buffers;
     size_t offset = 0;
     while (offset < info.mLength) {
         if (offset + 7 > mBuffer->size()) {
@@ -683,12 +681,9 @@
         size_t headerSize __unused = protection_absent ? 7 : 9;
 
         offset += aac_frame_length;
-        // TODO: move back to concatenation when codec can support arbitrary input buffers.
-        // For now only queue a single buffer
-        break;
     }
 
-    int64_t timeUs = fetchTimestampAAC(offset);
+    int64_t timeUs = fetchTimestamp(offset);
 
     sp<ABuffer> accessUnit = new ABuffer(offset);
     memcpy(accessUnit->data(), mBuffer->data(), offset);
@@ -735,50 +730,6 @@
     return timeUs;
 }
 
-// TODO: avoid interpolating timestamps once codec supports arbitrary sized input buffers
-int64_t ElementaryStreamQueue::fetchTimestampAAC(size_t size) {
-    int64_t timeUs = -1;
-    bool first = true;
-
-    size_t samplesize = size;
-    while (size > 0) {
-        CHECK(!mRangeInfos.empty());
-
-        RangeInfo *info = &*mRangeInfos.begin();
-
-        if (first) {
-            timeUs = info->mTimestampUs;
-            first = false;
-        }
-
-        if (info->mLength > size) {
-            int32_t sampleRate;
-            CHECK(mFormat->findInt32(kKeySampleRate, &sampleRate));
-            info->mLength -= size;
-            size_t numSamples = 1024 * size / samplesize;
-            info->mTimestampUs += numSamples * 1000000ll / sampleRate;
-            size = 0;
-        } else {
-            size -= info->mLength;
-
-            mRangeInfos.erase(mRangeInfos.begin());
-            info = NULL;
-        }
-
-    }
-
-    if (timeUs == 0ll) {
-        ALOGV("Returning 0 timestamp");
-    }
-
-    return timeUs;
-}
-
-struct NALPosition {
-    size_t nalOffset;
-    size_t nalSize;
-};
-
 sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitH264() {
     const uint8_t *data = mBuffer->data();
 
@@ -786,6 +737,7 @@
     Vector<NALPosition> nals;
 
     size_t totalSize = 0;
+    size_t seiCount = 0;
 
     status_t err;
     const uint8_t *nalStart;
@@ -815,6 +767,9 @@
             // next frame.
 
             flush = true;
+        } else if (nalType == 6 && nalSize > 0) {
+            // found non-zero sized SEI
+            ++seiCount;
         }
 
         if (flush) {
@@ -823,21 +778,29 @@
 
             size_t auSize = 4 * nals.size() + totalSize;
             sp<ABuffer> accessUnit = new ABuffer(auSize);
+            sp<ABuffer> sei;
+
+            if (seiCount > 0) {
+                sei = new ABuffer(seiCount * sizeof(NALPosition));
+                accessUnit->meta()->setBuffer("sei", sei);
+            }
 
 #if !LOG_NDEBUG
             AString out;
 #endif
 
             size_t dstOffset = 0;
+            size_t seiIndex = 0;
             for (size_t i = 0; i < nals.size(); ++i) {
                 const NALPosition &pos = nals.itemAt(i);
 
                 unsigned nalType = mBuffer->data()[pos.nalOffset] & 0x1f;
 
-                if (nalType == 6) {
-                    sp<ABuffer> sei = new ABuffer(pos.nalSize);
-                    memcpy(sei->data(), mBuffer->data() + pos.nalOffset, pos.nalSize);
-                    accessUnit->meta()->setBuffer("sei", sei);
+                if (nalType == 6 && pos.nalSize > 0) {
+                    CHECK_LT(seiIndex, sei->size() / sizeof(NALPosition));
+                    NALPosition &seiPos = ((NALPosition *)sei->data())[seiIndex++];
+                    seiPos.nalOffset = dstOffset + 4;
+                    seiPos.nalSize = pos.nalSize;
                 }
 
 #if !LOG_NDEBUG
diff --git a/media/libstagefright/mpeg2ts/ESQueue.h b/media/libstagefright/mpeg2ts/ESQueue.h
index eb4b1c9..45b4624 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.h
+++ b/media/libstagefright/mpeg2ts/ESQueue.h
@@ -77,7 +77,6 @@
     // consume a logical (compressed) access unit of size "size",
     // returns its timestamp in us (or -1 if no time information).
     int64_t fetchTimestamp(size_t size);
-    int64_t fetchTimestampAAC(size_t size);
 
     DISALLOW_EVIL_CONSTRUCTORS(ElementaryStreamQueue);
 };
diff --git a/media/libstagefright/omx/Android.mk b/media/libstagefright/omx/Android.mk
index aaa8334..be8cf46 100644
--- a/media/libstagefright/omx/Android.mk
+++ b/media/libstagefright/omx/Android.mk
@@ -6,6 +6,7 @@
 endif
 
 LOCAL_SRC_FILES:=                     \
+        FrameDropper.cpp              \
         GraphicBufferSource.cpp       \
         OMX.cpp                       \
         OMXMaster.cpp                 \
diff --git a/media/libstagefright/omx/FrameDropper.cpp b/media/libstagefright/omx/FrameDropper.cpp
new file mode 100644
index 0000000..9fba0b7
--- /dev/null
+++ b/media/libstagefright/omx/FrameDropper.cpp
@@ -0,0 +1,70 @@
+/*
+ * Copyright (C) 2015 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 "FrameDropper"
+#include <utils/Log.h>
+
+#include "FrameDropper.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+
+namespace android {
+
+static const int64_t kMaxJitterUs = 2000;
+
+FrameDropper::FrameDropper()
+    : mDesiredMinTimeUs(-1),
+      mMinIntervalUs(0) {
+}
+
+FrameDropper::~FrameDropper() {
+}
+
+status_t FrameDropper::setMaxFrameRate(float maxFrameRate) {
+    if (maxFrameRate <= 0) {
+        ALOGE("framerate should be positive but got %f.", maxFrameRate);
+        return BAD_VALUE;
+    }
+    mMinIntervalUs = (int64_t) (1000000.0f / maxFrameRate);
+    return OK;
+}
+
+bool FrameDropper::shouldDrop(int64_t timeUs) {
+    if (mMinIntervalUs <= 0) {
+        return false;
+    }
+
+    if (mDesiredMinTimeUs < 0) {
+        mDesiredMinTimeUs = timeUs + mMinIntervalUs;
+        ALOGV("first frame %lld, next desired frame %lld", timeUs, mDesiredMinTimeUs);
+        return false;
+    }
+
+    if (timeUs < (mDesiredMinTimeUs - kMaxJitterUs)) {
+        ALOGV("drop frame %lld, desired frame %lld, diff %lld",
+                timeUs, mDesiredMinTimeUs, mDesiredMinTimeUs - timeUs);
+        return true;
+    }
+
+    int64_t n = (timeUs - mDesiredMinTimeUs + kMaxJitterUs) / mMinIntervalUs;
+    mDesiredMinTimeUs += (n + 1) * mMinIntervalUs;
+    ALOGV("keep frame %lld, next desired frame %lld, diff %lld",
+            timeUs, mDesiredMinTimeUs, mDesiredMinTimeUs - timeUs);
+    return false;
+}
+
+}  // namespace android
diff --git a/media/libstagefright/omx/FrameDropper.h b/media/libstagefright/omx/FrameDropper.h
new file mode 100644
index 0000000..c5a6d4b
--- /dev/null
+++ b/media/libstagefright/omx/FrameDropper.h
@@ -0,0 +1,50 @@
+/*
+ * Copyright (C) 2015 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 FRAME_DROPPER_H_
+
+#define FRAME_DROPPER_H_
+
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+
+#include <media/stagefright/foundation/ABase.h>
+
+namespace android {
+
+struct FrameDropper : public RefBase {
+    // No frames will be dropped until a valid max frame rate is set.
+    FrameDropper();
+
+    // maxFrameRate required to be positive.
+    status_t setMaxFrameRate(float maxFrameRate);
+
+    // Returns false if max frame rate has not been set via setMaxFrameRate.
+    bool shouldDrop(int64_t timeUs);
+
+protected:
+    virtual ~FrameDropper();
+
+private:
+    int64_t mDesiredMinTimeUs;
+    int64_t mMinIntervalUs;
+
+    DISALLOW_EVIL_CONSTRUCTORS(FrameDropper);
+};
+
+}  // namespace android
+
+#endif  // FRAME_DROPPER_H_
diff --git a/media/libstagefright/omx/GraphicBufferSource.cpp b/media/libstagefright/omx/GraphicBufferSource.cpp
index 44c7edc..7afe699 100644
--- a/media/libstagefright/omx/GraphicBufferSource.cpp
+++ b/media/libstagefright/omx/GraphicBufferSource.cpp
@@ -30,6 +30,7 @@
 #include <ui/GraphicBuffer.h>
 
 #include <inttypes.h>
+#include "FrameDropper.h"
 
 namespace android {
 
@@ -53,9 +54,9 @@
     mRepeatAfterUs(-1ll),
     mRepeatLastFrameGeneration(0),
     mRepeatLastFrameTimestamp(-1ll),
-    mLatestSubmittedBufferId(-1),
-    mLatestSubmittedBufferFrameNum(0),
-    mLatestSubmittedBufferUseCount(0),
+    mLatestBufferId(-1),
+    mLatestBufferFrameNum(0),
+    mLatestBufferUseCount(0),
     mRepeatBufferDeferred(false),
     mTimePerCaptureUs(-1ll),
     mTimePerFrameUs(-1ll),
@@ -152,7 +153,7 @@
         mLooper->registerHandler(mReflector);
         mLooper->start();
 
-        if (mLatestSubmittedBufferId >= 0) {
+        if (mLatestBufferId >= 0) {
             sp<AMessage> msg =
                 new AMessage(kWhatRepeatLastFrame, mReflector->id());
 
@@ -287,8 +288,8 @@
         ALOGV("cbi %d matches bq slot %d, handle=%p",
                 cbi, id, mBufferSlot[id]->handle);
 
-        if (id == mLatestSubmittedBufferId) {
-            CHECK_GT(mLatestSubmittedBufferUseCount--, 0);
+        if (id == mLatestBufferId) {
+            CHECK_GT(mLatestBufferUseCount--, 0);
         } else {
             mConsumer->releaseBuffer(id, codecBuffer.mFrameNumber,
                     EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, Fence::NO_FENCE);
@@ -313,11 +314,11 @@
         ALOGV("buffer freed, EOS pending");
         submitEndOfInputStream_l();
     } else if (mRepeatBufferDeferred) {
-        bool success = repeatLatestSubmittedBuffer_l();
+        bool success = repeatLatestBuffer_l();
         if (success) {
-            ALOGV("deferred repeatLatestSubmittedBuffer_l SUCCESS");
+            ALOGV("deferred repeatLatestBuffer_l SUCCESS");
         } else {
-            ALOGV("deferred repeatLatestSubmittedBuffer_l FAILURE");
+            ALOGV("deferred repeatLatestBuffer_l FAILURE");
         }
         mRepeatBufferDeferred = false;
     }
@@ -382,12 +383,12 @@
     mSuspended = false;
 
     if (mExecuting && mNumFramesAvailable == 0 && mRepeatBufferDeferred) {
-        if (repeatLatestSubmittedBuffer_l()) {
-            ALOGV("suspend/deferred repeatLatestSubmittedBuffer_l SUCCESS");
+        if (repeatLatestBuffer_l()) {
+            ALOGV("suspend/deferred repeatLatestBuffer_l SUCCESS");
 
             mRepeatBufferDeferred = false;
         } else {
-            ALOGV("suspend/deferred repeatLatestSubmittedBuffer_l FAILURE");
+            ALOGV("suspend/deferred repeatLatestBuffer_l FAILURE");
         }
     }
 }
@@ -441,12 +442,22 @@
 
     // only submit sample if start time is unspecified, or sample
     // is queued after the specified start time
+    bool dropped = false;
     if (mSkipFramesBeforeNs < 0ll || item.mTimestamp >= mSkipFramesBeforeNs) {
         // if start time is set, offset time stamp by start time
         if (mSkipFramesBeforeNs > 0) {
             item.mTimestamp -= mSkipFramesBeforeNs;
         }
-        err = submitBuffer_l(item, cbi);
+
+        int64_t timeUs = item.mTimestamp / 1000;
+        if (mFrameDropper != NULL && mFrameDropper->shouldDrop(timeUs)) {
+            ALOGV("skipping frame (%lld) to meet max framerate", static_cast<long long>(timeUs));
+            // set err to OK so that the skipped frame can still be saved as the lastest frame
+            err = OK;
+            dropped = true;
+        } else {
+            err = submitBuffer_l(item, cbi);
+        }
     }
 
     if (err != OK) {
@@ -455,46 +466,46 @@
                 EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, Fence::NO_FENCE);
     } else {
         ALOGV("buffer submitted (bq %d, cbi %d)", item.mBuf, cbi);
-        setLatestSubmittedBuffer_l(item);
+        setLatestBuffer_l(item, dropped);
     }
 
     return true;
 }
 
-bool GraphicBufferSource::repeatLatestSubmittedBuffer_l() {
+bool GraphicBufferSource::repeatLatestBuffer_l() {
     CHECK(mExecuting && mNumFramesAvailable == 0);
 
-    if (mLatestSubmittedBufferId < 0 || mSuspended) {
+    if (mLatestBufferId < 0 || mSuspended) {
         return false;
     }
-    if (mBufferSlot[mLatestSubmittedBufferId] == NULL) {
+    if (mBufferSlot[mLatestBufferId] == NULL) {
         // This can happen if the remote side disconnects, causing
         // onBuffersReleased() to NULL out our copy of the slots.  The
         // buffer is gone, so we have nothing to show.
         //
         // To be on the safe side we try to release the buffer.
-        ALOGD("repeatLatestSubmittedBuffer_l: slot was NULL");
+        ALOGD("repeatLatestBuffer_l: slot was NULL");
         mConsumer->releaseBuffer(
-                mLatestSubmittedBufferId,
-                mLatestSubmittedBufferFrameNum,
+                mLatestBufferId,
+                mLatestBufferFrameNum,
                 EGL_NO_DISPLAY,
                 EGL_NO_SYNC_KHR,
                 Fence::NO_FENCE);
-        mLatestSubmittedBufferId = -1;
-        mLatestSubmittedBufferFrameNum = 0;
+        mLatestBufferId = -1;
+        mLatestBufferFrameNum = 0;
         return false;
     }
 
     int cbi = findAvailableCodecBuffer_l();
     if (cbi < 0) {
         // No buffers available, bail.
-        ALOGV("repeatLatestSubmittedBuffer_l: no codec buffers.");
+        ALOGV("repeatLatestBuffer_l: no codec buffers.");
         return false;
     }
 
     BufferQueue::BufferItem item;
-    item.mBuf = mLatestSubmittedBufferId;
-    item.mFrameNumber = mLatestSubmittedBufferFrameNum;
+    item.mBuf = mLatestBufferId;
+    item.mFrameNumber = mLatestBufferFrameNum;
     item.mTimestamp = mRepeatLastFrameTimestamp;
 
     status_t err = submitBuffer_l(item, cbi);
@@ -503,7 +514,7 @@
         return false;
     }
 
-    ++mLatestSubmittedBufferUseCount;
+    ++mLatestBufferUseCount;
 
     /* repeat last frame up to kRepeatLastFrameCount times.
      * in case of static scene, a single repeat might not get rid of encoder
@@ -522,26 +533,26 @@
     return true;
 }
 
-void GraphicBufferSource::setLatestSubmittedBuffer_l(
-        const BufferQueue::BufferItem &item) {
-    ALOGV("setLatestSubmittedBuffer_l");
+void GraphicBufferSource::setLatestBuffer_l(
+        const BufferQueue::BufferItem &item, bool dropped) {
+    ALOGV("setLatestBuffer_l");
 
-    if (mLatestSubmittedBufferId >= 0) {
-        if (mLatestSubmittedBufferUseCount == 0) {
+    if (mLatestBufferId >= 0) {
+        if (mLatestBufferUseCount == 0) {
             mConsumer->releaseBuffer(
-                    mLatestSubmittedBufferId,
-                    mLatestSubmittedBufferFrameNum,
+                    mLatestBufferId,
+                    mLatestBufferFrameNum,
                     EGL_NO_DISPLAY,
                     EGL_NO_SYNC_KHR,
                     Fence::NO_FENCE);
         }
     }
 
-    mLatestSubmittedBufferId = item.mBuf;
-    mLatestSubmittedBufferFrameNum = item.mFrameNumber;
+    mLatestBufferId = item.mBuf;
+    mLatestBufferFrameNum = item.mFrameNumber;
     mRepeatLastFrameTimestamp = item.mTimestamp + mRepeatAfterUs * 1000;
 
-    mLatestSubmittedBufferUseCount = 1;
+    mLatestBufferUseCount = dropped ? 0 : 1;
     mRepeatBufferDeferred = false;
     mRepeatLastFrameCount = kRepeatLastFrameCount;
 
@@ -841,6 +852,23 @@
     return OK;
 }
 
+status_t GraphicBufferSource::setMaxFps(float maxFps) {
+    Mutex::Autolock autoLock(mMutex);
+
+    if (mExecuting) {
+        return INVALID_OPERATION;
+    }
+
+    mFrameDropper = new FrameDropper();
+    status_t err = mFrameDropper->setMaxFrameRate(maxFps);
+    if (err != OK) {
+        mFrameDropper.clear();
+        return err;
+    }
+
+    return OK;
+}
+
 void GraphicBufferSource::setSkipFramesBeforeUs(int64_t skipFramesBeforeUs) {
     Mutex::Autolock autoLock(mMutex);
 
@@ -879,12 +907,12 @@
                 break;
             }
 
-            bool success = repeatLatestSubmittedBuffer_l();
+            bool success = repeatLatestBuffer_l();
 
             if (success) {
-                ALOGV("repeatLatestSubmittedBuffer_l SUCCESS");
+                ALOGV("repeatLatestBuffer_l SUCCESS");
             } else {
-                ALOGV("repeatLatestSubmittedBuffer_l FAILURE");
+                ALOGV("repeatLatestBuffer_l FAILURE");
                 mRepeatBufferDeferred = true;
             }
             break;
diff --git a/media/libstagefright/omx/GraphicBufferSource.h b/media/libstagefright/omx/GraphicBufferSource.h
index c8e3775..ce3881e 100644
--- a/media/libstagefright/omx/GraphicBufferSource.h
+++ b/media/libstagefright/omx/GraphicBufferSource.h
@@ -30,6 +30,8 @@
 
 namespace android {
 
+class FrameDropper;
+
 /*
  * This class is used to feed OMX codecs from a Surface via BufferQueue.
  *
@@ -119,6 +121,9 @@
     // of suspension on input.
     status_t setMaxTimestampGapUs(int64_t maxGapUs);
 
+    // When set, the max frame rate fed to the encoder will be capped at maxFps.
+    status_t setMaxFps(float maxFps);
+
     // Sets the time lapse (or slow motion) parameters.
     // data[0] is the time (us) between two frames for playback
     // data[1] is the time (us) between two frames for capture
@@ -193,8 +198,8 @@
     // doing anything if we don't have a codec buffer available.
     void submitEndOfInputStream_l();
 
-    void setLatestSubmittedBuffer_l(const BufferQueue::BufferItem &item);
-    bool repeatLatestSubmittedBuffer_l();
+    void setLatestBuffer_l(const BufferQueue::BufferItem &item, bool dropped);
+    bool repeatLatestBuffer_l();
     int64_t getTimestamp(const BufferQueue::BufferItem &item);
 
     // Lock, covers all member variables.
@@ -250,6 +255,8 @@
     int64_t mPrevModifiedTimeUs;
     int64_t mSkipFramesBeforeNs;
 
+    sp<FrameDropper> mFrameDropper;
+
     sp<ALooper> mLooper;
     sp<AHandlerReflector<GraphicBufferSource> > mReflector;
 
@@ -258,11 +265,11 @@
     int64_t mRepeatLastFrameTimestamp;
     int32_t mRepeatLastFrameCount;
 
-    int mLatestSubmittedBufferId;
-    uint64_t mLatestSubmittedBufferFrameNum;
-    int32_t mLatestSubmittedBufferUseCount;
+    int mLatestBufferId;
+    uint64_t mLatestBufferFrameNum;
+    int32_t mLatestBufferUseCount;
 
-    // The previously submitted buffer should've been repeated but
+    // The previous buffer should've been repeated but
     // no codec buffer was available at the time.
     bool mRepeatBufferDeferred;
 
diff --git a/media/libstagefright/omx/OMXNodeInstance.cpp b/media/libstagefright/omx/OMXNodeInstance.cpp
index ab7419f..4779d6a 100644
--- a/media/libstagefright/omx/OMXNodeInstance.cpp
+++ b/media/libstagefright/omx/OMXNodeInstance.cpp
@@ -1075,6 +1075,7 @@
         case IOMX::INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY:
             return "REPEAT_PREVIOUS_FRAME_DELAY";
         case IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP: return "MAX_TIMESTAMP_GAP";
+        case IOMX::INTERNAL_OPTION_MAX_FPS:           return "MAX_FPS";
         case IOMX::INTERNAL_OPTION_START_TIME:        return "START_TIME";
         case IOMX::INTERNAL_OPTION_TIME_LAPSE:        return "TIME_LAPSE";
         default:                                      return def;
@@ -1092,6 +1093,7 @@
         case IOMX::INTERNAL_OPTION_SUSPEND:
         case IOMX::INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY:
         case IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP:
+        case IOMX::INTERNAL_OPTION_MAX_FPS:
         case IOMX::INTERNAL_OPTION_START_TIME:
         case IOMX::INTERNAL_OPTION_TIME_LAPSE:
         {
@@ -1129,6 +1131,14 @@
                 int64_t maxGapUs = *(int64_t *)data;
                 CLOG_CONFIG(setInternalOption, "gapUs=%lld", (long long)maxGapUs);
                 return bufferSource->setMaxTimestampGapUs(maxGapUs);
+            } else if (type == IOMX::INTERNAL_OPTION_MAX_FPS) {
+                if (size != sizeof(float)) {
+                    return INVALID_OPERATION;
+                }
+
+                float maxFps = *(float *)data;
+                CLOG_CONFIG(setInternalOption, "maxFps=%f", maxFps);
+                return bufferSource->setMaxFps(maxFps);
             } else if (type == IOMX::INTERNAL_OPTION_START_TIME) {
                 if (size != sizeof(int64_t)) {
                     return INVALID_OPERATION;
diff --git a/media/libstagefright/omx/tests/Android.mk b/media/libstagefright/omx/tests/Android.mk
index 447b29e..9be637a 100644
--- a/media/libstagefright/omx/tests/Android.mk
+++ b/media/libstagefright/omx/tests/Android.mk
@@ -20,3 +20,21 @@
 LOCAL_32_BIT_ONLY := true
 
 include $(BUILD_EXECUTABLE)
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := FrameDropper_test
+
+LOCAL_MODULE_TAGS := tests
+
+LOCAL_SRC_FILES := \
+	FrameDropper_test.cpp \
+
+LOCAL_SHARED_LIBRARIES := \
+	libstagefright_omx \
+	libutils \
+
+LOCAL_C_INCLUDES := \
+	frameworks/av/media/libstagefright/omx \
+
+include $(BUILD_NATIVE_TEST)
diff --git a/media/libstagefright/omx/tests/FrameDropper_test.cpp b/media/libstagefright/omx/tests/FrameDropper_test.cpp
new file mode 100644
index 0000000..4ac72c4
--- /dev/null
+++ b/media/libstagefright/omx/tests/FrameDropper_test.cpp
@@ -0,0 +1,136 @@
+/*
+ * Copyright (C) 2015 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 "FrameDropper_test"
+#include <utils/Log.h>
+
+#include <gtest/gtest.h>
+
+#include "FrameDropper.h"
+#include <media/stagefright/foundation/ADebug.h>
+
+namespace android {
+
+struct TestFrame {
+  int64_t timeUs;
+  bool shouldDrop;
+};
+
+static const TestFrame testFrames20Fps[] = {
+    {1000000, false}, {1050000, false}, {1100000, false}, {1150000, false},
+    {1200000, false}, {1250000, false}, {1300000, false}, {1350000, false},
+    {1400000, false}, {1450000, false}, {1500000, false}, {1550000, false},
+    {1600000, false}, {1650000, false}, {1700000, false}, {1750000, false},
+    {1800000, false}, {1850000, false}, {1900000, false}, {1950000, false},
+};
+
+static const TestFrame testFrames30Fps[] = {
+    {1000000, false}, {1033333, false}, {1066667, false}, {1100000, false},
+    {1133333, false}, {1166667, false}, {1200000, false}, {1233333, false},
+    {1266667, false}, {1300000, false}, {1333333, false}, {1366667, false},
+    {1400000, false}, {1433333, false}, {1466667, false}, {1500000, false},
+    {1533333, false}, {1566667, false}, {1600000, false}, {1633333, false},
+};
+
+static const TestFrame testFrames40Fps[] = {
+    {1000000, false}, {1025000, true}, {1050000, false}, {1075000, false},
+    {1100000, false}, {1125000, true}, {1150000, false}, {1175000, false},
+    {1200000, false}, {1225000, true}, {1250000, false}, {1275000, false},
+    {1300000, false}, {1325000, true}, {1350000, false}, {1375000, false},
+    {1400000, false}, {1425000, true}, {1450000, false}, {1475000, false},
+};
+
+static const TestFrame testFrames60Fps[] = {
+    {1000000, false}, {1016667, true}, {1033333, false}, {1050000, true},
+    {1066667, false}, {1083333, true}, {1100000, false}, {1116667, true},
+    {1133333, false}, {1150000, true}, {1166667, false}, {1183333, true},
+    {1200000, false}, {1216667, true}, {1233333, false}, {1250000, true},
+    {1266667, false}, {1283333, true}, {1300000, false}, {1316667, true},
+};
+
+static const TestFrame testFramesVariableFps[] = {
+    // 40fps
+    {1000000, false}, {1025000, true}, {1050000, false}, {1075000, false},
+    {1100000, false}, {1125000, true}, {1150000, false}, {1175000, false},
+    {1200000, false}, {1225000, true}, {1250000, false}, {1275000, false},
+    {1300000, false}, {1325000, true}, {1350000, false}, {1375000, false},
+    {1400000, false}, {1425000, true}, {1450000, false}, {1475000, false},
+    // a timestamp jump plus switch to 20fps
+    {2000000, false}, {2050000, false}, {2100000, false}, {2150000, false},
+    {2200000, false}, {2250000, false}, {2300000, false}, {2350000, false},
+    {2400000, false}, {2450000, false}, {2500000, false}, {2550000, false},
+    {2600000, false}, {2650000, false}, {2700000, false}, {2750000, false},
+    {2800000, false}, {2850000, false}, {2900000, false}, {2950000, false},
+    // 60fps
+    {2966667, false}, {2983333, true}, {3000000, false}, {3016667, true},
+    {3033333, false}, {3050000, true}, {3066667, false}, {3083333, true},
+    {3100000, false}, {3116667, true}, {3133333, false}, {3150000, true},
+    {3166667, false}, {3183333, true}, {3200000, false}, {3216667, true},
+    {3233333, false}, {3250000, true}, {3266667, false}, {3283333, true},
+};
+
+static const int kMaxTestJitterUs = 2000;
+// return one of 1000, 0, -1000 as jitter.
+static int GetJitter(size_t i) {
+    return (1 - (i % 3)) * (kMaxTestJitterUs / 2);
+}
+
+class FrameDropperTest : public ::testing::Test {
+public:
+    FrameDropperTest() : mFrameDropper(new FrameDropper()) {
+        EXPECT_EQ(OK, mFrameDropper->setMaxFrameRate(30.0));
+    }
+
+protected:
+    void RunTest(const TestFrame* frames, size_t size) {
+        for (size_t i = 0; i < size; ++i) {
+            int jitter = GetJitter(i);
+            int64_t testTimeUs = frames[i].timeUs + jitter;
+            printf("time %lld, testTime %lld, jitter %d\n", frames[i].timeUs, testTimeUs, jitter);
+            EXPECT_EQ(frames[i].shouldDrop, mFrameDropper->shouldDrop(testTimeUs));
+        }
+    }
+
+    sp<FrameDropper> mFrameDropper;
+};
+
+TEST_F(FrameDropperTest, TestInvalidMaxFrameRate) {
+    EXPECT_NE(OK, mFrameDropper->setMaxFrameRate(-1.0));
+    EXPECT_NE(OK, mFrameDropper->setMaxFrameRate(0));
+}
+
+TEST_F(FrameDropperTest, Test20Fps) {
+    RunTest(testFrames20Fps, ARRAY_SIZE(testFrames20Fps));
+}
+
+TEST_F(FrameDropperTest, Test30Fps) {
+    RunTest(testFrames30Fps, ARRAY_SIZE(testFrames30Fps));
+}
+
+TEST_F(FrameDropperTest, Test40Fps) {
+    RunTest(testFrames40Fps, ARRAY_SIZE(testFrames40Fps));
+}
+
+TEST_F(FrameDropperTest, Test60Fps) {
+    RunTest(testFrames60Fps, ARRAY_SIZE(testFrames60Fps));
+}
+
+TEST_F(FrameDropperTest, TestVariableFps) {
+    RunTest(testFramesVariableFps, ARRAY_SIZE(testFramesVariableFps));
+}
+
+} // namespace android
diff --git a/media/libstagefright/webm/WebmWriter.cpp b/media/libstagefright/webm/WebmWriter.cpp
index 069961b..737f144 100644
--- a/media/libstagefright/webm/WebmWriter.cpp
+++ b/media/libstagefright/webm/WebmWriter.cpp
@@ -80,38 +80,6 @@
             mCuePoints);
 }
 
-WebmWriter::WebmWriter(const char *filename)
-    : mInitCheck(NO_INIT),
-      mTimeCodeScale(1000000),
-      mStartTimestampUs(0),
-      mStartTimeOffsetMs(0),
-      mSegmentOffset(0),
-      mSegmentDataStart(0),
-      mInfoOffset(0),
-      mInfoSize(0),
-      mTracksOffset(0),
-      mCuesOffset(0),
-      mPaused(false),
-      mStarted(false),
-      mIsFileSizeLimitExplicitlyRequested(false),
-      mIsRealTimeRecording(false),
-      mStreamableFile(true),
-      mEstimatedCuesSize(0) {
-    mFd = open(filename, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
-    if (mFd >= 0) {
-        ALOGV("fd %d; flags: %o", mFd, fcntl(mFd, F_GETFL, 0));
-        mInitCheck = OK;
-    }
-    mStreams[kAudioIndex] = WebmStream(kAudioType, "Audio", &WebmWriter::audioTrack);
-    mStreams[kVideoIndex] = WebmStream(kVideoType, "Video", &WebmWriter::videoTrack);
-    mSinkThread = new WebmFrameSinkThread(
-            mFd,
-            mSegmentDataStart,
-            mStreams[kVideoIndex].mSink,
-            mStreams[kAudioIndex].mSink,
-            mCuePoints);
-}
-
 // static
 sp<WebmElement> WebmWriter::videoTrack(const sp<MetaData>& md) {
     int32_t width, height;
diff --git a/media/libstagefright/webm/WebmWriter.h b/media/libstagefright/webm/WebmWriter.h
index 36b6965..4ad770e 100644
--- a/media/libstagefright/webm/WebmWriter.h
+++ b/media/libstagefright/webm/WebmWriter.h
@@ -37,7 +37,6 @@
 class WebmWriter : public MediaWriter {
 public:
     WebmWriter(int fd);
-    WebmWriter(const char *filename);
     ~WebmWriter() { reset(); }
 
 
diff --git a/media/mediaserver/Android.mk b/media/mediaserver/Android.mk
index 3a280f0..f1b84ad 100644
--- a/media/mediaserver/Android.mk
+++ b/media/mediaserver/Android.mk
@@ -11,7 +11,7 @@
 include $(CLEAR_VARS)
 
 LOCAL_SRC_FILES:= \
-	main_mediaserver.cpp 
+	main_mediaserver.cpp
 
 LOCAL_SHARED_LIBRARIES := \
 	libaudioflinger \
diff --git a/media/mediaserver/main_mediaserver.cpp b/media/mediaserver/main_mediaserver.cpp
index af1c9e6..263dd32 100644
--- a/media/mediaserver/main_mediaserver.cpp
+++ b/media/mediaserver/main_mediaserver.cpp
@@ -33,7 +33,7 @@
 #include "CameraService.h"
 #include "MediaLogService.h"
 #include "MediaPlayerService.h"
-#include "AudioPolicyService.h"
+#include "service/AudioPolicyService.h"
 #include "SoundTriggerHwService.h"
 
 using namespace android;
diff --git a/media/ndk/NdkMediaCodec.cpp b/media/ndk/NdkMediaCodec.cpp
index ed00b72..3124e4a 100644
--- a/media/ndk/NdkMediaCodec.cpp
+++ b/media/ndk/NdkMediaCodec.cpp
@@ -352,7 +352,8 @@
 }
 
 //EXPORT
-media_status_t AMediaCodec_setNotificationCallback(AMediaCodec *mData, OnCodecEvent callback, void *userdata) {
+media_status_t AMediaCodec_setNotificationCallback(AMediaCodec *mData, OnCodecEvent callback,
+        void *userdata) {
     mData->mCallback = callback;
     mData->mCallbackUserData = userdata;
     return AMEDIA_OK;
diff --git a/media/ndk/NdkMediaExtractor.cpp b/media/ndk/NdkMediaExtractor.cpp
index db57d0b..0ecd64f 100644
--- a/media/ndk/NdkMediaExtractor.cpp
+++ b/media/ndk/NdkMediaExtractor.cpp
@@ -70,7 +70,8 @@
 }
 
 EXPORT
-media_status_t AMediaExtractor_setDataSourceFd(AMediaExtractor *mData, int fd, off64_t offset, off64_t length) {
+media_status_t AMediaExtractor_setDataSourceFd(AMediaExtractor *mData, int fd, off64_t offset,
+        off64_t length) {
     ALOGV("setDataSource(%d, %lld, %lld)", fd, offset, length);
     return translate_error(mData->mImpl->setDataSource(fd, offset, length));
 }
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index 44d2553..642ff82 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -74,9 +74,17 @@
 LOCAL_MODULE:= libaudioflinger
 LOCAL_32_BIT_ONLY := true
 
-LOCAL_SRC_FILES += FastMixer.cpp FastMixerState.cpp AudioWatchdog.cpp
-LOCAL_SRC_FILES += FastThread.cpp FastThreadState.cpp
-LOCAL_SRC_FILES += FastCapture.cpp FastCaptureState.cpp
+LOCAL_SRC_FILES += \
+    AudioWatchdog.cpp        \
+    FastCapture.cpp          \
+    FastCaptureDumpState.cpp \
+    FastCaptureState.cpp     \
+    FastMixer.cpp            \
+    FastMixerDumpState.cpp   \
+    FastMixerState.cpp       \
+    FastThread.cpp           \
+    FastThreadDumpState.cpp  \
+    FastThreadState.cpp
 
 LOCAL_CFLAGS += -DSTATE_QUEUE_INSTANTIATIONS='"StateQueueInstantiations.cpp"'
 
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 993db73..f3780a9 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -185,7 +185,8 @@
     char value[PROPERTY_VALUE_MAX];
     bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
     if (doLog) {
-        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY);
+        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters",
+                MemoryHeapBase::READ_ONLY);
     }
 
 #ifdef TEE_SINK
@@ -401,6 +402,9 @@
             String8 result(kClientLockedString);
             write(fd, result.string(), result.size());
         }
+
+        EffectDumpEffects(fd);
+
         dumpClients(fd, args);
         if (clientLocked) {
             mClientLock.unlock();
@@ -822,14 +826,20 @@
     if (ret != NO_ERROR) {
         return false;
     }
-
+    bool mute = true;
     bool state = AUDIO_MODE_INVALID;
     AutoMutex lock(mHardwareLock);
-    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
     mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
-    dev->get_mic_mute(dev, &state);
+    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
+        audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
+        status_t result = dev->get_mic_mute(dev, &state);
+        if (result == NO_ERROR) {
+            mute = mute && state;
+        }
+    }
     mHardwareStatus = AUDIO_HW_IDLE;
-    return state;
+
+    return mute;
 }
 
 status_t AudioFlinger::setMasterMute(bool muted)
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index fd28ea1..0d4b358 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -430,6 +430,10 @@
     mState.mLog = log;
 }
 
+static inline audio_format_t selectMixerInFormat(audio_format_t inputFormat __unused) {
+    return kUseFloat && kUseNewMixer ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+}
+
 int AudioMixer::getTrackName(audio_channel_mask_t channelMask,
         audio_format_t format, int sessionId)
 {
@@ -492,24 +496,23 @@
         t->mInputBufferProvider = NULL;
         t->mReformatBufferProvider = NULL;
         t->downmixerBufferProvider = NULL;
+        t->mPostDownmixReformatBufferProvider = NULL;
         t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
         t->mFormat = format;
-        t->mMixerInFormat = kUseFloat && kUseNewMixer
-                ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+        t->mMixerInFormat = selectMixerInFormat(format);
+        t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required
         t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
                 AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
         t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
         // Check the downmixing (or upmixing) requirements.
-        status_t status = initTrackDownmix(t, n);
+        status_t status = t->prepareForDownmix();
         if (status != OK) {
             ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
             return -1;
         }
-        // initTrackDownmix() may change the input format requirement.
-        // If you desire floating point input to the mixer, it may change
-        // to integer because the downmixer requires integer to process.
+        // prepareForDownmix() may change mDownmixRequiresFormat
         ALOGVV("mMixerFormat:%#x  mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
-        prepareTrackForReformat(t, n);
+        t->prepareForReformat();
         mTrackNames |= 1 << n;
         return TRACK0 + n;
     }
@@ -526,7 +529,7 @@
  }
 
 // Called when channel masks have changed for a track name
-// TODO: Fix Downmixbufferprofider not to (possibly) change mixer input format,
+// TODO: Fix DownmixerBufferProvider not to (possibly) change mixer input format,
 // which will simplify this logic.
 bool AudioMixer::setChannelMasks(int name,
         audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) {
@@ -551,21 +554,18 @@
 
     // channel masks have changed, does this track need a downmixer?
     // update to try using our desired format (if we aren't already using it)
-    const audio_format_t prevMixerInFormat = track.mMixerInFormat;
-    track.mMixerInFormat = kUseFloat && kUseNewMixer
-            ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
-    const status_t status = initTrackDownmix(&mState.tracks[name], name);
+    const audio_format_t prevDownmixerFormat = track.mDownmixRequiresFormat;
+    const status_t status = mState.tracks[name].prepareForDownmix();
     ALOGE_IF(status != OK,
-            "initTrackDownmix error %d, track channel mask %#x, mixer channel mask %#x",
+            "prepareForDownmix error %d, track channel mask %#x, mixer channel mask %#x",
             status, track.channelMask, track.mMixerChannelMask);
 
-    const bool mixerInFormatChanged = prevMixerInFormat != track.mMixerInFormat;
-    if (mixerInFormatChanged) {
-        prepareTrackForReformat(&track, name); // because of downmixer, track format may change!
+    if (prevDownmixerFormat != track.mDownmixRequiresFormat) {
+        track.prepareForReformat(); // because of downmixer, track format may change!
     }
 
-    if (track.resampler && (mixerInFormatChanged || mixerChannelCountChanged)) {
-        // resampler input format or channels may have changed.
+    if (track.resampler && mixerChannelCountChanged) {
+        // resampler channels may have changed.
         const uint32_t resetToSampleRate = track.sampleRate;
         delete track.resampler;
         track.resampler = NULL;
@@ -576,99 +576,122 @@
     return true;
 }
 
-status_t AudioMixer::initTrackDownmix(track_t* pTrack, int trackName)
-{
-    // Only remix (upmix or downmix) if the track and mixer/device channel masks
-    // are not the same and not handled internally, as mono -> stereo currently is.
-    if (pTrack->channelMask != pTrack->mMixerChannelMask
-            && !(pTrack->channelMask == AUDIO_CHANNEL_OUT_MONO
-                    && pTrack->mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO)) {
-        return prepareTrackForDownmix(pTrack, trackName);
-    }
-    // no remix necessary
-    unprepareTrackForDownmix(pTrack, trackName);
-    return NO_ERROR;
-}
+void AudioMixer::track_t::unprepareForDownmix() {
+    ALOGV("AudioMixer::unprepareForDownmix(%p)", this);
 
-void AudioMixer::unprepareTrackForDownmix(track_t* pTrack, int trackName __unused) {
-    ALOGV("AudioMixer::unprepareTrackForDownmix(%d)", trackName);
-
-    if (pTrack->downmixerBufferProvider != NULL) {
+    mDownmixRequiresFormat = AUDIO_FORMAT_INVALID;
+    if (downmixerBufferProvider != NULL) {
         // this track had previously been configured with a downmixer, delete it
         ALOGV(" deleting old downmixer");
-        delete pTrack->downmixerBufferProvider;
-        pTrack->downmixerBufferProvider = NULL;
-        reconfigureBufferProviders(pTrack);
+        delete downmixerBufferProvider;
+        downmixerBufferProvider = NULL;
+        reconfigureBufferProviders();
     } else {
         ALOGV(" nothing to do, no downmixer to delete");
     }
 }
 
-status_t AudioMixer::prepareTrackForDownmix(track_t* pTrack, int trackName)
+status_t AudioMixer::track_t::prepareForDownmix()
 {
-    ALOGV("AudioMixer::prepareTrackForDownmix(%d) with mask 0x%x", trackName, pTrack->channelMask);
+    ALOGV("AudioMixer::prepareForDownmix(%p) with mask 0x%x",
+            this, channelMask);
 
     // discard the previous downmixer if there was one
-    unprepareTrackForDownmix(pTrack, trackName);
+    unprepareForDownmix();
+    // Only remix (upmix or downmix) if the track and mixer/device channel masks
+    // are not the same and not handled internally, as mono -> stereo currently is.
+    if (channelMask == mMixerChannelMask
+            || (channelMask == AUDIO_CHANNEL_OUT_MONO
+                    && mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO)) {
+        return NO_ERROR;
+    }
     if (DownmixerBufferProvider::isMultichannelCapable()) {
-        DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(pTrack->channelMask,
-                pTrack->mMixerChannelMask,
-                AUDIO_FORMAT_PCM_16_BIT /* TODO: use pTrack->mMixerInFormat, now only PCM 16 */,
-                pTrack->sampleRate, pTrack->sessionId, kCopyBufferFrameCount);
+        DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(channelMask,
+                mMixerChannelMask,
+                AUDIO_FORMAT_PCM_16_BIT /* TODO: use mMixerInFormat, now only PCM 16 */,
+                sampleRate, sessionId, kCopyBufferFrameCount);
 
         if (pDbp->isValid()) { // if constructor completed properly
-            pTrack->mMixerInFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix
-            pTrack->downmixerBufferProvider = pDbp;
-            reconfigureBufferProviders(pTrack);
+            mDownmixRequiresFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix
+            downmixerBufferProvider = pDbp;
+            reconfigureBufferProviders();
             return NO_ERROR;
         }
         delete pDbp;
     }
 
     // Effect downmixer does not accept the channel conversion.  Let's use our remixer.
-    RemixBufferProvider* pRbp = new RemixBufferProvider(pTrack->channelMask,
-            pTrack->mMixerChannelMask, pTrack->mMixerInFormat, kCopyBufferFrameCount);
+    RemixBufferProvider* pRbp = new RemixBufferProvider(channelMask,
+            mMixerChannelMask, mMixerInFormat, kCopyBufferFrameCount);
     // Remix always finds a conversion whereas Downmixer effect above may fail.
-    pTrack->downmixerBufferProvider = pRbp;
-    reconfigureBufferProviders(pTrack);
+    downmixerBufferProvider = pRbp;
+    reconfigureBufferProviders();
     return NO_ERROR;
 }
 
-void AudioMixer::unprepareTrackForReformat(track_t* pTrack, int trackName __unused) {
-    ALOGV("AudioMixer::unprepareTrackForReformat(%d)", trackName);
-    if (pTrack->mReformatBufferProvider != NULL) {
-        delete pTrack->mReformatBufferProvider;
-        pTrack->mReformatBufferProvider = NULL;
-        reconfigureBufferProviders(pTrack);
+void AudioMixer::track_t::unprepareForReformat() {
+    ALOGV("AudioMixer::unprepareForReformat(%p)", this);
+    bool requiresReconfigure = false;
+    if (mReformatBufferProvider != NULL) {
+        delete mReformatBufferProvider;
+        mReformatBufferProvider = NULL;
+        requiresReconfigure = true;
+    }
+    if (mPostDownmixReformatBufferProvider != NULL) {
+        delete mPostDownmixReformatBufferProvider;
+        mPostDownmixReformatBufferProvider = NULL;
+        requiresReconfigure = true;
+    }
+    if (requiresReconfigure) {
+        reconfigureBufferProviders();
     }
 }
 
-status_t AudioMixer::prepareTrackForReformat(track_t* pTrack, int trackName)
+status_t AudioMixer::track_t::prepareForReformat()
 {
-    ALOGV("AudioMixer::prepareTrackForReformat(%d) with format %#x", trackName, pTrack->mFormat);
-    // discard the previous reformatter if there was one
-    unprepareTrackForReformat(pTrack, trackName);
-    // only configure reformatter if needed
-    if (pTrack->mFormat != pTrack->mMixerInFormat) {
-        pTrack->mReformatBufferProvider = new ReformatBufferProvider(
-                audio_channel_count_from_out_mask(pTrack->channelMask),
-                pTrack->mFormat, pTrack->mMixerInFormat,
+    ALOGV("AudioMixer::prepareForReformat(%p) with format %#x", this, mFormat);
+    // discard previous reformatters
+    unprepareForReformat();
+    // only configure reformatters as needed
+    const audio_format_t targetFormat = mDownmixRequiresFormat != AUDIO_FORMAT_INVALID
+            ? mDownmixRequiresFormat : mMixerInFormat;
+    bool requiresReconfigure = false;
+    if (mFormat != targetFormat) {
+        mReformatBufferProvider = new ReformatBufferProvider(
+                audio_channel_count_from_out_mask(channelMask),
+                mFormat,
+                targetFormat,
                 kCopyBufferFrameCount);
-        reconfigureBufferProviders(pTrack);
+        requiresReconfigure = true;
+    }
+    if (targetFormat != mMixerInFormat) {
+        mPostDownmixReformatBufferProvider = new ReformatBufferProvider(
+                audio_channel_count_from_out_mask(mMixerChannelMask),
+                targetFormat,
+                mMixerInFormat,
+                kCopyBufferFrameCount);
+        requiresReconfigure = true;
+    }
+    if (requiresReconfigure) {
+        reconfigureBufferProviders();
     }
     return NO_ERROR;
 }
 
-void AudioMixer::reconfigureBufferProviders(track_t* pTrack)
+void AudioMixer::track_t::reconfigureBufferProviders()
 {
-    pTrack->bufferProvider = pTrack->mInputBufferProvider;
-    if (pTrack->mReformatBufferProvider) {
-        pTrack->mReformatBufferProvider->setBufferProvider(pTrack->bufferProvider);
-        pTrack->bufferProvider = pTrack->mReformatBufferProvider;
+    bufferProvider = mInputBufferProvider;
+    if (mReformatBufferProvider) {
+        mReformatBufferProvider->setBufferProvider(bufferProvider);
+        bufferProvider = mReformatBufferProvider;
     }
-    if (pTrack->downmixerBufferProvider) {
-        pTrack->downmixerBufferProvider->setBufferProvider(pTrack->bufferProvider);
-        pTrack->bufferProvider = pTrack->downmixerBufferProvider;
+    if (downmixerBufferProvider) {
+        downmixerBufferProvider->setBufferProvider(bufferProvider);
+        bufferProvider = downmixerBufferProvider;
+    }
+    if (mPostDownmixReformatBufferProvider) {
+        mPostDownmixReformatBufferProvider->setBufferProvider(bufferProvider);
+        bufferProvider = mPostDownmixReformatBufferProvider;
     }
 }
 
@@ -687,9 +710,9 @@
     delete track.resampler;
     track.resampler = NULL;
     // delete the downmixer
-    unprepareTrackForDownmix(&mState.tracks[name], name);
+    mState.tracks[name].unprepareForDownmix();
     // delete the reformatter
-    unprepareTrackForReformat(&mState.tracks[name], name);
+    mState.tracks[name].unprepareForReformat();
 
     mTrackNames &= ~(1<<name);
 }
@@ -828,7 +851,7 @@
                 ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format);
                 track.mFormat = format;
                 ALOGV("setParameter(TRACK, FORMAT, %#x)", format);
-                prepareTrackForReformat(&track, name);
+                track.prepareForReformat();
                 invalidateState(1 << name);
             }
             } break;
@@ -1032,10 +1055,13 @@
     if (mState.tracks[name].mReformatBufferProvider != NULL) {
         mState.tracks[name].mReformatBufferProvider->reset();
     } else if (mState.tracks[name].downmixerBufferProvider != NULL) {
+        mState.tracks[name].downmixerBufferProvider->reset();
+    } else if (mState.tracks[name].mPostDownmixReformatBufferProvider != NULL) {
+        mState.tracks[name].mPostDownmixReformatBufferProvider->reset();
     }
 
     mState.tracks[name].mInputBufferProvider = bufferProvider;
-    reconfigureBufferProviders(&mState.tracks[name]);
+    mState.tracks[name].reconfigureBufferProviders();
 }
 
 
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index f4f142b..c5df08a 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -127,10 +127,16 @@
     size_t      getUnreleasedFrames(int name) const;
 
     static inline bool isValidPcmTrackFormat(audio_format_t format) {
-        return format == AUDIO_FORMAT_PCM_16_BIT ||
-                format == AUDIO_FORMAT_PCM_24_BIT_PACKED ||
-                format == AUDIO_FORMAT_PCM_32_BIT ||
-                format == AUDIO_FORMAT_PCM_FLOAT;
+        switch (format) {
+        case AUDIO_FORMAT_PCM_8_BIT:
+        case AUDIO_FORMAT_PCM_16_BIT:
+        case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+        case AUDIO_FORMAT_PCM_32_BIT:
+        case AUDIO_FORMAT_PCM_FLOAT:
+            return true;
+        default:
+            return false;
+        }
     }
 
 private:
@@ -205,17 +211,34 @@
         int32_t*           auxBuffer;
 
         // 16-byte boundary
+
+        /* Buffer providers are constructed to translate the track input data as needed.
+         *
+         * 1) mInputBufferProvider: The AudioTrack buffer provider.
+         * 2) mReformatBufferProvider: If not NULL, performs the audio reformat to
+         *    match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer
+         *    requires reformat. For example, it may convert floating point input to
+         *    PCM_16_bit if that's required by the downmixer.
+         * 3) downmixerBufferProvider: If not NULL, performs the channel remixing to match
+         *    the number of channels required by the mixer sink.
+         * 4) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from
+         *    the downmixer requirements to the mixer engine input requirements.
+         */
         AudioBufferProvider*     mInputBufferProvider;    // externally provided buffer provider.
         CopyBufferProvider*      mReformatBufferProvider; // provider wrapper for reformatting.
         CopyBufferProvider*      downmixerBufferProvider; // wrapper for channel conversion.
-
-        int32_t     sessionId;
+        CopyBufferProvider*      mPostDownmixReformatBufferProvider;
 
         // 16-byte boundary
+        int32_t     sessionId;
+
         audio_format_t mMixerFormat;     // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
         audio_format_t mFormat;          // input track format
         audio_format_t mMixerInFormat;   // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
                                          // each track must be converted to this format.
+        audio_format_t mDownmixRequiresFormat;  // required downmixer format
+                                                // AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary
+                                                // AUDIO_FORMAT_INVALID if no required format
 
         float          mVolume[MAX_NUM_VOLUMES];     // floating point set volume
         float          mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
@@ -225,7 +248,6 @@
         float          mPrevAuxLevel;                 // floating point prev aux level
         float          mAuxInc;                       // floating point aux increment
 
-        // 16-byte boundary
         audio_channel_mask_t mMixerChannelMask;
         uint32_t             mMixerChannelCount;
 
@@ -236,6 +258,12 @@
         void        adjustVolumeRamp(bool aux, bool useFloat = false);
         size_t      getUnreleasedFrames() const { return resampler != NULL ?
                                                     resampler->getUnreleasedFrames() : 0; };
+
+        status_t    prepareForDownmix();
+        void        unprepareForDownmix();
+        status_t    prepareForReformat();
+        void        unprepareForReformat();
+        void        reconfigureBufferProviders();
     };
 
     typedef void (*process_hook_t)(state_t* state, int64_t pts);
@@ -382,14 +410,6 @@
     bool setChannelMasks(int name,
             audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
 
-    // TODO: remove unused trackName/trackNum from functions below.
-    static status_t initTrackDownmix(track_t* pTrack, int trackName);
-    static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum);
-    static void unprepareTrackForDownmix(track_t* pTrack, int trackName);
-    static status_t prepareTrackForReformat(track_t* pTrack, int trackNum);
-    static void unprepareTrackForReformat(track_t* pTrack, int trackName);
-    static void reconfigureBufferProviders(track_t* pTrack);
-
     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);
diff --git a/services/audioflinger/AudioResamplerFirGen.h b/services/audioflinger/AudioResamplerFirGen.h
index f3718b6..a9c84de 100644
--- a/services/audioflinger/AudioResamplerFirGen.h
+++ b/services/audioflinger/AudioResamplerFirGen.h
@@ -204,7 +204,8 @@
 
 template <>
 struct I0ATerm<0> { // 1/sqrt(2*PI);
-    static const CONSTEXPR double value = 0.398942280401432677939946059934381868475858631164934657665925;
+    static const CONSTEXPR double value =
+            0.398942280401432677939946059934381868475858631164934657665925;
 };
 
 #if USE_HORNERS_METHOD
diff --git a/services/audioflinger/AudioResamplerFirProcess.h b/services/audioflinger/AudioResamplerFirProcess.h
index efc8055..1118bf8 100644
--- a/services/audioflinger/AudioResamplerFirProcess.h
+++ b/services/audioflinger/AudioResamplerFirProcess.h
@@ -174,7 +174,8 @@
  * Process() calls ProcessBase() with TFUNC = InterpCompute, for interpolated phase.
  */
 
-template <int CHANNELS, int STRIDE, typename TFUNC, typename TC, typename TI, typename TO, typename TINTERP>
+template <int CHANNELS, int STRIDE, typename TFUNC, typename TC, typename TI, typename TO,
+        typename TINTERP>
 static inline
 void ProcessBase(TO* const out,
         size_t count,
@@ -268,7 +269,8 @@
         TINTERP lerpP,
         const TO* const volumeLR)
 {
-    ProcessBase<CHANNELS, STRIDE, InterpCompute>(out, count, coefsP, coefsN, sP, sN, lerpP, volumeLR);
+    ProcessBase<CHANNELS, STRIDE, InterpCompute>(out, count, coefsP, coefsN, sP, sN, lerpP,
+            volumeLR);
 }
 
 /*
diff --git a/services/audioflinger/FastCapture.cpp b/services/audioflinger/FastCapture.cpp
index 0c9b976..1c4f670 100644
--- a/services/audioflinger/FastCapture.cpp
+++ b/services/audioflinger/FastCapture.cpp
@@ -210,13 +210,4 @@
     }
 }
 
-FastCaptureDumpState::FastCaptureDumpState() : FastThreadDumpState(),
-    mReadSequence(0), mFramesRead(0), mReadErrors(0), mSampleRate(0), mFrameCount(0)
-{
-}
-
-FastCaptureDumpState::~FastCaptureDumpState()
-{
-}
-
 }   // namespace android
diff --git a/services/audioflinger/FastCapture.h b/services/audioflinger/FastCapture.h
index e535b9d..da0fe2f 100644
--- a/services/audioflinger/FastCapture.h
+++ b/services/audioflinger/FastCapture.h
@@ -20,23 +20,12 @@
 #include "FastThread.h"
 #include "StateQueue.h"
 #include "FastCaptureState.h"
+#include "FastCaptureDumpState.h"
 
 namespace android {
 
 typedef StateQueue<FastCaptureState> FastCaptureStateQueue;
 
-struct FastCaptureDumpState : FastThreadDumpState {
-    FastCaptureDumpState();
-    /*virtual*/ ~FastCaptureDumpState();
-
-    // FIXME by renaming, could pull up many of these to FastThreadDumpState
-    uint32_t mReadSequence;     // incremented before and after each read()
-    uint32_t mFramesRead;       // total number of frames read successfully
-    uint32_t mReadErrors;       // total number of read() errors
-    uint32_t mSampleRate;
-    size_t   mFrameCount;
-};
-
 class FastCapture : public FastThread {
 
 public:
diff --git a/services/audiopolicy/AudioPolicyFactory.cpp b/services/audioflinger/FastCaptureDumpState.cpp
similarity index 66%
copy from services/audiopolicy/AudioPolicyFactory.cpp
copy to services/audioflinger/FastCaptureDumpState.cpp
index 2ae7bc1..00f8da0 100644
--- a/services/audiopolicy/AudioPolicyFactory.cpp
+++ b/services/audioflinger/FastCaptureDumpState.cpp
@@ -14,19 +14,17 @@
  * limitations under the License.
  */
 
-#include "AudioPolicyManager.h"
+#include "FastCaptureDumpState.h"
 
 namespace android {
 
-extern "C" AudioPolicyInterface* createAudioPolicyManager(
-        AudioPolicyClientInterface *clientInterface)
+FastCaptureDumpState::FastCaptureDumpState() : FastThreadDumpState(),
+    mReadSequence(0), mFramesRead(0), mReadErrors(0), mSampleRate(0), mFrameCount(0)
 {
-    return new AudioPolicyManager(clientInterface);
 }
 
-extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
+FastCaptureDumpState::~FastCaptureDumpState()
 {
-    delete interface;
 }
 
-}; // namespace android
+}   // android
diff --git a/services/audioflinger/FastCaptureDumpState.h b/services/audioflinger/FastCaptureDumpState.h
new file mode 100644
index 0000000..ee99099
--- /dev/null
+++ b/services/audioflinger/FastCaptureDumpState.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIO_FAST_CAPTURE_DUMP_STATE_H
+#define ANDROID_AUDIO_FAST_CAPTURE_DUMP_STATE_H
+
+#include <stdint.h>
+#include "Configuration.h"
+#include "FastThreadDumpState.h"
+
+namespace android {
+
+struct FastCaptureDumpState : FastThreadDumpState {
+    FastCaptureDumpState();
+    /*virtual*/ ~FastCaptureDumpState();
+
+    // FIXME by renaming, could pull up many of these to FastThreadDumpState
+    uint32_t mReadSequence;     // incremented before and after each read()
+    uint32_t mFramesRead;       // total number of frames read successfully
+    uint32_t mReadErrors;       // total number of read() errors
+    uint32_t mSampleRate;
+    size_t   mFrameCount;
+};
+
+}   // android
+
+#endif  // ANDROID_AUDIO_FAST_CAPTURE_DUMP_STATE_H
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 2678cbf..67e2e6e 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -27,6 +27,7 @@
 
 #include "Configuration.h"
 #include <time.h>
+#include <utils/Debug.h>
 #include <utils/Log.h>
 #include <utils/Trace.h>
 #include <system/audio.h>
@@ -456,222 +457,4 @@
     }
 }
 
-FastMixerDumpState::FastMixerDumpState(
-#ifdef FAST_MIXER_STATISTICS
-        uint32_t samplingN
-#endif
-        ) : FastThreadDumpState(),
-    mWriteSequence(0), mFramesWritten(0),
-    mNumTracks(0), mWriteErrors(0),
-    mSampleRate(0), mFrameCount(0),
-    mTrackMask(0)
-{
-#ifdef FAST_MIXER_STATISTICS
-    increaseSamplingN(samplingN);
-#endif
-}
-
-#ifdef FAST_MIXER_STATISTICS
-void FastMixerDumpState::increaseSamplingN(uint32_t samplingN)
-{
-    if (samplingN <= mSamplingN || samplingN > kSamplingN || roundup(samplingN) != samplingN) {
-        return;
-    }
-    uint32_t additional = samplingN - mSamplingN;
-    // sample arrays aren't accessed atomically with respect to the bounds,
-    // so clearing reduces chance for dumpsys to read random uninitialized samples
-    memset(&mMonotonicNs[mSamplingN], 0, sizeof(mMonotonicNs[0]) * additional);
-    memset(&mLoadNs[mSamplingN], 0, sizeof(mLoadNs[0]) * additional);
-#ifdef CPU_FREQUENCY_STATISTICS
-    memset(&mCpukHz[mSamplingN], 0, sizeof(mCpukHz[0]) * additional);
-#endif
-    mSamplingN = samplingN;
-}
-#endif
-
-FastMixerDumpState::~FastMixerDumpState()
-{
-}
-
-// helper function called by qsort()
-static int compare_uint32_t(const void *pa, const void *pb)
-{
-    uint32_t a = *(const uint32_t *)pa;
-    uint32_t b = *(const uint32_t *)pb;
-    if (a < b) {
-        return -1;
-    } else if (a > b) {
-        return 1;
-    } else {
-        return 0;
-    }
-}
-
-void FastMixerDumpState::dump(int fd) const
-{
-    if (mCommand == FastMixerState::INITIAL) {
-        dprintf(fd, "  FastMixer not initialized\n");
-        return;
-    }
-#define COMMAND_MAX 32
-    char string[COMMAND_MAX];
-    switch (mCommand) {
-    case FastMixerState::INITIAL:
-        strcpy(string, "INITIAL");
-        break;
-    case FastMixerState::HOT_IDLE:
-        strcpy(string, "HOT_IDLE");
-        break;
-    case FastMixerState::COLD_IDLE:
-        strcpy(string, "COLD_IDLE");
-        break;
-    case FastMixerState::EXIT:
-        strcpy(string, "EXIT");
-        break;
-    case FastMixerState::MIX:
-        strcpy(string, "MIX");
-        break;
-    case FastMixerState::WRITE:
-        strcpy(string, "WRITE");
-        break;
-    case FastMixerState::MIX_WRITE:
-        strcpy(string, "MIX_WRITE");
-        break;
-    default:
-        snprintf(string, COMMAND_MAX, "%d", mCommand);
-        break;
-    }
-    double measuredWarmupMs = (mMeasuredWarmupTs.tv_sec * 1000.0) +
-            (mMeasuredWarmupTs.tv_nsec / 1000000.0);
-    double mixPeriodSec = (double) mFrameCount / (double) mSampleRate;
-    dprintf(fd, "  FastMixer command=%s writeSequence=%u framesWritten=%u\n"
-                "            numTracks=%u writeErrors=%u underruns=%u overruns=%u\n"
-                "            sampleRate=%u frameCount=%zu measuredWarmup=%.3g ms, warmupCycles=%u\n"
-                "            mixPeriod=%.2f ms\n",
-                 string, mWriteSequence, mFramesWritten,
-                 mNumTracks, mWriteErrors, mUnderruns, mOverruns,
-                 mSampleRate, mFrameCount, measuredWarmupMs, mWarmupCycles,
-                 mixPeriodSec * 1e3);
-#ifdef FAST_MIXER_STATISTICS
-    // find the interval of valid samples
-    uint32_t bounds = mBounds;
-    uint32_t newestOpen = bounds & 0xFFFF;
-    uint32_t oldestClosed = bounds >> 16;
-    uint32_t n = (newestOpen - oldestClosed) & 0xFFFF;
-    if (n > mSamplingN) {
-        ALOGE("too many samples %u", n);
-        n = mSamplingN;
-    }
-    // statistics for monotonic (wall clock) time, thread raw CPU load in time, CPU clock frequency,
-    // and adjusted CPU load in MHz normalized for CPU clock frequency
-    CentralTendencyStatistics wall, loadNs;
-#ifdef CPU_FREQUENCY_STATISTICS
-    CentralTendencyStatistics kHz, loadMHz;
-    uint32_t previousCpukHz = 0;
-#endif
-    // Assuming a normal distribution for cycle times, three standard deviations on either side of
-    // the mean account for 99.73% of the population.  So if we take each tail to be 1/1000 of the
-    // sample set, we get 99.8% combined, or close to three standard deviations.
-    static const uint32_t kTailDenominator = 1000;
-    uint32_t *tail = n >= kTailDenominator ? new uint32_t[n] : NULL;
-    // loop over all the samples
-    for (uint32_t j = 0; j < n; ++j) {
-        size_t i = oldestClosed++ & (mSamplingN - 1);
-        uint32_t wallNs = mMonotonicNs[i];
-        if (tail != NULL) {
-            tail[j] = wallNs;
-        }
-        wall.sample(wallNs);
-        uint32_t sampleLoadNs = mLoadNs[i];
-        loadNs.sample(sampleLoadNs);
-#ifdef CPU_FREQUENCY_STATISTICS
-        uint32_t sampleCpukHz = mCpukHz[i];
-        // skip bad kHz samples
-        if ((sampleCpukHz & ~0xF) != 0) {
-            kHz.sample(sampleCpukHz >> 4);
-            if (sampleCpukHz == previousCpukHz) {
-                double megacycles = (double) sampleLoadNs * (double) (sampleCpukHz >> 4) * 1e-12;
-                double adjMHz = megacycles / mixPeriodSec;  // _not_ wallNs * 1e9
-                loadMHz.sample(adjMHz);
-            }
-        }
-        previousCpukHz = sampleCpukHz;
-#endif
-    }
-    if (n) {
-        dprintf(fd, "  Simple moving statistics over last %.1f seconds:\n",
-                    wall.n() * mixPeriodSec);
-        dprintf(fd, "    wall clock time in ms per mix cycle:\n"
-                    "      mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
-                    wall.mean()*1e-6, wall.minimum()*1e-6, wall.maximum()*1e-6,
-                    wall.stddev()*1e-6);
-        dprintf(fd, "    raw CPU load in us per mix cycle:\n"
-                    "      mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
-                    loadNs.mean()*1e-3, loadNs.minimum()*1e-3, loadNs.maximum()*1e-3,
-                    loadNs.stddev()*1e-3);
-    } else {
-        dprintf(fd, "  No FastMixer statistics available currently\n");
-    }
-#ifdef CPU_FREQUENCY_STATISTICS
-    dprintf(fd, "  CPU clock frequency in MHz:\n"
-                "    mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
-                kHz.mean()*1e-3, kHz.minimum()*1e-3, kHz.maximum()*1e-3, kHz.stddev()*1e-3);
-    dprintf(fd, "  adjusted CPU load in MHz (i.e. normalized for CPU clock frequency):\n"
-                "    mean=%.1f min=%.1f max=%.1f stddev=%.1f\n",
-                loadMHz.mean(), loadMHz.minimum(), loadMHz.maximum(), loadMHz.stddev());
-#endif
-    if (tail != NULL) {
-        qsort(tail, n, sizeof(uint32_t), compare_uint32_t);
-        // assume same number of tail samples on each side, left and right
-        uint32_t count = n / kTailDenominator;
-        CentralTendencyStatistics left, right;
-        for (uint32_t i = 0; i < count; ++i) {
-            left.sample(tail[i]);
-            right.sample(tail[n - (i + 1)]);
-        }
-        dprintf(fd, "  Distribution of mix cycle times in ms for the tails (> ~3 stddev outliers):\n"
-                    "    left tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n"
-                    "    right tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
-                    left.mean()*1e-6, left.minimum()*1e-6, left.maximum()*1e-6, left.stddev()*1e-6,
-                    right.mean()*1e-6, right.minimum()*1e-6, right.maximum()*1e-6,
-                    right.stddev()*1e-6);
-        delete[] tail;
-    }
-#endif
-    // The active track mask and track states are updated non-atomically.
-    // So if we relied on isActive to decide whether to display,
-    // then we might display an obsolete track or omit an active track.
-    // Instead we always display all tracks, with an indication
-    // of whether we think the track is active.
-    uint32_t trackMask = mTrackMask;
-    dprintf(fd, "  Fast tracks: kMaxFastTracks=%u activeMask=%#x\n",
-            FastMixerState::kMaxFastTracks, trackMask);
-    dprintf(fd, "  Index Active Full Partial Empty  Recent Ready\n");
-    for (uint32_t i = 0; i < FastMixerState::kMaxFastTracks; ++i, trackMask >>= 1) {
-        bool isActive = trackMask & 1;
-        const FastTrackDump *ftDump = &mTracks[i];
-        const FastTrackUnderruns& underruns = ftDump->mUnderruns;
-        const char *mostRecent;
-        switch (underruns.mBitFields.mMostRecent) {
-        case UNDERRUN_FULL:
-            mostRecent = "full";
-            break;
-        case UNDERRUN_PARTIAL:
-            mostRecent = "partial";
-            break;
-        case UNDERRUN_EMPTY:
-            mostRecent = "empty";
-            break;
-        default:
-            mostRecent = "?";
-            break;
-        }
-        dprintf(fd, "  %5u %6s %4u %7u %5u %7s %5zu\n", i, isActive ? "yes" : "no",
-                (underruns.mBitFields.mFull) & UNDERRUN_MASK,
-                (underruns.mBitFields.mPartial) & UNDERRUN_MASK,
-                (underruns.mBitFields.mEmpty) & UNDERRUN_MASK,
-                mostRecent, ftDump->mFramesReady);
-    }
-}
-
 }   // namespace android
diff --git a/services/audioflinger/FastMixer.h b/services/audioflinger/FastMixer.h
index fde8c2b..7649db2 100644
--- a/services/audioflinger/FastMixer.h
+++ b/services/audioflinger/FastMixer.h
@@ -17,11 +17,7 @@
 #ifndef ANDROID_AUDIO_FAST_MIXER_H
 #define ANDROID_AUDIO_FAST_MIXER_H
 
-#include <linux/futex.h>
-#include <sys/syscall.h>
-#include <utils/Debug.h>
 #include "FastThread.h"
-#include <utils/Thread.h>
 #include "StateQueue.h"
 #include "FastMixerState.h"
 #include "FastMixerDumpState.h"
diff --git a/services/audioflinger/FastMixerDumpState.cpp b/services/audioflinger/FastMixerDumpState.cpp
new file mode 100644
index 0000000..0ddd908
--- /dev/null
+++ b/services/audioflinger/FastMixerDumpState.cpp
@@ -0,0 +1,252 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "FastMixerDumpState"
+//#define LOG_NDEBUG 0
+
+#include "Configuration.h"
+#ifdef FAST_MIXER_STATISTICS
+#include <cpustats/CentralTendencyStatistics.h>
+#ifdef CPU_FREQUENCY_STATISTICS
+#include <cpustats/ThreadCpuUsage.h>
+#endif
+#endif
+#include <utils/Debug.h>
+#include <utils/Log.h>
+#include "FastMixerDumpState.h"
+
+namespace android {
+
+FastMixerDumpState::FastMixerDumpState(
+#ifdef FAST_MIXER_STATISTICS
+        uint32_t samplingN
+#endif
+        ) : FastThreadDumpState(),
+    mWriteSequence(0), mFramesWritten(0),
+    mNumTracks(0), mWriteErrors(0),
+    mSampleRate(0), mFrameCount(0),
+    mTrackMask(0)
+{
+#ifdef FAST_MIXER_STATISTICS
+    increaseSamplingN(samplingN);
+#endif
+}
+
+#ifdef FAST_MIXER_STATISTICS
+void FastMixerDumpState::increaseSamplingN(uint32_t samplingN)
+{
+    if (samplingN <= mSamplingN || samplingN > kSamplingN || roundup(samplingN) != samplingN) {
+        return;
+    }
+    uint32_t additional = samplingN - mSamplingN;
+    // sample arrays aren't accessed atomically with respect to the bounds,
+    // so clearing reduces chance for dumpsys to read random uninitialized samples
+    memset(&mMonotonicNs[mSamplingN], 0, sizeof(mMonotonicNs[0]) * additional);
+    memset(&mLoadNs[mSamplingN], 0, sizeof(mLoadNs[0]) * additional);
+#ifdef CPU_FREQUENCY_STATISTICS
+    memset(&mCpukHz[mSamplingN], 0, sizeof(mCpukHz[0]) * additional);
+#endif
+    mSamplingN = samplingN;
+}
+#endif
+
+FastMixerDumpState::~FastMixerDumpState()
+{
+}
+
+// helper function called by qsort()
+static int compare_uint32_t(const void *pa, const void *pb)
+{
+    uint32_t a = *(const uint32_t *)pa;
+    uint32_t b = *(const uint32_t *)pb;
+    if (a < b) {
+        return -1;
+    } else if (a > b) {
+        return 1;
+    } else {
+        return 0;
+    }
+}
+
+void FastMixerDumpState::dump(int fd) const
+{
+    if (mCommand == FastMixerState::INITIAL) {
+        dprintf(fd, "  FastMixer not initialized\n");
+        return;
+    }
+#define COMMAND_MAX 32
+    char string[COMMAND_MAX];
+    switch (mCommand) {
+    case FastMixerState::INITIAL:
+        strcpy(string, "INITIAL");
+        break;
+    case FastMixerState::HOT_IDLE:
+        strcpy(string, "HOT_IDLE");
+        break;
+    case FastMixerState::COLD_IDLE:
+        strcpy(string, "COLD_IDLE");
+        break;
+    case FastMixerState::EXIT:
+        strcpy(string, "EXIT");
+        break;
+    case FastMixerState::MIX:
+        strcpy(string, "MIX");
+        break;
+    case FastMixerState::WRITE:
+        strcpy(string, "WRITE");
+        break;
+    case FastMixerState::MIX_WRITE:
+        strcpy(string, "MIX_WRITE");
+        break;
+    default:
+        snprintf(string, COMMAND_MAX, "%d", mCommand);
+        break;
+    }
+    double measuredWarmupMs = (mMeasuredWarmupTs.tv_sec * 1000.0) +
+            (mMeasuredWarmupTs.tv_nsec / 1000000.0);
+    double mixPeriodSec = (double) mFrameCount / (double) mSampleRate;
+    dprintf(fd, "  FastMixer command=%s writeSequence=%u framesWritten=%u\n"
+                "            numTracks=%u writeErrors=%u underruns=%u overruns=%u\n"
+                "            sampleRate=%u frameCount=%zu measuredWarmup=%.3g ms, warmupCycles=%u\n"
+                "            mixPeriod=%.2f ms\n",
+                 string, mWriteSequence, mFramesWritten,
+                 mNumTracks, mWriteErrors, mUnderruns, mOverruns,
+                 mSampleRate, mFrameCount, measuredWarmupMs, mWarmupCycles,
+                 mixPeriodSec * 1e3);
+#ifdef FAST_MIXER_STATISTICS
+    // find the interval of valid samples
+    uint32_t bounds = mBounds;
+    uint32_t newestOpen = bounds & 0xFFFF;
+    uint32_t oldestClosed = bounds >> 16;
+    uint32_t n = (newestOpen - oldestClosed) & 0xFFFF;
+    if (n > mSamplingN) {
+        ALOGE("too many samples %u", n);
+        n = mSamplingN;
+    }
+    // statistics for monotonic (wall clock) time, thread raw CPU load in time, CPU clock frequency,
+    // and adjusted CPU load in MHz normalized for CPU clock frequency
+    CentralTendencyStatistics wall, loadNs;
+#ifdef CPU_FREQUENCY_STATISTICS
+    CentralTendencyStatistics kHz, loadMHz;
+    uint32_t previousCpukHz = 0;
+#endif
+    // Assuming a normal distribution for cycle times, three standard deviations on either side of
+    // the mean account for 99.73% of the population.  So if we take each tail to be 1/1000 of the
+    // sample set, we get 99.8% combined, or close to three standard deviations.
+    static const uint32_t kTailDenominator = 1000;
+    uint32_t *tail = n >= kTailDenominator ? new uint32_t[n] : NULL;
+    // loop over all the samples
+    for (uint32_t j = 0; j < n; ++j) {
+        size_t i = oldestClosed++ & (mSamplingN - 1);
+        uint32_t wallNs = mMonotonicNs[i];
+        if (tail != NULL) {
+            tail[j] = wallNs;
+        }
+        wall.sample(wallNs);
+        uint32_t sampleLoadNs = mLoadNs[i];
+        loadNs.sample(sampleLoadNs);
+#ifdef CPU_FREQUENCY_STATISTICS
+        uint32_t sampleCpukHz = mCpukHz[i];
+        // skip bad kHz samples
+        if ((sampleCpukHz & ~0xF) != 0) {
+            kHz.sample(sampleCpukHz >> 4);
+            if (sampleCpukHz == previousCpukHz) {
+                double megacycles = (double) sampleLoadNs * (double) (sampleCpukHz >> 4) * 1e-12;
+                double adjMHz = megacycles / mixPeriodSec;  // _not_ wallNs * 1e9
+                loadMHz.sample(adjMHz);
+            }
+        }
+        previousCpukHz = sampleCpukHz;
+#endif
+    }
+    if (n) {
+        dprintf(fd, "  Simple moving statistics over last %.1f seconds:\n",
+                    wall.n() * mixPeriodSec);
+        dprintf(fd, "    wall clock time in ms per mix cycle:\n"
+                    "      mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
+                    wall.mean()*1e-6, wall.minimum()*1e-6, wall.maximum()*1e-6,
+                    wall.stddev()*1e-6);
+        dprintf(fd, "    raw CPU load in us per mix cycle:\n"
+                    "      mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
+                    loadNs.mean()*1e-3, loadNs.minimum()*1e-3, loadNs.maximum()*1e-3,
+                    loadNs.stddev()*1e-3);
+    } else {
+        dprintf(fd, "  No FastMixer statistics available currently\n");
+    }
+#ifdef CPU_FREQUENCY_STATISTICS
+    dprintf(fd, "  CPU clock frequency in MHz:\n"
+                "    mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
+                kHz.mean()*1e-3, kHz.minimum()*1e-3, kHz.maximum()*1e-3, kHz.stddev()*1e-3);
+    dprintf(fd, "  adjusted CPU load in MHz (i.e. normalized for CPU clock frequency):\n"
+                "    mean=%.1f min=%.1f max=%.1f stddev=%.1f\n",
+                loadMHz.mean(), loadMHz.minimum(), loadMHz.maximum(), loadMHz.stddev());
+#endif
+    if (tail != NULL) {
+        qsort(tail, n, sizeof(uint32_t), compare_uint32_t);
+        // assume same number of tail samples on each side, left and right
+        uint32_t count = n / kTailDenominator;
+        CentralTendencyStatistics left, right;
+        for (uint32_t i = 0; i < count; ++i) {
+            left.sample(tail[i]);
+            right.sample(tail[n - (i + 1)]);
+        }
+        dprintf(fd, "  Distribution of mix cycle times in ms for the tails "
+                    "(> ~3 stddev outliers):\n"
+                    "    left tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n"
+                    "    right tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
+                    left.mean()*1e-6, left.minimum()*1e-6, left.maximum()*1e-6, left.stddev()*1e-6,
+                    right.mean()*1e-6, right.minimum()*1e-6, right.maximum()*1e-6,
+                    right.stddev()*1e-6);
+        delete[] tail;
+    }
+#endif
+    // The active track mask and track states are updated non-atomically.
+    // So if we relied on isActive to decide whether to display,
+    // then we might display an obsolete track or omit an active track.
+    // Instead we always display all tracks, with an indication
+    // of whether we think the track is active.
+    uint32_t trackMask = mTrackMask;
+    dprintf(fd, "  Fast tracks: kMaxFastTracks=%u activeMask=%#x\n",
+            FastMixerState::kMaxFastTracks, trackMask);
+    dprintf(fd, "  Index Active Full Partial Empty  Recent Ready\n");
+    for (uint32_t i = 0; i < FastMixerState::kMaxFastTracks; ++i, trackMask >>= 1) {
+        bool isActive = trackMask & 1;
+        const FastTrackDump *ftDump = &mTracks[i];
+        const FastTrackUnderruns& underruns = ftDump->mUnderruns;
+        const char *mostRecent;
+        switch (underruns.mBitFields.mMostRecent) {
+        case UNDERRUN_FULL:
+            mostRecent = "full";
+            break;
+        case UNDERRUN_PARTIAL:
+            mostRecent = "partial";
+            break;
+        case UNDERRUN_EMPTY:
+            mostRecent = "empty";
+            break;
+        default:
+            mostRecent = "?";
+            break;
+        }
+        dprintf(fd, "  %5u %6s %4u %7u %5u %7s %5zu\n", i, isActive ? "yes" : "no",
+                (underruns.mBitFields.mFull) & UNDERRUN_MASK,
+                (underruns.mBitFields.mPartial) & UNDERRUN_MASK,
+                (underruns.mBitFields.mEmpty) & UNDERRUN_MASK,
+                mostRecent, ftDump->mFramesReady);
+    }
+}
+
+}   // android
diff --git a/services/audioflinger/FastMixerDumpState.h b/services/audioflinger/FastMixerDumpState.h
index 6a1e4649..f8354dd 100644
--- a/services/audioflinger/FastMixerDumpState.h
+++ b/services/audioflinger/FastMixerDumpState.h
@@ -17,7 +17,10 @@
 #ifndef ANDROID_AUDIO_FAST_MIXER_DUMP_STATE_H
 #define ANDROID_AUDIO_FAST_MIXER_DUMP_STATE_H
 
+#include <stdint.h>
 #include "Configuration.h"
+#include "FastThreadDumpState.h"
+#include "FastMixerState.h"
 
 namespace android {
 
diff --git a/services/audioflinger/FastThread.cpp b/services/audioflinger/FastThread.cpp
index 216dace..3e12cca 100644
--- a/services/audioflinger/FastThread.cpp
+++ b/services/audioflinger/FastThread.cpp
@@ -25,6 +25,7 @@
 #include <utils/Log.h>
 #include <utils/Trace.h>
 #include "FastThread.h"
+#include "FastThreadDumpState.h"
 
 #define FAST_DEFAULT_NS    999999999L   // ~1 sec: default time to sleep
 #define FAST_HOT_IDLE_NS     1000000L   // 1 ms: time to sleep while hot idling
diff --git a/services/audiopolicy/AudioPolicyFactory.cpp b/services/audioflinger/FastThreadDumpState.cpp
similarity index 61%
copy from services/audiopolicy/AudioPolicyFactory.cpp
copy to services/audioflinger/FastThreadDumpState.cpp
index 2ae7bc1..d7b825d 100644
--- a/services/audiopolicy/AudioPolicyFactory.cpp
+++ b/services/audioflinger/FastThreadDumpState.cpp
@@ -14,19 +14,24 @@
  * limitations under the License.
  */
 
-#include "AudioPolicyManager.h"
+#include "FastThreadDumpState.h"
 
 namespace android {
 
-extern "C" AudioPolicyInterface* createAudioPolicyManager(
-        AudioPolicyClientInterface *clientInterface)
+FastThreadDumpState::FastThreadDumpState() :
+    mCommand(FastThreadState::INITIAL), mUnderruns(0), mOverruns(0),
+    /* mMeasuredWarmupTs({0, 0}), */
+    mWarmupCycles(0)
+#ifdef FAST_MIXER_STATISTICS
+    , mSamplingN(1), mBounds(0)
+#endif
 {
-    return new AudioPolicyManager(clientInterface);
+    mMeasuredWarmupTs.tv_sec = 0;
+    mMeasuredWarmupTs.tv_nsec = 0;
 }
 
-extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
+FastThreadDumpState::~FastThreadDumpState()
 {
-    delete interface;
 }
 
-}; // namespace android
+}   // android
diff --git a/services/audioflinger/FastThreadDumpState.h b/services/audioflinger/FastThreadDumpState.h
new file mode 100644
index 0000000..17afbe5
--- /dev/null
+++ b/services/audioflinger/FastThreadDumpState.h
@@ -0,0 +1,61 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIO_FAST_THREAD_DUMP_STATE_H
+#define ANDROID_AUDIO_FAST_THREAD_DUMP_STATE_H
+
+#include "Configuration.h"
+#include "FastThreadState.h"
+
+namespace android {
+
+// FIXME extract common part of comment at FastMixerDumpState
+struct FastThreadDumpState {
+    FastThreadDumpState();
+    /*virtual*/ ~FastThreadDumpState();
+
+    FastThreadState::Command mCommand;   // current command
+    uint32_t mUnderruns;        // total number of underruns
+    uint32_t mOverruns;         // total number of overruns
+    struct timespec mMeasuredWarmupTs;  // measured warmup time
+    uint32_t mWarmupCycles;     // number of loop cycles required to warmup
+
+#ifdef FAST_MIXER_STATISTICS
+    // Recently collected samples of per-cycle monotonic time, thread CPU time, and CPU frequency.
+    // kSamplingN is max size of sampling frame (statistics), and must be a power of 2 <= 0x8000.
+    // The sample arrays are virtually allocated based on this compile-time constant,
+    // but are only initialized and used based on the runtime parameter mSamplingN.
+    static const uint32_t kSamplingN = 0x8000;
+    // Corresponding runtime maximum size of sample arrays, must be a power of 2 <= kSamplingN.
+    uint32_t mSamplingN;
+    // The bounds define the interval of valid samples, and are represented as follows:
+    //      newest open (excluded) endpoint   = lower 16 bits of bounds, modulo N
+    //      oldest closed (included) endpoint = upper 16 bits of bounds, modulo N
+    // Number of valid samples is newest - oldest.
+    uint32_t mBounds;                   // bounds for mMonotonicNs, mThreadCpuNs, and mCpukHz
+    // The elements in the *Ns arrays are in units of nanoseconds <= 3999999999.
+    uint32_t mMonotonicNs[kSamplingN];  // delta monotonic (wall clock) time
+    uint32_t mLoadNs[kSamplingN];       // delta CPU load in time
+#ifdef CPU_FREQUENCY_STATISTICS
+    uint32_t mCpukHz[kSamplingN];       // absolute CPU clock frequency in kHz, bits 0-3 are CPU#
+#endif
+#endif
+
+};  // struct FastThreadDumpState
+
+}   // android
+
+#endif  // ANDROID_AUDIO_FAST_THREAD_DUMP_STATE_H
diff --git a/services/audioflinger/FastThreadState.cpp b/services/audioflinger/FastThreadState.cpp
index 6994872..e6cf85c 100644
--- a/services/audioflinger/FastThreadState.cpp
+++ b/services/audioflinger/FastThreadState.cpp
@@ -29,21 +29,4 @@
 {
 }
 
-
-FastThreadDumpState::FastThreadDumpState() :
-    mCommand(FastThreadState::INITIAL), mUnderruns(0), mOverruns(0),
-    /* mMeasuredWarmupTs({0, 0}), */
-    mWarmupCycles(0)
-#ifdef FAST_MIXER_STATISTICS
-    , mSamplingN(1), mBounds(0)
-#endif
-{
-    mMeasuredWarmupTs.tv_sec = 0;
-    mMeasuredWarmupTs.tv_nsec = 0;
-}
-
-FastThreadDumpState::~FastThreadDumpState()
-{
-}
-
 }   // namespace android
diff --git a/services/audioflinger/FastThreadState.h b/services/audioflinger/FastThreadState.h
index 1ab8a0a..011921d 100644
--- a/services/audioflinger/FastThreadState.h
+++ b/services/audioflinger/FastThreadState.h
@@ -48,41 +48,6 @@
 
 };  // struct FastThreadState
 
-
-// FIXME extract common part of comment at FastMixerDumpState
-struct FastThreadDumpState {
-    FastThreadDumpState();
-    /*virtual*/ ~FastThreadDumpState();
-
-    FastThreadState::Command mCommand;   // current command
-    uint32_t mUnderruns;        // total number of underruns
-    uint32_t mOverruns;         // total number of overruns
-    struct timespec mMeasuredWarmupTs;  // measured warmup time
-    uint32_t mWarmupCycles;     // number of loop cycles required to warmup
-
-#ifdef FAST_MIXER_STATISTICS
-    // Recently collected samples of per-cycle monotonic time, thread CPU time, and CPU frequency.
-    // kSamplingN is max size of sampling frame (statistics), and must be a power of 2 <= 0x8000.
-    // The sample arrays are virtually allocated based on this compile-time constant,
-    // but are only initialized and used based on the runtime parameter mSamplingN.
-    static const uint32_t kSamplingN = 0x8000;
-    // Corresponding runtime maximum size of sample arrays, must be a power of 2 <= kSamplingN.
-    uint32_t mSamplingN;
-    // The bounds define the interval of valid samples, and are represented as follows:
-    //      newest open (excluded) endpoint   = lower 16 bits of bounds, modulo N
-    //      oldest closed (included) endpoint = upper 16 bits of bounds, modulo N
-    // Number of valid samples is newest - oldest.
-    uint32_t mBounds;                   // bounds for mMonotonicNs, mThreadCpuNs, and mCpukHz
-    // The elements in the *Ns arrays are in units of nanoseconds <= 3999999999.
-    uint32_t mMonotonicNs[kSamplingN];  // delta monotonic (wall clock) time
-    uint32_t mLoadNs[kSamplingN];       // delta CPU load in time
-#ifdef CPU_FREQUENCY_STATISTICS
-    uint32_t mCpukHz[kSamplingN];       // absolute CPU clock frequency in kHz, bits 0-3 are CPU#
-#endif
-#endif
-
-};  // struct FastThreadDumpState
-
 }   // android
 
 #endif  // ANDROID_AUDIO_FAST_THREAD_STATE_H
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index ee48276..902d5e4 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -255,7 +255,7 @@
 
     class Buffer : public AudioBufferProvider::Buffer {
     public:
-        int16_t *mBuffer;
+        void *mBuffer;
     };
 
                         OutputTrack(PlaybackThread *thread,
@@ -271,7 +271,7 @@
                                     AudioSystem::SYNC_EVENT_NONE,
                              int triggerSession = 0);
     virtual void        stop();
-            bool        write(int16_t* data, uint32_t frames);
+            bool        write(void* data, uint32_t frames);
             bool        bufferQueueEmpty() const { return mBufferQueue.size() == 0; }
             bool        isActive() const { return mActive; }
     const wp<ThreadBase>& thread() const { return mThread; }
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 51025fe..384bd25 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -23,7 +23,9 @@
 #include "Configuration.h"
 #include <math.h>
 #include <fcntl.h>
+#include <linux/futex.h>
 #include <sys/stat.h>
+#include <sys/syscall.h>
 #include <cutils/properties.h>
 #include <media/AudioParameter.h>
 #include <media/AudioResamplerPublic.h>
@@ -172,6 +174,18 @@
 // and that all "fast" AudioRecord clients read from.  In either case, the size can be small.
 static const size_t kRecordThreadReadOnlyHeapSize = 0x2000;
 
+// Returns the source frames needed to resample to destination frames.  This is not a precise
+// value and depends on the resampler (and possibly how it handles rounding internally).
+// If srcSampleRate and dstSampleRate are equal, then it returns destination frames, which
+// may not be a true if the resampler is asynchronous.
+static inline size_t sourceFramesNeeded(
+        uint32_t srcSampleRate, size_t dstFramesRequired, uint32_t dstSampleRate) {
+    // +1 for rounding - always do this even if matched ratio
+    // +1 for additional sample needed for interpolation
+    return srcSampleRate == dstSampleRate ? dstFramesRequired :
+            size_t((uint64_t)dstFramesRequired * srcSampleRate / dstSampleRate + 1 + 1);
+}
+
 // ----------------------------------------------------------------------------
 
 static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT;
@@ -314,6 +328,64 @@
 //      ThreadBase
 // ----------------------------------------------------------------------------
 
+// static
+const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type)
+{
+    switch (type) {
+    case MIXER:
+        return "MIXER";
+    case DIRECT:
+        return "DIRECT";
+    case DUPLICATING:
+        return "DUPLICATING";
+    case RECORD:
+        return "RECORD";
+    case OFFLOAD:
+        return "OFFLOAD";
+    default:
+        return "unknown";
+    }
+}
+
+static String8 outputFlagsToString(audio_output_flags_t flags)
+{
+    static const struct mapping {
+        audio_output_flags_t    mFlag;
+        const char *            mString;
+    } mappings[] = {
+        AUDIO_OUTPUT_FLAG_DIRECT,           "DIRECT",
+        AUDIO_OUTPUT_FLAG_PRIMARY,          "PRIMARY",
+        AUDIO_OUTPUT_FLAG_FAST,             "FAST",
+        AUDIO_OUTPUT_FLAG_DEEP_BUFFER,      "DEEP_BUFFER",
+        AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAAD",
+        AUDIO_OUTPUT_FLAG_NON_BLOCKING,     "NON_BLOCKING",
+        AUDIO_OUTPUT_FLAG_HW_AV_SYNC,       "HW_AV_SYNC",
+        AUDIO_OUTPUT_FLAG_NONE,             "NONE",         // must be last
+    };
+    String8 result;
+    audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE;
+    const mapping *entry;
+    for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) {
+        allFlags = (audio_output_flags_t) (allFlags | entry->mFlag);
+        if (flags & entry->mFlag) {
+            if (!result.isEmpty()) {
+                result.append("|");
+            }
+            result.append(entry->mString);
+        }
+    }
+    if (flags & ~allFlags) {
+        if (!result.isEmpty()) {
+            result.append("|");
+        }
+        result.appendFormat("0x%X", flags & ~allFlags);
+    }
+    if (result.isEmpty()) {
+        result.append(entry->mString);
+    }
+    return result;
+}
+
 AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
         audio_devices_t outDevice, audio_devices_t inDevice, type_t type)
     :   Thread(false /*canCallJava*/),
@@ -577,20 +649,21 @@
 
     bool locked = AudioFlinger::dumpTryLock(mLock);
     if (!locked) {
-        dprintf(fd, "thread %p maybe dead locked\n", this);
+        dprintf(fd, "thread %p may be deadlocked\n", this);
     }
 
     dprintf(fd, "  I/O handle: %d\n", mId);
     dprintf(fd, "  TID: %d\n", getTid());
     dprintf(fd, "  Standby: %s\n", mStandby ? "yes" : "no");
-    dprintf(fd, "  Sample rate: %u\n", mSampleRate);
+    dprintf(fd, "  Sample rate: %u Hz\n", mSampleRate);
     dprintf(fd, "  HAL frame count: %zu\n", mFrameCount);
+    dprintf(fd, "  HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat));
     dprintf(fd, "  HAL buffer size: %u bytes\n", mBufferSize);
-    dprintf(fd, "  Channel Count: %u\n", mChannelCount);
-    dprintf(fd, "  Channel Mask: 0x%08x (%s)\n", mChannelMask,
+    dprintf(fd, "  Channel count: %u\n", mChannelCount);
+    dprintf(fd, "  Channel mask: 0x%08x (%s)\n", mChannelMask,
             channelMaskToString(mChannelMask, mType != RECORD).string());
-    dprintf(fd, "  Format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat));
-    dprintf(fd, "  Frame size: %zu\n", mFrameSize);
+    dprintf(fd, "  Format: 0x%x (%s)\n", mFormat, formatToString(mFormat));
+    dprintf(fd, "  Frame size: %zu bytes\n", mFrameSize);
     dprintf(fd, "  Pending config events:");
     size_t numConfig = mConfigEvents.size();
     if (numConfig) {
@@ -1315,7 +1388,7 @@
 
 void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
 {
-    dprintf(fd, "\nOutput thread %p:\n", this);
+    dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type()));
     dprintf(fd, "  Normal frame count: %zu\n", mNormalFrameCount);
     dprintf(fd, "  Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
     dprintf(fd, "  Total writes: %d\n", mNumWrites);
@@ -1326,6 +1399,10 @@
     dprintf(fd, "  Mixer buffer: %p\n", mMixerBuffer);
     dprintf(fd, "  Effect buffer: %p\n", mEffectBuffer);
     dprintf(fd, "  Fast track availMask=%#x\n", mFastTrackAvailMask);
+    AudioStreamOut *output = mOutput;
+    audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE;
+    String8 flagsAsString = outputFlagsToString(flags);
+    dprintf(fd, "  AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string());
 
     dumpBase(fd, args);
 }
@@ -1861,6 +1938,22 @@
         }
     }
 
+    if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) {
+        // For best precision, we use float instead of the associated output
+        // device format (typically PCM 16 bit).
+
+        mFormat = AUDIO_FORMAT_PCM_FLOAT;
+        mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat);
+        mBufferSize = mFrameSize * mFrameCount;
+
+        // TODO: We currently use the associated output device channel mask and sample rate.
+        // (1) Perhaps use the ORed channel mask of all downstream MixerThreads
+        // (if a valid mask) to avoid premature downmix.
+        // (2) Perhaps use the maximum sample rate of all downstream MixerThreads
+        // instead of the output device sample rate to avoid loss of high frequency information.
+        // This may need to be updated as MixerThread/OutputTracks are added and not here.
+    }
+
     // Calculate size of normal sink buffer relative to the HAL output buffer size
     double multiplier = 1.0;
     if (mType == MIXER && (kUseFastMixer == FastMixer_Static ||
@@ -2137,6 +2230,7 @@
         } else {
             bytesWritten = framesWritten;
         }
+        mLatchDValid = false;
         status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp);
         if (status == NO_ERROR) {
             size_t totalFramesWritten = mNormalSink->framesWritten();
@@ -2640,7 +2734,9 @@
                 }
 
             } else {
+                ATRACE_BEGIN("sleep");
                 usleep(sleepTime);
+                ATRACE_END();
             }
         }
 
@@ -2800,6 +2896,12 @@
             mNormalFrameCount);
     mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
 
+    if (type == DUPLICATING) {
+        // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks
+        // (downstream MixerThreads) in DuplicatingThread::threadLoop_write().
+        // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink.
+        return;
+    }
     // create an NBAIO sink for the HAL output stream, and negotiate
     mOutputSink = new AudioStreamOutSink(output->stream);
     size_t numCounterOffers = 0;
@@ -2841,6 +2943,7 @@
         NBAIO_Format format = mOutputSink->format();
         NBAIO_Format origformat = format;
         // adjust format to match that of the Fast Mixer
+        ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat);
         format.mFormat = fastMixerFormat;
         format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount;
 
@@ -3386,8 +3489,7 @@
         if (sr == mSampleRate) {
             desiredFrames = mNormalFrameCount;
         } else {
-            // +1 for rounding and +1 for additional sample needed for interpolation
-            desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1;
+            desiredFrames = sourceFramesNeeded(sr, mNormalFrameCount, mSampleRate);
             // add frames already consumed but not yet released by the resampler
             // because mAudioTrackServerProxy->framesReady() will include these frames
             desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
@@ -3405,6 +3507,23 @@
         }
 
         size_t framesReady = track->framesReady();
+        if (ATRACE_ENABLED()) {
+            // I wish we had formatted trace names
+            char traceName[16];
+            strcpy(traceName, "nRdy");
+            int name = track->name();
+            if (AudioMixer::TRACK0 <= name &&
+                    name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) {
+                name -= AudioMixer::TRACK0;
+                traceName[4] = (name / 10) + '0';
+                traceName[5] = (name % 10) + '0';
+            } else {
+                traceName[4] = '?';
+                traceName[5] = '?';
+            }
+            traceName[6] = '\0';
+            ATRACE_INT(traceName, framesReady);
+        }
         if ((framesReady >= minFrames) && track->isReady() &&
                 !track->isPaused() && !track->isTerminated())
         {
@@ -4797,16 +4916,8 @@
 
 ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
 {
-    // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT
-    // for delivery downstream as needed. This in-place conversion is safe as
-    // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format
-    // (AUDIO_FORMAT_PCM_8_BIT is not allowed here).
-    if (mFormat != AUDIO_FORMAT_PCM_16_BIT) {
-        memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT,
-                               mSinkBuffer, mFormat, writeFrames * mChannelCount);
-    }
     for (size_t i = 0; i < outputTracks.size(); i++) {
-        outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames);
+        outputTracks[i]->write(mSinkBuffer, writeFrames);
     }
     mStandby = false;
     return (ssize_t)mSinkBufferSize;
@@ -4833,25 +4944,26 @@
 void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
 {
     Mutex::Autolock _l(mLock);
-    // FIXME explain this formula
-    size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
-    // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat
-    // due to current usage case and restrictions on the AudioBufferProvider.
-    // Actual buffer conversion is done in threadLoop_write().
-    //
-    // TODO: This may change in the future, depending on multichannel
-    // (and non int16_t*) support on AF::PlaybackThread::OutputTrack
-    OutputTrack *outputTrack = new OutputTrack(thread,
+    // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass.
+    // Adjust for thread->sampleRate() to determine minimum buffer frame count.
+    // Then triple buffer because Threads do not run synchronously and may not be clock locked.
+    const size_t frameCount =
+            3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate());
+    // TODO: Consider asynchronous sample rate conversion to handle clock disparity
+    // from different OutputTracks and their associated MixerThreads (e.g. one may
+    // nearly empty and the other may be dropping data).
+
+    sp<OutputTrack> outputTrack = new OutputTrack(thread,
                                             this,
                                             mSampleRate,
-                                            AUDIO_FORMAT_PCM_16_BIT,
+                                            mFormat,
                                             mChannelMask,
                                             frameCount,
                                             IPCThreadState::self()->getCallingUid());
     if (outputTrack->cblk() != NULL) {
         thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f);
         mOutputTracks.add(outputTrack);
-        ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
+        ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread);
         updateWaitTime_l();
     }
 }
@@ -5135,7 +5247,9 @@
 
         // sleep with mutex unlocked
         if (sleepUs > 0) {
+            ATRACE_BEGIN("sleep");
             usleep(sleepUs);
+            ATRACE_END();
             sleepUs = 0;
         }
 
@@ -5279,7 +5393,8 @@
                 state->mCommand = FastCaptureState::READ_WRITE;
 #if 0   // FIXME
                 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
-                        FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN);
+                        FastCaptureDumpState::kSamplingNforLowRamDevice :
+                        FastMixerDumpState::kSamplingN);
 #endif
                 didModify = true;
             }
@@ -5427,8 +5542,8 @@
                             upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src,
                                     part1);
                         } else {
-                            downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src,
-                                    part1);
+                            downmix_to_mono_i16_from_stereo_i16((int16_t *)dst,
+                                    (const int16_t *)src, part1);
                         }
                         dst += part1 * activeTrack->mFrameSize;
                         front += part1;
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 1088843..a1ac42c 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -32,6 +32,8 @@
         OFFLOAD             // Thread class is OffloadThread
     };
 
+    static const char *threadTypeToString(type_t type);
+
     ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
                 audio_devices_t outDevice, audio_devices_t inDevice, type_t type);
     virtual             ~ThreadBase();
@@ -406,6 +408,7 @@
                 audio_channel_mask_t    mChannelMask;
                 uint32_t                mChannelCount;
                 size_t                  mFrameSize;
+                // not HAL frame size, this is for output sink (to pipe to fast mixer)
                 audio_format_t          mFormat;           // Source format for Recording and
                                                            // Sink format for Playback.
                                                            // Sink format may be different than
@@ -1167,7 +1170,8 @@
             const sp<MemoryDealer>              mReadOnlyHeap;
 
             // one-time initialization, no locks required
-            sp<FastCapture>                     mFastCapture;   // non-0 if there is also a fast capture
+            sp<FastCapture>                     mFastCapture;   // non-0 if there is also
+                                                                // a fast capture
             // FIXME audio watchdog thread
 
             // contents are not guaranteed to be consistent, no locks required
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index e970036..78cec31 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -20,6 +20,7 @@
 //#define LOG_NDEBUG 0
 
 #include "Configuration.h"
+#include <linux/futex.h>
 #include <math.h>
 #include <sys/syscall.h>
 #include <utils/Log.h>
@@ -859,6 +860,7 @@
         if (mState == FLUSHED) {
             mState = IDLE;
         }
+        mPreviousValid = false;
     }
 }
 
@@ -1709,14 +1711,13 @@
     mActive = false;
 }
 
-bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
+bool AudioFlinger::PlaybackThread::OutputTrack::write(void* data, uint32_t frames)
 {
     Buffer *pInBuffer;
     Buffer inBuffer;
-    uint32_t channelCount = mChannelCount;
     bool outputBufferFull = false;
     inBuffer.frameCount = frames;
-    inBuffer.i16 = data;
+    inBuffer.raw = data;
 
     uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
 
@@ -1726,13 +1727,17 @@
         if (thread != 0) {
             MixerThread *mixerThread = (MixerThread *)thread.get();
             if (mFrameCount > frames) {
+                // For the first write after being inactive, ensure that we have
+                // enough frames to fill mFrameCount (which should be multiples of
+                // the minimum buffer requirements of the downstream MixerThread).
+                // This provides enough frames for the downstream mixer to begin
+                // (see AudioFlinger::PlaybackThread::Track::isReady()).
                 if (mBufferQueue.size() < kMaxOverFlowBuffers) {
                     uint32_t startFrames = (mFrameCount - frames);
                     pInBuffer = new Buffer;
-                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
+                    pInBuffer->mBuffer = calloc(1, startFrames * mFrameSize);
                     pInBuffer->frameCount = startFrames;
-                    pInBuffer->i16 = pInBuffer->mBuffer;
-                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
+                    pInBuffer->raw = pInBuffer->mBuffer;
                     mBufferQueue.add(pInBuffer);
                 } else {
                     ALOGW("OutputTrack::write() %p no more buffers in queue", this);
@@ -1773,20 +1778,20 @@
 
         uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
                 pInBuffer->frameCount;
-        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
+        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * mFrameSize);
         Proxy::Buffer buf;
         buf.mFrameCount = outFrames;
         buf.mRaw = NULL;
         mClientProxy->releaseBuffer(&buf);
         pInBuffer->frameCount -= outFrames;
-        pInBuffer->i16 += outFrames * channelCount;
+        pInBuffer->raw = (int8_t *)pInBuffer->raw + outFrames * mFrameSize;
         mOutBuffer.frameCount -= outFrames;
-        mOutBuffer.i16 += outFrames * channelCount;
+        mOutBuffer.raw = (int8_t *)mOutBuffer.raw + outFrames * mFrameSize;
 
         if (pInBuffer->frameCount == 0) {
             if (mBufferQueue.size()) {
                 mBufferQueue.removeAt(0);
-                delete [] pInBuffer->mBuffer;
+                free(pInBuffer->mBuffer);
                 delete pInBuffer;
                 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
                         mThread.unsafe_get(), mBufferQueue.size());
@@ -1802,11 +1807,10 @@
         if (thread != 0 && !thread->standby()) {
             if (mBufferQueue.size() < kMaxOverFlowBuffers) {
                 pInBuffer = new Buffer;
-                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
+                pInBuffer->mBuffer = malloc(inBuffer.frameCount * mFrameSize);
                 pInBuffer->frameCount = inBuffer.frameCount;
-                pInBuffer->i16 = pInBuffer->mBuffer;
-                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
-                        sizeof(int16_t));
+                pInBuffer->raw = pInBuffer->mBuffer;
+                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * mFrameSize);
                 mBufferQueue.add(pInBuffer);
                 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
                         mThread.unsafe_get(), mBufferQueue.size());
@@ -1817,23 +1821,10 @@
         }
     }
 
-    // Calling write() with a 0 length buffer, means that no more data will be written:
-    // If no more buffers are pending, fill output track buffer to make sure it is started
-    // by output mixer.
-    if (frames == 0 && mBufferQueue.size() == 0) {
-        // FIXME borken, replace by getting framesReady() from proxy
-        size_t user = 0;    // was mCblk->user
-        if (user < mFrameCount) {
-            frames = mFrameCount - user;
-            pInBuffer = new Buffer;
-            pInBuffer->mBuffer = new int16_t[frames * channelCount];
-            pInBuffer->frameCount = frames;
-            pInBuffer->i16 = pInBuffer->mBuffer;
-            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
-            mBufferQueue.add(pInBuffer);
-        } else if (mActive) {
-            stop();
-        }
+    // Calling write() with a 0 length buffer means that no more data will be written:
+    // We rely on stop() to set the appropriate flags to allow the remaining frames to play out.
+    if (frames == 0 && mBufferQueue.size() == 0 && mActive) {
+        stop();
     }
 
     return outputBufferFull;
@@ -1859,7 +1850,7 @@
 
     for (size_t i = 0; i < size; i++) {
         Buffer *pBuffer = mBufferQueue.itemAt(i);
-        delete [] pBuffer->mBuffer;
+        free(pBuffer->mBuffer);
         delete pBuffer;
     }
     mBufferQueue.clear();
diff --git a/services/audioflinger/tests/mixer_to_wav_tests.sh b/services/audioflinger/tests/mixer_to_wav_tests.sh
index 9b39e77..e60e6d5 100755
--- a/services/audioflinger/tests/mixer_to_wav_tests.sh
+++ b/services/audioflinger/tests/mixer_to_wav_tests.sh
@@ -63,8 +63,18 @@
 # process__genericResampling
 # track__Resample / track__genericResample
     adb shell test-mixer $1 -s 48000 \
+        -o /sdcard/tm48000grif.wav \
+        sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000 \
+        sine:f,6,6000,19000  chirp:i,4,30000
+    adb pull /sdcard/tm48000grif.wav $2
+
+# Test:
+# process__genericResampling
+# track__Resample / track__genericResample
+    adb shell test-mixer $1 -s 48000 \
         -o /sdcard/tm48000gr.wav \
-        sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000
+        sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000 \
+        sine:6,6000,19000
     adb pull /sdcard/tm48000gr.wav $2
 
 # Test:
diff --git a/services/audioflinger/tests/test-mixer.cpp b/services/audioflinger/tests/test-mixer.cpp
index 9a4fad6..8da6245 100644
--- a/services/audioflinger/tests/test-mixer.cpp
+++ b/services/audioflinger/tests/test-mixer.cpp
@@ -39,7 +39,7 @@
     fprintf(stderr, "Usage: %s [-f] [-m] [-c channels]"
                     " [-s sample-rate] [-o <output-file>] [-a <aux-buffer-file>] [-P csv]"
                     " (<input-file> | <command>)+\n", name);
-    fprintf(stderr, "    -f    enable floating point input track\n");
+    fprintf(stderr, "    -f    enable floating point input track by default\n");
     fprintf(stderr, "    -m    enable floating point mixer output\n");
     fprintf(stderr, "    -c    number of mixer output channels\n");
     fprintf(stderr, "    -s    mixer sample-rate\n");
@@ -47,8 +47,8 @@
     fprintf(stderr, "    -a    <aux-buffer-file>\n");
     fprintf(stderr, "    -P    # frames provided per call to resample() in CSV format\n");
     fprintf(stderr, "    <input-file> is a WAV file\n");
-    fprintf(stderr, "    <command> can be 'sine:<channels>,<frequency>,<samplerate>'\n");
-    fprintf(stderr, "                     'chirp:<channels>,<samplerate>'\n");
+    fprintf(stderr, "    <command> can be 'sine:[(i|f),]<channels>,<frequency>,<samplerate>'\n");
+    fprintf(stderr, "                     'chirp:[(i|f),]<channels>,<samplerate>'\n");
 }
 
 static int writeFile(const char *filename, const void *buffer,
@@ -78,6 +78,18 @@
     return EXIT_SUCCESS;
 }
 
+const char *parseFormat(const char *s, bool *useFloat) {
+    if (!strncmp(s, "f,", 2)) {
+        *useFloat = true;
+        return s + 2;
+    }
+    if (!strncmp(s, "i,", 2)) {
+        *useFloat = false;
+        return s + 2;
+    }
+    return s;
+}
+
 int main(int argc, char* argv[]) {
     const char* const progname = argv[0];
     bool useInputFloat = false;
@@ -88,8 +100,9 @@
     std::vector<int> Pvalues;
     const char* outputFilename = NULL;
     const char* auxFilename = NULL;
-    std::vector<int32_t> Names;
-    std::vector<SignalProvider> Providers;
+    std::vector<int32_t> names;
+    std::vector<SignalProvider> providers;
+    std::vector<audio_format_t> formats;
 
     for (int ch; (ch = getopt(argc, argv, "fmc:s:o:a:P:")) != -1;) {
         switch (ch) {
@@ -138,54 +151,65 @@
     size_t outputFrames = 0;
 
     // create providers for each track
-    Providers.resize(argc);
+    names.resize(argc);
+    providers.resize(argc);
+    formats.resize(argc);
     for (int i = 0; i < argc; ++i) {
         static const char chirp[] = "chirp:";
         static const char sine[] = "sine:";
         static const double kSeconds = 1;
+        bool useFloat = useInputFloat;
 
         if (!strncmp(argv[i], chirp, strlen(chirp))) {
             std::vector<int> v;
+            const char *s = parseFormat(argv[i] + strlen(chirp), &useFloat);
 
-            parseCSV(argv[i] + strlen(chirp), v);
+            parseCSV(s, v);
             if (v.size() == 2) {
                 printf("creating chirp(%d %d)\n", v[0], v[1]);
-                if (useInputFloat) {
-                    Providers[i].setChirp<float>(v[0], 0, v[1]/2, v[1], kSeconds);
+                if (useFloat) {
+                    providers[i].setChirp<float>(v[0], 0, v[1]/2, v[1], kSeconds);
+                    formats[i] = AUDIO_FORMAT_PCM_FLOAT;
                 } else {
-                    Providers[i].setChirp<int16_t>(v[0], 0, v[1]/2, v[1], kSeconds);
+                    providers[i].setChirp<int16_t>(v[0], 0, v[1]/2, v[1], kSeconds);
+                    formats[i] = AUDIO_FORMAT_PCM_16_BIT;
                 }
-                Providers[i].setIncr(Pvalues);
+                providers[i].setIncr(Pvalues);
             } else {
                 fprintf(stderr, "malformed input '%s'\n", argv[i]);
             }
         } else if (!strncmp(argv[i], sine, strlen(sine))) {
             std::vector<int> v;
+            const char *s = parseFormat(argv[i] + strlen(sine), &useFloat);
 
-            parseCSV(argv[i] + strlen(sine), v);
+            parseCSV(s, v);
             if (v.size() == 3) {
                 printf("creating sine(%d %d %d)\n", v[0], v[1], v[2]);
-                if (useInputFloat) {
-                    Providers[i].setSine<float>(v[0], v[1], v[2], kSeconds);
+                if (useFloat) {
+                    providers[i].setSine<float>(v[0], v[1], v[2], kSeconds);
+                    formats[i] = AUDIO_FORMAT_PCM_FLOAT;
                 } else {
-                    Providers[i].setSine<int16_t>(v[0], v[1], v[2], kSeconds);
+                    providers[i].setSine<int16_t>(v[0], v[1], v[2], kSeconds);
+                    formats[i] = AUDIO_FORMAT_PCM_16_BIT;
                 }
-                Providers[i].setIncr(Pvalues);
+                providers[i].setIncr(Pvalues);
             } else {
                 fprintf(stderr, "malformed input '%s'\n", argv[i]);
             }
         } else {
             printf("creating filename(%s)\n", argv[i]);
             if (useInputFloat) {
-                Providers[i].setFile<float>(argv[i]);
+                providers[i].setFile<float>(argv[i]);
+                formats[i] = AUDIO_FORMAT_PCM_FLOAT;
             } else {
-                Providers[i].setFile<short>(argv[i]);
+                providers[i].setFile<short>(argv[i]);
+                formats[i] = AUDIO_FORMAT_PCM_16_BIT;
             }
-            Providers[i].setIncr(Pvalues);
+            providers[i].setIncr(Pvalues);
         }
         // calculate the number of output frames
-        size_t nframes = (int64_t) Providers[i].getNumFrames() * outputSampleRate
-                / Providers[i].getSampleRate();
+        size_t nframes = (int64_t) providers[i].getNumFrames() * outputSampleRate
+                / providers[i].getSampleRate();
         if (i == 0 || outputFrames > nframes) { // choose minimum for outputFrames
             outputFrames = nframes;
         }
@@ -213,22 +237,20 @@
     // create the mixer.
     const size_t mixerFrameCount = 320; // typical numbers may range from 240 or 960
     AudioMixer *mixer = new AudioMixer(mixerFrameCount, outputSampleRate);
-    audio_format_t inputFormat = useInputFloat
-            ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
     audio_format_t mixerFormat = useMixerFloat
             ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
-    float f = AudioMixer::UNITY_GAIN_FLOAT / Providers.size(); // normalize volume by # tracks
+    float f = AudioMixer::UNITY_GAIN_FLOAT / providers.size(); // normalize volume by # tracks
     static float f0; // zero
 
     // set up the tracks.
-    for (size_t i = 0; i < Providers.size(); ++i) {
-        //printf("track %d out of %d\n", i, Providers.size());
-        uint32_t channelMask = audio_channel_out_mask_from_count(Providers[i].getNumChannels());
+    for (size_t i = 0; i < providers.size(); ++i) {
+        //printf("track %d out of %d\n", i, providers.size());
+        uint32_t channelMask = audio_channel_out_mask_from_count(providers[i].getNumChannels());
         int32_t name = mixer->getTrackName(channelMask,
-                inputFormat, AUDIO_SESSION_OUTPUT_MIX);
+                formats[i], AUDIO_SESSION_OUTPUT_MIX);
         ALOG_ASSERT(name >= 0);
-        Names.push_back(name);
-        mixer->setBufferProvider(name, &Providers[i]);
+        names[i] = name;
+        mixer->setBufferProvider(name, &providers[i]);
         mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
                 (void *)outputAddr);
         mixer->setParameter(
@@ -240,7 +262,7 @@
                 name,
                 AudioMixer::TRACK,
                 AudioMixer::FORMAT,
-                (void *)(uintptr_t)inputFormat);
+                (void *)(uintptr_t)formats[i]);
         mixer->setParameter(
                 name,
                 AudioMixer::TRACK,
@@ -255,7 +277,7 @@
                 name,
                 AudioMixer::RESAMPLE,
                 AudioMixer::SAMPLE_RATE,
-                (void *)(uintptr_t)Providers[i].getSampleRate());
+                (void *)(uintptr_t)providers[i].getSampleRate());
         if (useRamp) {
             mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f0);
             mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f0);
@@ -277,11 +299,11 @@
     // pump the mixer to process data.
     size_t i;
     for (i = 0; i < outputFrames - mixerFrameCount; i += mixerFrameCount) {
-        for (size_t j = 0; j < Names.size(); ++j) {
-            mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
+        for (size_t j = 0; j < names.size(); ++j) {
+            mixer->setParameter(names[j], AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
                     (char *) outputAddr + i * outputFrameSize);
             if (auxFilename) {
-                mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::AUX_BUFFER,
+                mixer->setParameter(names[j], AudioMixer::TRACK, AudioMixer::AUX_BUFFER,
                         (char *) auxAddr + i * auxFrameSize);
             }
         }
diff --git a/services/audiopolicy/Android.mk b/services/audiopolicy/Android.mk
index 188fc89..351ed79 100644
--- a/services/audiopolicy/Android.mk
+++ b/services/audiopolicy/Android.mk
@@ -3,19 +3,19 @@
 include $(CLEAR_VARS)
 
 LOCAL_SRC_FILES:= \
-    AudioPolicyService.cpp \
-    AudioPolicyEffects.cpp
+    service/AudioPolicyService.cpp \
+    service/AudioPolicyEffects.cpp
 
 ifeq ($(USE_LEGACY_AUDIO_POLICY), 1)
 LOCAL_SRC_FILES += \
-    AudioPolicyInterfaceImplLegacy.cpp \
-    AudioPolicyClientImplLegacy.cpp
+    service/AudioPolicyInterfaceImplLegacy.cpp \
+    service/AudioPolicyClientImplLegacy.cpp
 
     LOCAL_CFLAGS += -DUSE_LEGACY_AUDIO_POLICY
 else
 LOCAL_SRC_FILES += \
-    AudioPolicyInterfaceImpl.cpp \
-    AudioPolicyClientImpl.cpp
+    service/AudioPolicyInterfaceImpl.cpp \
+    service/AudioPolicyClientImpl.cpp
 endif
 
 LOCAL_C_INCLUDES := \
@@ -53,7 +53,15 @@
 include $(CLEAR_VARS)
 
 LOCAL_SRC_FILES:= \
-    AudioPolicyManager.cpp
+    managerdefault/AudioPolicyManager.cpp \
+    managerdefault/ConfigParsingUtils.cpp \
+    managerdefault/Devices.cpp \
+    managerdefault/Gains.cpp \
+    managerdefault/HwModule.cpp \
+    managerdefault/IOProfile.cpp \
+    managerdefault/Ports.cpp \
+    managerdefault/AudioInputDescriptor.cpp \
+    managerdefault/AudioOutputDescriptor.cpp
 
 LOCAL_SHARED_LIBRARIES := \
     libcutils \
@@ -73,7 +81,7 @@
 include $(CLEAR_VARS)
 
 LOCAL_SRC_FILES:= \
-    AudioPolicyFactory.cpp
+    manager/AudioPolicyFactory.cpp
 
 LOCAL_SHARED_LIBRARIES := \
     libaudiopolicymanagerdefault
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 4508fa7..116d0d6 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -75,7 +75,8 @@
     // indicate a change in device connection status
     virtual status_t setDeviceConnectionState(audio_devices_t device,
                                               audio_policy_dev_state_t state,
-                                          const char *device_address) = 0;
+                                              const char *device_address,
+                                              const char *device_name) = 0;
     // retrieve a device connection status
     virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
                                                                           const char *device_address) = 0;
diff --git a/services/audiopolicy/AudioPolicyManager.h b/services/audiopolicy/AudioPolicyManager.h
deleted file mode 100644
index cbdafa6..0000000
--- a/services/audiopolicy/AudioPolicyManager.h
+++ /dev/null
@@ -1,937 +0,0 @@
-/*
- * Copyright (C) 2009 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 <stdint.h>
-#include <sys/types.h>
-#include <cutils/config_utils.h>
-#include <cutils/misc.h>
-#include <utils/Timers.h>
-#include <utils/Errors.h>
-#include <utils/KeyedVector.h>
-#include <utils/SortedVector.h>
-#include <media/AudioPolicy.h>
-#include "AudioPolicyInterface.h"
-
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
-#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5
-// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB
-#define SONIFICATION_HEADSET_VOLUME_MIN  0.016
-// Time in milliseconds during which we consider that music is still active after a music
-// track was stopped - see computeVolume()
-#define SONIFICATION_HEADSET_MUSIC_DELAY  5000
-// Time in milliseconds after media stopped playing during which we consider that the
-// sonification should be as unobtrusive as during the time media was playing.
-#define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000
-// Time in milliseconds during witch some streams are muted while the audio path
-// is switched
-#define MUTE_TIME_MS 2000
-
-#define NUM_TEST_OUTPUTS 5
-
-#define NUM_VOL_CURVE_KNEES 2
-
-// Default minimum length allowed for offloading a compressed track
-// Can be overridden by the audio.offload.min.duration.secs property
-#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60
-
-#define MAX_MIXER_SAMPLING_RATE 48000
-#define MAX_MIXER_CHANNEL_COUNT 8
-
-// ----------------------------------------------------------------------------
-// AudioPolicyManager implements audio policy manager behavior common to all platforms.
-// ----------------------------------------------------------------------------
-
-class AudioPolicyManager: public AudioPolicyInterface
-#ifdef AUDIO_POLICY_TEST
-    , public Thread
-#endif //AUDIO_POLICY_TEST
-{
-
-public:
-                AudioPolicyManager(AudioPolicyClientInterface *clientInterface);
-        virtual ~AudioPolicyManager();
-
-        // AudioPolicyInterface
-        virtual status_t setDeviceConnectionState(audio_devices_t device,
-                                                          audio_policy_dev_state_t state,
-                                                          const char *device_address);
-        virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
-                                                                              const char *device_address);
-        virtual void setPhoneState(audio_mode_t state);
-        virtual void setForceUse(audio_policy_force_use_t usage,
-                                 audio_policy_forced_cfg_t config);
-        virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
-        virtual void setSystemProperty(const char* property, const char* value);
-        virtual status_t initCheck();
-        virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
-                                            uint32_t samplingRate,
-                                            audio_format_t format,
-                                            audio_channel_mask_t channelMask,
-                                            audio_output_flags_t flags,
-                                            const audio_offload_info_t *offloadInfo);
-        virtual status_t getOutputForAttr(const audio_attributes_t *attr,
-                                          audio_io_handle_t *output,
-                                          audio_session_t session,
-                                          audio_stream_type_t *stream,
-                                          uint32_t samplingRate,
-                                          audio_format_t format,
-                                          audio_channel_mask_t channelMask,
-                                          audio_output_flags_t flags,
-                                          const audio_offload_info_t *offloadInfo);
-        virtual status_t startOutput(audio_io_handle_t output,
-                                     audio_stream_type_t stream,
-                                     audio_session_t session);
-        virtual status_t stopOutput(audio_io_handle_t output,
-                                    audio_stream_type_t stream,
-                                    audio_session_t session);
-        virtual void releaseOutput(audio_io_handle_t output,
-                                   audio_stream_type_t stream,
-                                   audio_session_t session);
-        virtual status_t getInputForAttr(const audio_attributes_t *attr,
-                                         audio_io_handle_t *input,
-                                         audio_session_t session,
-                                         uint32_t samplingRate,
-                                         audio_format_t format,
-                                         audio_channel_mask_t channelMask,
-                                         audio_input_flags_t flags,
-                                         input_type_t *inputType);
-
-        // indicates to the audio policy manager that the input starts being used.
-        virtual status_t startInput(audio_io_handle_t input,
-                                    audio_session_t session);
-
-        // indicates to the audio policy manager that the input stops being used.
-        virtual status_t stopInput(audio_io_handle_t input,
-                                   audio_session_t session);
-        virtual void releaseInput(audio_io_handle_t input,
-                                  audio_session_t session);
-        virtual void closeAllInputs();
-        virtual void initStreamVolume(audio_stream_type_t stream,
-                                                    int indexMin,
-                                                    int indexMax);
-        virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
-                                              int index,
-                                              audio_devices_t device);
-        virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
-                                              int *index,
-                                              audio_devices_t device);
-
-        // return the strategy corresponding to a given stream type
-        virtual uint32_t getStrategyForStream(audio_stream_type_t stream);
-        // return the strategy corresponding to the given audio attributes
-        virtual uint32_t getStrategyForAttr(const audio_attributes_t *attr);
-
-        // return the enabled output devices for the given stream type
-        virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream);
-
-        virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL);
-        virtual status_t registerEffect(const effect_descriptor_t *desc,
-                                        audio_io_handle_t io,
-                                        uint32_t strategy,
-                                        int session,
-                                        int id);
-        virtual status_t unregisterEffect(int id);
-        virtual status_t setEffectEnabled(int id, bool enabled);
-
-        virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
-        // return whether a stream is playing remotely, override to change the definition of
-        //   local/remote playback, used for instance by notification manager to not make
-        //   media players lose audio focus when not playing locally
-        //   For the base implementation, "remotely" means playing during screen mirroring which
-        //   uses an output for playback with a non-empty, non "0" address.
-        virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
-        virtual bool isSourceActive(audio_source_t source) const;
-
-        virtual status_t dump(int fd);
-
-        virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
-
-        virtual status_t listAudioPorts(audio_port_role_t role,
-                                        audio_port_type_t type,
-                                        unsigned int *num_ports,
-                                        struct audio_port *ports,
-                                        unsigned int *generation);
-        virtual status_t getAudioPort(struct audio_port *port);
-        virtual status_t createAudioPatch(const struct audio_patch *patch,
-                                           audio_patch_handle_t *handle,
-                                           uid_t uid);
-        virtual status_t releaseAudioPatch(audio_patch_handle_t handle,
-                                              uid_t uid);
-        virtual status_t listAudioPatches(unsigned int *num_patches,
-                                          struct audio_patch *patches,
-                                          unsigned int *generation);
-        virtual status_t setAudioPortConfig(const struct audio_port_config *config);
-        virtual void clearAudioPatches(uid_t uid);
-
-        virtual status_t acquireSoundTriggerSession(audio_session_t *session,
-                                               audio_io_handle_t *ioHandle,
-                                               audio_devices_t *device);
-
-        virtual status_t releaseSoundTriggerSession(audio_session_t session);
-
-        virtual status_t registerPolicyMixes(Vector<AudioMix> mixes);
-        virtual status_t unregisterPolicyMixes(Vector<AudioMix> mixes);
-
-protected:
-
-        enum routing_strategy {
-            STRATEGY_MEDIA,
-            STRATEGY_PHONE,
-            STRATEGY_SONIFICATION,
-            STRATEGY_SONIFICATION_RESPECTFUL,
-            STRATEGY_DTMF,
-            STRATEGY_ENFORCED_AUDIBLE,
-            STRATEGY_TRANSMITTED_THROUGH_SPEAKER,
-            STRATEGY_ACCESSIBILITY,
-            STRATEGY_REROUTING,
-            NUM_STRATEGIES
-        };
-
-        // 4 points to define the volume attenuation curve, each characterized by the volume
-        // index (from 0 to 100) at which they apply, and the attenuation in dB at that index.
-        // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl()
-
-        enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4};
-
-        class VolumeCurvePoint
-        {
-        public:
-            int mIndex;
-            float mDBAttenuation;
-        };
-
-        // device categories used for volume curve management.
-        enum device_category {
-            DEVICE_CATEGORY_HEADSET,
-            DEVICE_CATEGORY_SPEAKER,
-            DEVICE_CATEGORY_EARPIECE,
-            DEVICE_CATEGORY_EXT_MEDIA,
-            DEVICE_CATEGORY_CNT
-        };
-
-        class HwModule;
-
-        class AudioGain: public RefBase
-        {
-        public:
-            AudioGain(int index, bool useInChannelMask);
-            virtual ~AudioGain() {}
-
-            void dump(int fd, int spaces, int index) const;
-
-            void getDefaultConfig(struct audio_gain_config *config);
-            status_t checkConfig(const struct audio_gain_config *config);
-            int               mIndex;
-            struct audio_gain mGain;
-            bool              mUseInChannelMask;
-        };
-
-        class AudioPort: public virtual RefBase
-        {
-        public:
-            AudioPort(const String8& name, audio_port_type_t type,
-                      audio_port_role_t role, const sp<HwModule>& module);
-            virtual ~AudioPort() {}
-
-            virtual void toAudioPort(struct audio_port *port) const;
-
-            void importAudioPort(const sp<AudioPort> port);
-            void clearCapabilities();
-
-            void loadSamplingRates(char *name);
-            void loadFormats(char *name);
-            void loadOutChannels(char *name);
-            void loadInChannels(char *name);
-
-            audio_gain_mode_t loadGainMode(char *name);
-            void loadGain(cnode *root, int index);
-            virtual void loadGains(cnode *root);
-
-            // searches for an exact match
-            status_t checkExactSamplingRate(uint32_t samplingRate) const;
-            // searches for a compatible match, and returns the best match via updatedSamplingRate
-            status_t checkCompatibleSamplingRate(uint32_t samplingRate,
-                    uint32_t *updatedSamplingRate) const;
-            // searches for an exact match
-            status_t checkExactChannelMask(audio_channel_mask_t channelMask) const;
-            // searches for a compatible match, currently implemented for input channel masks only
-            status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask) const;
-            status_t checkFormat(audio_format_t format) const;
-            status_t checkGain(const struct audio_gain_config *gainConfig, int index) const;
-
-            uint32_t pickSamplingRate() const;
-            audio_channel_mask_t pickChannelMask() const;
-            audio_format_t pickFormat() const;
-
-            static const audio_format_t sPcmFormatCompareTable[];
-            static int compareFormats(audio_format_t format1, audio_format_t format2);
-
-            void dump(int fd, int spaces) const;
-
-            String8           mName;
-            audio_port_type_t mType;
-            audio_port_role_t mRole;
-            bool              mUseInChannelMask;
-            // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats
-            // indicates the supported parameters should be read from the output stream
-            // after it is opened for the first time
-            Vector <uint32_t> mSamplingRates; // supported sampling rates
-            Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks
-            Vector <audio_format_t> mFormats; // supported audio formats
-            Vector < sp<AudioGain> > mGains; // gain controllers
-            sp<HwModule> mModule;                 // audio HW module exposing this I/O stream
-            uint32_t mFlags; // attribute flags (e.g primary output,
-                                                // direct output...).
-        };
-
-        class AudioPortConfig: public virtual RefBase
-        {
-        public:
-            AudioPortConfig();
-            virtual ~AudioPortConfig() {}
-
-            status_t applyAudioPortConfig(const struct audio_port_config *config,
-                                          struct audio_port_config *backupConfig = NULL);
-            virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
-                                   const struct audio_port_config *srcConfig = NULL) const = 0;
-            virtual sp<AudioPort> getAudioPort() const = 0;
-            uint32_t mSamplingRate;
-            audio_format_t mFormat;
-            audio_channel_mask_t mChannelMask;
-            struct audio_gain_config mGain;
-        };
-
-
-        class AudioPatch: public RefBase
-        {
-        public:
-            AudioPatch(audio_patch_handle_t handle,
-                       const struct audio_patch *patch, uid_t uid) :
-                           mHandle(handle), mPatch(*patch), mUid(uid), mAfPatchHandle(0) {}
-
-            status_t dump(int fd, int spaces, int index) const;
-
-            audio_patch_handle_t mHandle;
-            struct audio_patch mPatch;
-            uid_t mUid;
-            audio_patch_handle_t mAfPatchHandle;
-        };
-
-        class DeviceDescriptor: public AudioPort, public AudioPortConfig
-        {
-        public:
-            DeviceDescriptor(const String8& name, audio_devices_t type);
-
-            virtual ~DeviceDescriptor() {}
-
-            bool equals(const sp<DeviceDescriptor>& other) const;
-
-            // AudioPortConfig
-            virtual sp<AudioPort> getAudioPort() const { return (AudioPort*) this; }
-            virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
-                                   const struct audio_port_config *srcConfig = NULL) const;
-
-            // AudioPort
-            virtual void loadGains(cnode *root);
-            virtual void toAudioPort(struct audio_port *port) const;
-
-            status_t dump(int fd, int spaces, int index) const;
-
-            audio_devices_t mDeviceType;
-            String8 mAddress;
-            audio_port_handle_t mId;
-        };
-
-        class DeviceVector : public SortedVector< sp<DeviceDescriptor> >
-        {
-        public:
-            DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {}
-
-            ssize_t         add(const sp<DeviceDescriptor>& item);
-            ssize_t         remove(const sp<DeviceDescriptor>& item);
-            ssize_t         indexOf(const sp<DeviceDescriptor>& item) const;
-
-            audio_devices_t types() const { return mDeviceTypes; }
-
-            void loadDevicesFromType(audio_devices_t types);
-            void loadDevicesFromName(char *name, const DeviceVector& declaredDevices);
-
-            sp<DeviceDescriptor> getDevice(audio_devices_t type, String8 address) const;
-            DeviceVector getDevicesFromType(audio_devices_t types) const;
-            sp<DeviceDescriptor> getDeviceFromId(audio_port_handle_t id) const;
-            sp<DeviceDescriptor> getDeviceFromName(const String8& name) const;
-            DeviceVector getDevicesFromTypeAddr(audio_devices_t type, String8 address)
-                    const;
-
-        private:
-            void refreshTypes();
-            audio_devices_t mDeviceTypes;
-        };
-
-        // the IOProfile class describes the capabilities of an output or input stream.
-        // It is currently assumed that all combination of listed parameters are supported.
-        // It is used by the policy manager to determine if an output or input is suitable for
-        // a given use case,  open/close it accordingly and connect/disconnect audio tracks
-        // to/from it.
-        class IOProfile : public AudioPort
-        {
-        public:
-            IOProfile(const String8& name, audio_port_role_t role, const sp<HwModule>& module);
-            virtual ~IOProfile();
-
-            // This method is used for both output and input.
-            // If parameter updatedSamplingRate is non-NULL, it is assigned the actual sample rate.
-            // For input, flags is interpreted as audio_input_flags_t.
-            // TODO: merge audio_output_flags_t and audio_input_flags_t.
-            bool isCompatibleProfile(audio_devices_t device,
-                                     String8 address,
-                                     uint32_t samplingRate,
-                                     uint32_t *updatedSamplingRate,
-                                     audio_format_t format,
-                                     audio_channel_mask_t channelMask,
-                                     uint32_t flags) const;
-
-            void dump(int fd);
-            void log();
-
-            DeviceVector  mSupportedDevices; // supported devices
-                                             // (devices this output can be routed to)
-        };
-
-        class HwModule : public RefBase
-        {
-        public:
-                    HwModule(const char *name);
-                    ~HwModule();
-
-            status_t loadOutput(cnode *root);
-            status_t loadInput(cnode *root);
-            status_t loadDevice(cnode *root);
-
-            status_t addOutputProfile(String8 name, const audio_config_t *config,
-                                      audio_devices_t device, String8 address);
-            status_t removeOutputProfile(String8 name);
-            status_t addInputProfile(String8 name, const audio_config_t *config,
-                                      audio_devices_t device, String8 address);
-            status_t removeInputProfile(String8 name);
-
-            void dump(int fd);
-
-            const char *const        mName; // base name of the audio HW module (primary, a2dp ...)
-            uint32_t                 mHalVersion; // audio HAL API version
-            audio_module_handle_t    mHandle;
-            Vector < sp<IOProfile> > mOutputProfiles; // output profiles exposed by this module
-            Vector < sp<IOProfile> > mInputProfiles;  // input profiles exposed by this module
-            DeviceVector             mDeclaredDevices; // devices declared in audio_policy.conf
-
-        };
-
-        // default volume curve
-        static const VolumeCurvePoint sDefaultVolumeCurve[AudioPolicyManager::VOLCNT];
-        // default volume curve for media strategy
-        static const VolumeCurvePoint sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT];
-        // volume curve for non-media audio on ext media outputs (HDMI, Line, etc)
-        static const VolumeCurvePoint sExtMediaSystemVolumeCurve[AudioPolicyManager::VOLCNT];
-        // volume curve for media strategy on speakers
-        static const VolumeCurvePoint sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT];
-        static const VolumeCurvePoint sSpeakerMediaVolumeCurveDrc[AudioPolicyManager::VOLCNT];
-        // volume curve for sonification strategy on speakers
-        static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT];
-        static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT];
-        static const VolumeCurvePoint sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT];
-        static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT];
-        static const VolumeCurvePoint sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT];
-        static const VolumeCurvePoint sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT];
-        static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT];
-        static const VolumeCurvePoint sLinearVolumeCurve[AudioPolicyManager::VOLCNT];
-        static const VolumeCurvePoint sSilentVolumeCurve[AudioPolicyManager::VOLCNT];
-        static const VolumeCurvePoint sFullScaleVolumeCurve[AudioPolicyManager::VOLCNT];
-        // default volume curves per stream and device category. See initializeVolumeCurves()
-        static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][DEVICE_CATEGORY_CNT];
-
-        // descriptor for audio outputs. Used to maintain current configuration of each opened audio output
-        // and keep track of the usage of this output by each audio stream type.
-        class AudioOutputDescriptor: public AudioPortConfig
-        {
-        public:
-            AudioOutputDescriptor(const sp<IOProfile>& profile);
-
-            status_t    dump(int fd);
-
-            audio_devices_t device() const;
-            void changeRefCount(audio_stream_type_t stream, int delta);
-
-            bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
-            audio_devices_t supportedDevices();
-            uint32_t latency();
-            bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc);
-            bool isActive(uint32_t inPastMs = 0) const;
-            bool isStreamActive(audio_stream_type_t stream,
-                                uint32_t inPastMs = 0,
-                                nsecs_t sysTime = 0) const;
-            bool isStrategyActive(routing_strategy strategy,
-                             uint32_t inPastMs = 0,
-                             nsecs_t sysTime = 0) const;
-
-            virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
-                                   const struct audio_port_config *srcConfig = NULL) const;
-            virtual sp<AudioPort> getAudioPort() const { return mProfile; }
-            void toAudioPort(struct audio_port *port) const;
-
-            audio_port_handle_t mId;
-            audio_io_handle_t mIoHandle;              // output handle
-            uint32_t mLatency;                  //
-            audio_output_flags_t mFlags;   //
-            audio_devices_t mDevice;                   // current device this output is routed to
-            AudioMix *mPolicyMix;             // non NULL when used by a dynamic policy
-            audio_patch_handle_t mPatchHandle;
-            uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output
-            nsecs_t mStopTime[AUDIO_STREAM_CNT];
-            sp<AudioOutputDescriptor> mOutput1;    // used by duplicated outputs: first output
-            sp<AudioOutputDescriptor> mOutput2;    // used by duplicated outputs: second output
-            float mCurVolume[AUDIO_STREAM_CNT];   // current stream volume
-            int mMuteCount[AUDIO_STREAM_CNT];     // mute request counter
-            const sp<IOProfile> mProfile;          // I/O profile this output derives from
-            bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible
-                                                // device selection. See checkDeviceMuteStrategies()
-            uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
-        };
-
-        // descriptor for audio inputs. Used to maintain current configuration of each opened audio input
-        // and keep track of the usage of this input.
-        class AudioInputDescriptor: public AudioPortConfig
-        {
-        public:
-            AudioInputDescriptor(const sp<IOProfile>& profile);
-
-            status_t    dump(int fd);
-
-            audio_port_handle_t           mId;
-            audio_io_handle_t             mIoHandle;       // input handle
-            audio_devices_t               mDevice;         // current device this input is routed to
-            AudioMix                      *mPolicyMix;     // non NULL when used by a dynamic policy
-            audio_patch_handle_t          mPatchHandle;
-            uint32_t                      mRefCount;       // number of AudioRecord clients using
-                                                           // this input
-            uint32_t                      mOpenRefCount;
-            audio_source_t                mInputSource;    // input source selected by application
-                                                           //(mediarecorder.h)
-            const sp<IOProfile>           mProfile;        // I/O profile this output derives from
-            SortedVector<audio_session_t> mSessions;       // audio sessions attached to this input
-            bool                          mIsSoundTrigger; // used by a soundtrigger capture
-
-            virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
-                                   const struct audio_port_config *srcConfig = NULL) const;
-            virtual sp<AudioPort> getAudioPort() const { return mProfile; }
-            void toAudioPort(struct audio_port *port) const;
-        };
-
-        // stream descriptor used for volume control
-        class StreamDescriptor
-        {
-        public:
-            StreamDescriptor();
-
-            int getVolumeIndex(audio_devices_t device);
-            void dump(int fd);
-
-            int mIndexMin;      // min volume index
-            int mIndexMax;      // max volume index
-            KeyedVector<audio_devices_t, int> mIndexCur;   // current volume index per device
-            bool mCanBeMuted;   // true is the stream can be muted
-
-            const VolumeCurvePoint *mVolumeCurve[DEVICE_CATEGORY_CNT];
-        };
-
-        // stream descriptor used for volume control
-        class EffectDescriptor : public RefBase
-        {
-        public:
-
-            status_t dump(int fd);
-
-            int mIo;                // io the effect is attached to
-            routing_strategy mStrategy; // routing strategy the effect is associated to
-            int mSession;               // audio session the effect is on
-            effect_descriptor_t mDesc;  // effect descriptor
-            bool mEnabled;              // enabled state: CPU load being used or not
-        };
-
-        void addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc);
-        void addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc);
-
-        // return the strategy corresponding to a given stream type
-        static routing_strategy getStrategy(audio_stream_type_t stream);
-
-        // return appropriate device for streams handled by the specified strategy according to current
-        // phone state, connected devices...
-        // if fromCache is true, the device is returned from mDeviceForStrategy[],
-        // otherwise it is determine by current state
-        // (device connected,phone state, force use, a2dp output...)
-        // This allows to:
-        //  1 speed up process when the state is stable (when starting or stopping an output)
-        //  2 access to either current device selection (fromCache == true) or
-        // "future" device selection (fromCache == false) when called from a context
-        //  where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
-        //  before updateDevicesAndOutputs() is called.
-        virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy,
-                                                     bool fromCache);
-
-        // change the route of the specified output. Returns the number of ms we have slept to
-        // allow new routing to take effect in certain cases.
-        virtual uint32_t setOutputDevice(audio_io_handle_t output,
-                             audio_devices_t device,
-                             bool force = false,
-                             int delayMs = 0,
-                             audio_patch_handle_t *patchHandle = NULL,
-                             const char* address = NULL);
-        status_t resetOutputDevice(audio_io_handle_t output,
-                                   int delayMs = 0,
-                                   audio_patch_handle_t *patchHandle = NULL);
-        status_t setInputDevice(audio_io_handle_t input,
-                                audio_devices_t device,
-                                bool force = false,
-                                audio_patch_handle_t *patchHandle = NULL);
-        status_t resetInputDevice(audio_io_handle_t input,
-                                  audio_patch_handle_t *patchHandle = NULL);
-
-        // select input device corresponding to requested audio source
-        virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource);
-
-        // return io handle of active input or 0 if no input is active
-        //    Only considers inputs from physical devices (e.g. main mic, headset mic) when
-        //    ignoreVirtualInputs is true.
-        audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true);
-
-        uint32_t activeInputsCount() const;
-
-        // initialize volume curves for each strategy and device category
-        void initializeVolumeCurves();
-
-        // compute the actual volume for a given stream according to the requested index and a particular
-        // device
-        virtual float computeVolume(audio_stream_type_t stream, int index,
-                                    audio_io_handle_t output, audio_devices_t device);
-
-        // check that volume change is permitted, compute and send new volume to audio hardware
-        virtual status_t checkAndSetVolume(audio_stream_type_t stream, int index,
-                                           audio_io_handle_t output,
-                                           audio_devices_t device,
-                                           int delayMs = 0, bool force = false);
-
-        // apply all stream volumes to the specified output and device
-        void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
-
-        // Mute or unmute all streams handled by the specified strategy on the specified output
-        void setStrategyMute(routing_strategy strategy,
-                             bool on,
-                             audio_io_handle_t output,
-                             int delayMs = 0,
-                             audio_devices_t device = (audio_devices_t)0);
-
-        // Mute or unmute the stream on the specified output
-        void setStreamMute(audio_stream_type_t stream,
-                           bool on,
-                           audio_io_handle_t output,
-                           int delayMs = 0,
-                           audio_devices_t device = (audio_devices_t)0);
-
-        // handle special cases for sonification strategy while in call: mute streams or replace by
-        // a special tone in the device used for communication
-        void handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange);
-
-        // true if device is in a telephony or VoIP call
-        virtual bool isInCall();
-
-        // true if given state represents a device in a telephony or VoIP call
-        virtual bool isStateInCall(int state);
-
-        // when a device is connected, checks if an open output can be routed
-        // to this device. If none is open, tries to open one of the available outputs.
-        // Returns an output suitable to this device or 0.
-        // when a device is disconnected, checks if an output is not used any more and
-        // returns its handle if any.
-        // transfers the audio tracks and effects from one output thread to another accordingly.
-        status_t checkOutputsForDevice(const sp<DeviceDescriptor> devDesc,
-                                       audio_policy_dev_state_t state,
-                                       SortedVector<audio_io_handle_t>& outputs,
-                                       const String8 address);
-
-        status_t checkInputsForDevice(audio_devices_t device,
-                                      audio_policy_dev_state_t state,
-                                      SortedVector<audio_io_handle_t>& inputs,
-                                      const String8 address);
-
-        // close an output and its companion duplicating output.
-        void closeOutput(audio_io_handle_t output);
-
-        // close an input.
-        void closeInput(audio_io_handle_t input);
-
-        // checks and if necessary changes outputs used for all strategies.
-        // must be called every time a condition that affects the output choice for a given strategy
-        // changes: connected device, phone state, force use...
-        // Must be called before updateDevicesAndOutputs()
-        void checkOutputForStrategy(routing_strategy strategy);
-
-        // Same as checkOutputForStrategy() but for a all strategies in order of priority
-        void checkOutputForAllStrategies();
-
-        // manages A2DP output suspend/restore according to phone state and BT SCO usage
-        void checkA2dpSuspend();
-
-        // returns the A2DP output handle if it is open or 0 otherwise
-        audio_io_handle_t getA2dpOutput();
-
-        // selects the most appropriate device on output for current state
-        // must be called every time a condition that affects the device choice for a given output is
-        // changed: connected device, phone state, force use, output start, output stop..
-        // see getDeviceForStrategy() for the use of fromCache parameter
-        audio_devices_t getNewOutputDevice(audio_io_handle_t output, bool fromCache);
-
-        // updates cache of device used by all strategies (mDeviceForStrategy[])
-        // must be called every time a condition that affects the device choice for a given strategy is
-        // changed: connected device, phone state, force use...
-        // cached values are used by getDeviceForStrategy() if parameter fromCache is true.
-         // Must be called after checkOutputForAllStrategies()
-        void updateDevicesAndOutputs();
-
-        // selects the most appropriate device on input for current state
-        audio_devices_t getNewInputDevice(audio_io_handle_t input);
-
-        virtual uint32_t getMaxEffectsCpuLoad();
-        virtual uint32_t getMaxEffectsMemory();
-#ifdef AUDIO_POLICY_TEST
-        virtual     bool        threadLoop();
-                    void        exit();
-        int testOutputIndex(audio_io_handle_t output);
-#endif //AUDIO_POLICY_TEST
-
-        status_t setEffectEnabled(const sp<EffectDescriptor>& effectDesc, bool enabled);
-
-        // returns the category the device belongs to with regard to volume curve management
-        static device_category getDeviceCategory(audio_devices_t device);
-
-        // extract one device relevant for volume control from multiple device selection
-        static audio_devices_t getDeviceForVolume(audio_devices_t device);
-
-        SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
-                        DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > openOutputs);
-        bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
-                                           SortedVector<audio_io_handle_t>& outputs2);
-
-        // mute/unmute strategies using an incompatible device combination
-        // if muting, wait for the audio in pcm buffer to be drained before proceeding
-        // if unmuting, unmute only after the specified delay
-        // Returns the number of ms waited
-        virtual uint32_t  checkDeviceMuteStrategies(sp<AudioOutputDescriptor> outputDesc,
-                                            audio_devices_t prevDevice,
-                                            uint32_t delayMs);
-
-        audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
-                                       audio_output_flags_t flags,
-                                       audio_format_t format);
-        // samplingRate parameter is an in/out and so may be modified
-        sp<IOProfile> getInputProfile(audio_devices_t device,
-                                      String8 address,
-                                      uint32_t& samplingRate,
-                                      audio_format_t format,
-                                      audio_channel_mask_t channelMask,
-                                      audio_input_flags_t flags);
-        sp<IOProfile> getProfileForDirectOutput(audio_devices_t device,
-                                                       uint32_t samplingRate,
-                                                       audio_format_t format,
-                                                       audio_channel_mask_t channelMask,
-                                                       audio_output_flags_t flags);
-
-        audio_io_handle_t selectOutputForEffects(const SortedVector<audio_io_handle_t>& outputs);
-
-        bool isNonOffloadableEffectEnabled();
-
-        virtual status_t addAudioPatch(audio_patch_handle_t handle,
-                               const sp<AudioPatch>& patch);
-        virtual status_t removeAudioPatch(audio_patch_handle_t handle);
-
-        sp<AudioOutputDescriptor> getOutputFromId(audio_port_handle_t id) const;
-        sp<AudioInputDescriptor> getInputFromId(audio_port_handle_t id) const;
-        sp<HwModule> getModuleForDevice(audio_devices_t device) const;
-        sp<HwModule> getModuleFromName(const char *name) const;
-        audio_devices_t availablePrimaryOutputDevices();
-        audio_devices_t availablePrimaryInputDevices();
-
-        void updateCallRouting(audio_devices_t rxDevice, int delayMs = 0);
-
-        //
-        // Audio policy configuration file parsing (audio_policy.conf)
-        //
-        static uint32_t stringToEnum(const struct StringToEnum *table,
-                                     size_t size,
-                                     const char *name);
-        static const char *enumToString(const struct StringToEnum *table,
-                                      size_t size,
-                                      uint32_t value);
-        static bool stringToBool(const char *value);
-        static uint32_t parseOutputFlagNames(char *name);
-        static uint32_t parseInputFlagNames(char *name);
-        static audio_devices_t parseDeviceNames(char *name);
-        void loadHwModule(cnode *root);
-        void loadHwModules(cnode *root);
-        void loadGlobalConfig(cnode *root, const sp<HwModule>& module);
-        status_t loadAudioPolicyConfig(const char *path);
-        void defaultAudioPolicyConfig(void);
-
-
-        uid_t mUidCached;
-        AudioPolicyClientInterface *mpClientInterface;  // audio policy client interface
-        audio_io_handle_t mPrimaryOutput;              // primary output handle
-        // list of descriptors for outputs currently opened
-        DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mOutputs;
-        // copy of mOutputs before setDeviceConnectionState() opens new outputs
-        // reset to mOutputs when updateDevicesAndOutputs() is called.
-        DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mPreviousOutputs;
-        DefaultKeyedVector<audio_io_handle_t, sp<AudioInputDescriptor> > mInputs;     // list of input descriptors
-        DeviceVector  mAvailableOutputDevices; // all available output devices
-        DeviceVector  mAvailableInputDevices;  // all available input devices
-        int mPhoneState;                                                    // current phone state
-        audio_policy_forced_cfg_t mForceUse[AUDIO_POLICY_FORCE_USE_CNT];   // current forced use configuration
-
-        StreamDescriptor mStreams[AUDIO_STREAM_CNT];           // stream descriptors for volume control
-        bool    mLimitRingtoneVolume;                                       // limit ringtone volume to music volume if headset connected
-        audio_devices_t mDeviceForStrategy[NUM_STRATEGIES];
-        float   mLastVoiceVolume;                                           // last voice volume value sent to audio HAL
-
-        // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units
-        static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000;
-        // Maximum memory allocated to audio effects in KB
-        static const uint32_t MAX_EFFECTS_MEMORY = 512;
-        uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects
-        uint32_t mTotalEffectsMemory;  // current memory used by effects
-        KeyedVector<int, sp<EffectDescriptor> > mEffects;  // list of registered audio effects
-        bool    mA2dpSuspended;  // true if A2DP output is suspended
-        sp<DeviceDescriptor> mDefaultOutputDevice; // output device selected by default at boot time
-        bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path
-                                // to boost soft sounds, used to adjust volume curves accordingly
-
-        Vector < sp<HwModule> > mHwModules;
-        volatile int32_t mNextUniqueId;
-        volatile int32_t mAudioPortGeneration;
-
-        DefaultKeyedVector<audio_patch_handle_t, sp<AudioPatch> > mAudioPatches;
-
-        DefaultKeyedVector<audio_session_t, audio_io_handle_t> mSoundTriggerSessions;
-
-        sp<AudioPatch> mCallTxPatch;
-        sp<AudioPatch> mCallRxPatch;
-
-        // for supporting "beacon" streams, i.e. streams that only play on speaker, and never
-        // when something other than STREAM_TTS (a.k.a. "Transmitted Through Speaker") is playing
-        enum {
-            STARTING_OUTPUT,
-            STARTING_BEACON,
-            STOPPING_OUTPUT,
-            STOPPING_BEACON
-        };
-        uint32_t mBeaconMuteRefCount;   // ref count for stream that would mute beacon
-        uint32_t mBeaconPlayingRefCount;// ref count for the playing beacon streams
-        bool mBeaconMuted;              // has STREAM_TTS been muted
-
-        // custom mix entry in mPolicyMixes
-        class AudioPolicyMix : public RefBase {
-        public:
-            AudioPolicyMix() {}
-
-            AudioMix    mMix;                   // Audio policy mix descriptor
-            sp<AudioOutputDescriptor> mOutput;  // Corresponding output stream
-        };
-        DefaultKeyedVector<String8, sp<AudioPolicyMix> > mPolicyMixes; // list of registered mixes
-
-
-#ifdef AUDIO_POLICY_TEST
-        Mutex   mLock;
-        Condition mWaitWorkCV;
-
-        int             mCurOutput;
-        bool            mDirectOutput;
-        audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS];
-        int             mTestInput;
-        uint32_t        mTestDevice;
-        uint32_t        mTestSamplingRate;
-        uint32_t        mTestFormat;
-        uint32_t        mTestChannels;
-        uint32_t        mTestLatencyMs;
-#endif //AUDIO_POLICY_TEST
-        static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
-                int indexInUi);
-        static bool isVirtualInputDevice(audio_devices_t device);
-        uint32_t nextUniqueId();
-        uint32_t nextAudioPortGeneration();
-private:
-        // updates device caching and output for streams that can influence the
-        //    routing of notifications
-        void handleNotificationRoutingForStream(audio_stream_type_t stream);
-        static bool deviceDistinguishesOnAddress(audio_devices_t device);
-        // find the outputs on a given output descriptor that have the given address.
-        // to be called on an AudioOutputDescriptor whose supported devices (as defined
-        //   in mProfile->mSupportedDevices) matches the device whose address is to be matched.
-        // see deviceDistinguishesOnAddress(audio_devices_t) for whether the device type is one
-        //   where addresses are used to distinguish between one connected device and another.
-        void findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/,
-                const audio_devices_t device /*in*/,
-                const String8 address /*in*/,
-                SortedVector<audio_io_handle_t>& outputs /*out*/);
-        uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; }
-        // internal method to return the output handle for the given device and format
-        audio_io_handle_t getOutputForDevice(
-                audio_devices_t device,
-                audio_session_t session,
-                audio_stream_type_t stream,
-                uint32_t samplingRate,
-                audio_format_t format,
-                audio_channel_mask_t channelMask,
-                audio_output_flags_t flags,
-                const audio_offload_info_t *offloadInfo);
-        // internal function to derive a stream type value from audio attributes
-        audio_stream_type_t streamTypefromAttributesInt(const audio_attributes_t *attr);
-        // return true if any output is playing anything besides the stream to ignore
-        bool isAnyOutputActive(audio_stream_type_t streamToIgnore);
-        // event is one of STARTING_OUTPUT, STARTING_BEACON, STOPPING_OUTPUT, STOPPING_BEACON
-        // returns 0 if no mute/unmute event happened, the largest latency of the device where
-        //   the mute/unmute happened
-        uint32_t handleEventForBeacon(int event);
-        uint32_t setBeaconMute(bool mute);
-        bool     isValidAttributes(const audio_attributes_t *paa);
-
-        // select input device corresponding to requested audio source and return associated policy
-        // mix if any. Calls getDeviceForInputSource().
-        audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource,
-                                                        AudioMix **policyMix = NULL);
-
-        // Called by setDeviceConnectionState().
-        status_t setDeviceConnectionStateInt(audio_devices_t device,
-                                                          audio_policy_dev_state_t state,
-                                                          const char *device_address);
-        sp<DeviceDescriptor>  getDeviceDescriptor(const audio_devices_t device,
-                                                  const char *device_address);
-
-};
-
-};
diff --git a/services/audiopolicy/AudioPolicyFactory.cpp b/services/audiopolicy/manager/AudioPolicyFactory.cpp
similarity index 94%
rename from services/audiopolicy/AudioPolicyFactory.cpp
rename to services/audiopolicy/manager/AudioPolicyFactory.cpp
index 2ae7bc1..9910a1f 100644
--- a/services/audiopolicy/AudioPolicyFactory.cpp
+++ b/services/audiopolicy/manager/AudioPolicyFactory.cpp
@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#include "AudioPolicyManager.h"
+#include "managerdefault/AudioPolicyManager.h"
 
 namespace android {
 
diff --git a/services/audiopolicy/AudioPolicyFactory.cpp b/services/audiopolicy/managerdefault/ApmImplDefinitions.h
similarity index 60%
copy from services/audiopolicy/AudioPolicyFactory.cpp
copy to services/audiopolicy/managerdefault/ApmImplDefinitions.h
index 2ae7bc1..620979b 100644
--- a/services/audiopolicy/AudioPolicyFactory.cpp
+++ b/services/audiopolicy/managerdefault/ApmImplDefinitions.h
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2014 The Android Open Source Project
+ * Copyright (C) 2015 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.
@@ -14,19 +14,19 @@
  * limitations under the License.
  */
 
-#include "AudioPolicyManager.h"
-
 namespace android {
 
-extern "C" AudioPolicyInterface* createAudioPolicyManager(
-        AudioPolicyClientInterface *clientInterface)
-{
-    return new AudioPolicyManager(clientInterface);
-}
+enum routing_strategy {
+    STRATEGY_MEDIA,
+    STRATEGY_PHONE,
+    STRATEGY_SONIFICATION,
+    STRATEGY_SONIFICATION_RESPECTFUL,
+    STRATEGY_DTMF,
+    STRATEGY_ENFORCED_AUDIBLE,
+    STRATEGY_TRANSMITTED_THROUGH_SPEAKER,
+    STRATEGY_ACCESSIBILITY,
+    STRATEGY_REROUTING,
+    NUM_STRATEGIES
+};
 
-extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
-{
-    delete interface;
-}
-
-}; // namespace android
+}; //namespace android
diff --git a/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp b/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp
new file mode 100644
index 0000000..f4054c8
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp
@@ -0,0 +1,100 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::AudioInputDescriptor"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+AudioInputDescriptor::AudioInputDescriptor(const sp<IOProfile>& profile)
+    : mId(0), mIoHandle(0),
+      mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0), mRefCount(0),
+      mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false)
+{
+    if (profile != NULL) {
+        mSamplingRate = profile->pickSamplingRate();
+        mFormat = profile->pickFormat();
+        mChannelMask = profile->pickChannelMask();
+        if (profile->mGains.size() > 0) {
+            profile->mGains[0]->getDefaultConfig(&mGain);
+        }
+    }
+}
+
+void AudioInputDescriptor::toAudioPortConfig(
+                                                   struct audio_port_config *dstConfig,
+                                                   const struct audio_port_config *srcConfig) const
+{
+    ALOG_ASSERT(mProfile != 0,
+                "toAudioPortConfig() called on input with null profile %d", mIoHandle);
+    dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
+                            AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
+    if (srcConfig != NULL) {
+        dstConfig->config_mask |= srcConfig->config_mask;
+    }
+
+    AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+    dstConfig->id = mId;
+    dstConfig->role = AUDIO_PORT_ROLE_SINK;
+    dstConfig->type = AUDIO_PORT_TYPE_MIX;
+    dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
+    dstConfig->ext.mix.handle = mIoHandle;
+    dstConfig->ext.mix.usecase.source = mInputSource;
+}
+
+void AudioInputDescriptor::toAudioPort(
+                                                    struct audio_port *port) const
+{
+    ALOG_ASSERT(mProfile != 0, "toAudioPort() called on input with null profile %d", mIoHandle);
+
+    mProfile->toAudioPort(port);
+    port->id = mId;
+    toAudioPortConfig(&port->active_config);
+    port->ext.mix.hw_module = mProfile->mModule->mHandle;
+    port->ext.mix.handle = mIoHandle;
+    port->ext.mix.latency_class = AUDIO_LATENCY_NORMAL;
+}
+
+status_t AudioInputDescriptor::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, " ID: %d\n", mId);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Format: %d\n", mFormat);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Open Ref Count %d\n", mOpenRefCount);
+    result.append(buffer);
+
+    write(fd, result.string(), result.size());
+
+    return NO_ERROR;
+}
+
+}; //namespace android
diff --git a/services/audiopolicy/managerdefault/AudioInputDescriptor.h b/services/audiopolicy/managerdefault/AudioInputDescriptor.h
new file mode 100644
index 0000000..02579e6
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioInputDescriptor.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+namespace android {
+
+// descriptor for audio inputs. Used to maintain current configuration of each opened audio input
+// and keep track of the usage of this input.
+class AudioInputDescriptor: public AudioPortConfig
+{
+public:
+    AudioInputDescriptor(const sp<IOProfile>& profile);
+
+    status_t    dump(int fd);
+
+    audio_port_handle_t           mId;
+    audio_io_handle_t             mIoHandle;       // input handle
+    audio_devices_t               mDevice;         // current device this input is routed to
+    AudioMix                      *mPolicyMix;     // non NULL when used by a dynamic policy
+    audio_patch_handle_t          mPatchHandle;
+    uint32_t                      mRefCount;       // number of AudioRecord clients using
+    // this input
+    uint32_t                      mOpenRefCount;
+    audio_source_t                mInputSource;    // input source selected by application
+    //(mediarecorder.h)
+    const sp<IOProfile>           mProfile;        // I/O profile this output derives from
+    SortedVector<audio_session_t> mSessions;       // audio sessions attached to this input
+    bool                          mIsSoundTrigger; // used by a soundtrigger capture
+
+    virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+            const struct audio_port_config *srcConfig = NULL) const;
+    virtual sp<AudioPort> getAudioPort() const { return mProfile; }
+    void toAudioPort(struct audio_port *port) const;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp b/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp
new file mode 100644
index 0000000..4b85972
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp
@@ -0,0 +1,221 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::AudioOutputDescriptor"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+AudioOutputDescriptor::AudioOutputDescriptor(
+        const sp<IOProfile>& profile)
+    : mId(0), mIoHandle(0), mLatency(0),
+    mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL),
+    mPatchHandle(0),
+    mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
+{
+    // clear usage count for all stream types
+    for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+        mRefCount[i] = 0;
+        mCurVolume[i] = -1.0;
+        mMuteCount[i] = 0;
+        mStopTime[i] = 0;
+    }
+    for (int i = 0; i < NUM_STRATEGIES; i++) {
+        mStrategyMutedByDevice[i] = false;
+    }
+    if (profile != NULL) {
+        mFlags = (audio_output_flags_t)profile->mFlags;
+        mSamplingRate = profile->pickSamplingRate();
+        mFormat = profile->pickFormat();
+        mChannelMask = profile->pickChannelMask();
+        if (profile->mGains.size() > 0) {
+            profile->mGains[0]->getDefaultConfig(&mGain);
+        }
+    }
+}
+
+audio_devices_t AudioOutputDescriptor::device() const
+{
+    if (isDuplicated()) {
+        return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
+    } else {
+        return mDevice;
+    }
+}
+
+uint32_t AudioOutputDescriptor::latency()
+{
+    if (isDuplicated()) {
+        return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
+    } else {
+        return mLatency;
+    }
+}
+
+bool AudioOutputDescriptor::sharesHwModuleWith(
+        const sp<AudioOutputDescriptor> outputDesc)
+{
+    if (isDuplicated()) {
+        return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
+    } else if (outputDesc->isDuplicated()){
+        return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
+    } else {
+        return (mProfile->mModule == outputDesc->mProfile->mModule);
+    }
+}
+
+void AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream,
+                                                                   int delta)
+{
+    // forward usage count change to attached outputs
+    if (isDuplicated()) {
+        mOutput1->changeRefCount(stream, delta);
+        mOutput2->changeRefCount(stream, delta);
+    }
+    if ((delta + (int)mRefCount[stream]) < 0) {
+        ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d",
+              delta, stream, mRefCount[stream]);
+        mRefCount[stream] = 0;
+        return;
+    }
+    mRefCount[stream] += delta;
+    ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
+}
+
+audio_devices_t AudioOutputDescriptor::supportedDevices()
+{
+    if (isDuplicated()) {
+        return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
+    } else {
+        return mProfile->mSupportedDevices.types() ;
+    }
+}
+
+bool AudioOutputDescriptor::isActive(uint32_t inPastMs) const
+{
+    return isStrategyActive(NUM_STRATEGIES, inPastMs);
+}
+
+bool AudioOutputDescriptor::isStrategyActive(routing_strategy strategy,
+                                                                       uint32_t inPastMs,
+                                                                       nsecs_t sysTime) const
+{
+    if ((sysTime == 0) && (inPastMs != 0)) {
+        sysTime = systemTime();
+    }
+    for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
+        if (i == AUDIO_STREAM_PATCH) {
+            continue;
+        }
+        if (((AudioPolicyManager::getStrategy((audio_stream_type_t)i) == strategy) ||
+                (NUM_STRATEGIES == strategy)) &&
+                isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) {
+            return true;
+        }
+    }
+    return false;
+}
+
+bool AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream,
+                                                                       uint32_t inPastMs,
+                                                                       nsecs_t sysTime) const
+{
+    if (mRefCount[stream] != 0) {
+        return true;
+    }
+    if (inPastMs == 0) {
+        return false;
+    }
+    if (sysTime == 0) {
+        sysTime = systemTime();
+    }
+    if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) {
+        return true;
+    }
+    return false;
+}
+
+void AudioOutputDescriptor::toAudioPortConfig(
+                                                 struct audio_port_config *dstConfig,
+                                                 const struct audio_port_config *srcConfig) const
+{
+    ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle);
+
+    dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
+                            AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
+    if (srcConfig != NULL) {
+        dstConfig->config_mask |= srcConfig->config_mask;
+    }
+    AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+    dstConfig->id = mId;
+    dstConfig->role = AUDIO_PORT_ROLE_SOURCE;
+    dstConfig->type = AUDIO_PORT_TYPE_MIX;
+    dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
+    dstConfig->ext.mix.handle = mIoHandle;
+    dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
+}
+
+void AudioOutputDescriptor::toAudioPort(
+                                                    struct audio_port *port) const
+{
+    ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle);
+    mProfile->toAudioPort(port);
+    port->id = mId;
+    toAudioPortConfig(&port->active_config);
+    port->ext.mix.hw_module = mProfile->mModule->mHandle;
+    port->ext.mix.handle = mIoHandle;
+    port->ext.mix.latency_class =
+            mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL;
+}
+
+status_t AudioOutputDescriptor::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, " ID: %d\n", mId);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Format: %08x\n", mFormat);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Devices %08x\n", device());
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
+    result.append(buffer);
+    for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
+        snprintf(buffer, SIZE, " %02d     %.03f     %02d       %02d\n",
+                 i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
+        result.append(buffer);
+    }
+    write(fd, result.string(), result.size());
+
+    return NO_ERROR;
+}
+
+
+
+}; //namespace android
diff --git a/services/audiopolicy/managerdefault/AudioOutputDescriptor.h b/services/audiopolicy/managerdefault/AudioOutputDescriptor.h
new file mode 100644
index 0000000..32f46e4
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioOutputDescriptor.h
@@ -0,0 +1,69 @@
+/*
+ * Copyright (C) 2015 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 "ApmImplDefinitions.h"
+
+namespace android {
+
+// descriptor for audio outputs. Used to maintain current configuration of each opened audio output
+// and keep track of the usage of this output by each audio stream type.
+class AudioOutputDescriptor: public AudioPortConfig
+{
+public:
+    AudioOutputDescriptor(const sp<IOProfile>& profile);
+
+    status_t    dump(int fd);
+
+    audio_devices_t device() const;
+    void changeRefCount(audio_stream_type_t stream, int delta);
+
+    bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
+    audio_devices_t supportedDevices();
+    uint32_t latency();
+    bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc);
+    bool isActive(uint32_t inPastMs = 0) const;
+    bool isStreamActive(audio_stream_type_t stream,
+                        uint32_t inPastMs = 0,
+                        nsecs_t sysTime = 0) const;
+    bool isStrategyActive(routing_strategy strategy,
+                     uint32_t inPastMs = 0,
+                     nsecs_t sysTime = 0) const;
+
+    virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+                           const struct audio_port_config *srcConfig = NULL) const;
+    virtual sp<AudioPort> getAudioPort() const { return mProfile; }
+    void toAudioPort(struct audio_port *port) const;
+
+    audio_port_handle_t mId;
+    audio_io_handle_t mIoHandle;              // output handle
+    uint32_t mLatency;                  //
+    audio_output_flags_t mFlags;   //
+    audio_devices_t mDevice;                   // current device this output is routed to
+    AudioMix *mPolicyMix;             // non NULL when used by a dynamic policy
+    audio_patch_handle_t mPatchHandle;
+    uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output
+    nsecs_t mStopTime[AUDIO_STREAM_CNT];
+    sp<AudioOutputDescriptor> mOutput1;    // used by duplicated outputs: first output
+    sp<AudioOutputDescriptor> mOutput2;    // used by duplicated outputs: second output
+    float mCurVolume[AUDIO_STREAM_CNT];   // current stream volume
+    int mMuteCount[AUDIO_STREAM_CNT];     // mute request counter
+    const sp<IOProfile> mProfile;          // I/O profile this output derives from
+    bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible
+                                        // device selection. See checkDeviceMuteStrategies()
+    uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
similarity index 72%
rename from services/audiopolicy/AudioPolicyManager.cpp
rename to services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 7f27659..b48dc80 100644
--- a/services/audiopolicy/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AudioPolicyManager"
+#define LOG_TAG "APM::AudioPolicyManager"
 //#define LOG_NDEBUG 0
 
 //#define VERY_VERBOSE_LOGGING
@@ -51,184 +51,29 @@
 namespace android {
 
 // ----------------------------------------------------------------------------
-// Definitions for audio_policy.conf file parsing
-// ----------------------------------------------------------------------------
-
-struct StringToEnum {
-    const char *name;
-    uint32_t value;
-};
-
-#define STRING_TO_ENUM(string) { #string, string }
-#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
-
-const StringToEnum sDeviceNameToEnumTable[] = {
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER_SAFE),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_TELEPHONY_TX),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_LINE),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI_ARC),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPDIF),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_FM),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_LINE),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_AMBIENT),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_ALL_SCO),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_HDMI),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_TELEPHONY_RX),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_DEVICE),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_FM_TUNER),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_TV_TUNER),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_LINE),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_SPDIF),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_LOOPBACK),
-};
-
-const StringToEnum sOutputFlagNameToEnumTable[] = {
-    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT),
-    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY),
-    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST),
-    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
-    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD),
-    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING),
-    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_HW_AV_SYNC),
-};
-
-const StringToEnum sInputFlagNameToEnumTable[] = {
-    STRING_TO_ENUM(AUDIO_INPUT_FLAG_FAST),
-    STRING_TO_ENUM(AUDIO_INPUT_FLAG_HW_HOTWORD),
-};
-
-const StringToEnum sFormatNameToEnumTable[] = {
-    STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
-    STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
-    STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT),
-    STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
-    STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT),
-    STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
-    STRING_TO_ENUM(AUDIO_FORMAT_MP3),
-    STRING_TO_ENUM(AUDIO_FORMAT_AAC),
-    STRING_TO_ENUM(AUDIO_FORMAT_AAC_MAIN),
-    STRING_TO_ENUM(AUDIO_FORMAT_AAC_LC),
-    STRING_TO_ENUM(AUDIO_FORMAT_AAC_SSR),
-    STRING_TO_ENUM(AUDIO_FORMAT_AAC_LTP),
-    STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V1),
-    STRING_TO_ENUM(AUDIO_FORMAT_AAC_SCALABLE),
-    STRING_TO_ENUM(AUDIO_FORMAT_AAC_ERLC),
-    STRING_TO_ENUM(AUDIO_FORMAT_AAC_LD),
-    STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V2),
-    STRING_TO_ENUM(AUDIO_FORMAT_AAC_ELD),
-    STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
-    STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1),
-    STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2),
-    STRING_TO_ENUM(AUDIO_FORMAT_OPUS),
-    STRING_TO_ENUM(AUDIO_FORMAT_AC3),
-    STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
-};
-
-const StringToEnum sOutChannelsNameToEnumTable[] = {
-    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
-    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
-    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD),
-    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
-    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
-};
-
-const StringToEnum sInChannelsNameToEnumTable[] = {
-    STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
-    STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
-    STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
-};
-
-const StringToEnum sGainModeNameToEnumTable[] = {
-    STRING_TO_ENUM(AUDIO_GAIN_MODE_JOINT),
-    STRING_TO_ENUM(AUDIO_GAIN_MODE_CHANNELS),
-    STRING_TO_ENUM(AUDIO_GAIN_MODE_RAMP),
-};
-
-
-uint32_t AudioPolicyManager::stringToEnum(const struct StringToEnum *table,
-                                              size_t size,
-                                              const char *name)
-{
-    for (size_t i = 0; i < size; i++) {
-        if (strcmp(table[i].name, name) == 0) {
-            ALOGV("stringToEnum() found %s", table[i].name);
-            return table[i].value;
-        }
-    }
-    return 0;
-}
-
-const char *AudioPolicyManager::enumToString(const struct StringToEnum *table,
-                                              size_t size,
-                                              uint32_t value)
-{
-    for (size_t i = 0; i < size; i++) {
-        if (table[i].value == value) {
-            return table[i].name;
-        }
-    }
-    return "";
-}
-
-bool AudioPolicyManager::stringToBool(const char *value)
-{
-    return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0));
-}
-
-
-// ----------------------------------------------------------------------------
 // AudioPolicyInterface implementation
 // ----------------------------------------------------------------------------
 
 status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device,
-                                                          audio_policy_dev_state_t state,
-                                                  const char *device_address)
+                                                      audio_policy_dev_state_t state,
+                                                      const char *device_address,
+                                                      const char *device_name)
 {
-    return setDeviceConnectionStateInt(device, state, device_address);
+    return setDeviceConnectionStateInt(device, state, device_address, device_name);
 }
 
 status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device,
                                                          audio_policy_dev_state_t state,
-                                                         const char *device_address)
+                                                         const char *device_address,
+                                                         const char *device_name)
 {
-    ALOGV("setDeviceConnectionState() device: %x, state %d, address %s",
-            device, state, device_address != NULL ? device_address : "");
+    ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s",
+-            device, state, device_address, device_name);
 
     // connect/disconnect only 1 device at a time
     if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
 
-    sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address);
+    sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address, device_name);
 
     // handle output devices
     if (audio_is_output_device(device)) {
@@ -259,8 +104,7 @@
                     mAvailableOutputDevices.remove(devDesc);
                     return INVALID_OPERATION;
                 }
-                mAvailableOutputDevices[index]->mId = nextUniqueId();
-                mAvailableOutputDevices[index]->mModule = module;
+                mAvailableOutputDevices[index]->attach(module);
             } else {
                 return NO_MEMORY;
             }
@@ -275,8 +119,7 @@
             ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
                   outputs.size());
 
-
-            // Set connect to HALs
+            // Send connect to HALs
             AudioParameter param = AudioParameter(devDesc->mAddress);
             param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
             mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
@@ -291,7 +134,7 @@
 
             ALOGV("setDeviceConnectionState() disconnecting output device %x", device);
 
-            // Set Disconnect to HALs
+            // Send Disconnect to HALs
             AudioParameter param = AudioParameter(devDesc->mAddress);
             param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
             mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
@@ -377,8 +220,7 @@
 
             index = mAvailableInputDevices.add(devDesc);
             if (index >= 0) {
-                mAvailableInputDevices[index]->mId = nextUniqueId();
-                mAvailableInputDevices[index]->mModule = module;
+                mAvailableInputDevices[index]->attach(module);
             } else {
                 return NO_MEMORY;
             }
@@ -432,7 +274,7 @@
 audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device,
                                                   const char *device_address)
 {
-    sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address);
+    sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address, "");
     DeviceVector *deviceVector;
 
     if (audio_is_output_device(device)) {
@@ -452,9 +294,9 @@
     }
 }
 
-sp<AudioPolicyManager::DeviceDescriptor>  AudioPolicyManager::getDeviceDescriptor(
-                                                                    const audio_devices_t device,
-                                                                    const char *device_address)
+sp<DeviceDescriptor>  AudioPolicyManager::getDeviceDescriptor(const audio_devices_t device,
+                                                              const char *device_address,
+                                                              const char *device_name)
 {
     String8 address = (device_address == NULL) ? String8("") : String8(device_address);
     // handle legacy remote submix case where the address was not always specified
@@ -477,7 +319,8 @@
         }
     }
 
-    sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device);
+    sp<DeviceDescriptor> devDesc =
+            new DeviceDescriptor(String8(device_name != NULL ? device_name : ""), device);
     devDesc->mAddress = address;
     return devDesc;
 }
@@ -640,18 +483,18 @@
         // force routing command to audio hardware when starting a call
         // even if no device change is needed
         force = true;
-        for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+        for (int j = 0; j < ApmGains::DEVICE_CATEGORY_CNT; j++) {
             mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
-                    sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j];
+                    ApmGains::sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j];
         }
     } else if (isStateInCall(oldState) && !isStateInCall(state)) {
         ALOGV("  Exiting call in setPhoneState()");
         // force routing command to audio hardware when exiting a call
         // even if no device change is needed
         force = true;
-        for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+        for (int j = 0; j < ApmGains::DEVICE_CATEGORY_CNT; j++) {
             mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
-                    sVolumeProfiles[AUDIO_STREAM_DTMF][j];
+                    ApmGains::sVolumeProfiles[AUDIO_STREAM_DTMF][j];
         }
     } else if (isStateInCall(state) && (state != oldState)) {
         ALOGV("  Switching between telephony and VoIP in setPhoneState()");
@@ -842,7 +685,7 @@
 
 // Find a direct output profile compatible with the parameters passed, even if the input flags do
 // not explicitly request a direct output
-sp<AudioPolicyManager::IOProfile> AudioPolicyManager::getProfileForDirectOutput(
+sp<IOProfile> AudioPolicyManager::getProfileForDirectOutput(
                                                                audio_devices_t device,
                                                                uint32_t samplingRate,
                                                                audio_format_t format,
@@ -1130,6 +973,10 @@
             if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) {
                 goto non_direct_output;
             }
+            // fall back to mixer output if possible when the direct output could not be open
+            if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) {
+                goto non_direct_output;
+            }
             return AUDIO_IO_HANDLE_NONE;
         }
         outputDesc->mSamplingRate = config.sample_rate;
@@ -1322,7 +1169,8 @@
                 outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
                 setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
                         AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
-                        outputDesc->mPolicyMix->mRegistrationId);
+                        outputDesc->mPolicyMix->mRegistrationId,
+                        "remote-submix");
         }
 
         // force reevaluating accessibility routing when ringtone or alarm starts
@@ -1371,7 +1219,8 @@
                     outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
                 setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
                         AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
-                        outputDesc->mPolicyMix->mRegistrationId);
+                        outputDesc->mPolicyMix->mRegistrationId,
+                        "remote-submix");
             }
 
             outputDesc->mStopTime[stream] = systemTime();
@@ -1672,7 +1521,7 @@
             if (address != "") {
                 setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
                         AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
-                        address);
+                        address, "remote-submix");
             }
         }
     }
@@ -1720,7 +1569,7 @@
             if (address != "") {
                 setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
                                          AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
-                                         address);
+                                         address, "remote-submix");
             }
         }
 
@@ -1849,7 +1698,7 @@
     status_t status = NO_ERROR;
     for (size_t i = 0; i < mOutputs.size(); i++) {
         audio_devices_t curDevice =
-                getDeviceForVolume(mOutputs.valueAt(i)->device());
+                ApmGains::getDeviceForVolume(mOutputs.valueAt(i)->device());
         if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & strategyDevice) != 0)) {
             status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice);
             if (volStatus != NO_ERROR) {
@@ -1879,7 +1728,7 @@
     if (device == AUDIO_DEVICE_OUT_DEFAULT) {
         device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
     }
-    device = getDeviceForVolume(device);
+    device = ApmGains::getDeviceForVolume(device);
 
     *index =  mStreams[stream].getVolumeIndex(device);
     ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index);
@@ -2177,11 +2026,11 @@
         if (mixes[i].mMixType == MIX_TYPE_PLAYERS) {
             setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
                                      AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
-                                     address.string());
+                                     address.string(), "remote-submix");
         } else {
             setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
                                      AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
-                                     address.string());
+                                     address.string(), "remote-submix");
         }
     }
     return NO_ERROR;
@@ -2219,7 +2068,7 @@
         {
             setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
                                      AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
-                                     address.string());
+                                     address.string(), "remote-submix");
         }
 
         if (getDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address.string()) ==
@@ -2227,7 +2076,7 @@
         {
             setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
                                      AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
-                                     address.string());
+                                     address.string(), "remote-submix");
         }
         module->removeOutputProfile(address);
         module->removeInputProfile(address);
@@ -2463,7 +2312,7 @@
     return NO_ERROR;
 }
 
-sp<AudioPolicyManager::AudioOutputDescriptor> AudioPolicyManager::getOutputFromId(
+sp<AudioOutputDescriptor> AudioPolicyManager::getOutputFromId(
                                                                     audio_port_handle_t id) const
 {
     sp<AudioOutputDescriptor> outputDesc = NULL;
@@ -2476,7 +2325,7 @@
     return outputDesc;
 }
 
-sp<AudioPolicyManager::AudioInputDescriptor> AudioPolicyManager::getInputFromId(
+sp<AudioInputDescriptor> AudioPolicyManager::getInputFromId(
                                                                     audio_port_handle_t id) const
 {
     sp<AudioInputDescriptor> inputDesc = NULL;
@@ -2489,7 +2338,7 @@
     return inputDesc;
 }
 
-sp <AudioPolicyManager::HwModule> AudioPolicyManager::getModuleForDevice(
+sp <HwModule> AudioPolicyManager::getModuleForDevice(
                                                                     audio_devices_t device) const
 {
     sp <HwModule> module;
@@ -2517,7 +2366,7 @@
     return module;
 }
 
-sp <AudioPolicyManager::HwModule> AudioPolicyManager::getModuleFromName(const char *name) const
+sp <HwModule> AudioPolicyManager::getModuleFromName(const char *name) const
 {
     sp <HwModule> module;
 
@@ -3042,6 +2891,8 @@
     return android_atomic_inc(&mAudioPortGeneration);
 }
 
+int32_t volatile AudioPolicyManager::mNextUniqueId = 1;
+
 AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
     :
 #ifdef AUDIO_POLICY_TEST
@@ -3052,7 +2903,7 @@
     mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
     mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
     mA2dpSuspended(false),
-    mSpeakerDrcEnabled(false), mNextUniqueId(1),
+    mSpeakerDrcEnabled(false),
     mAudioPortGeneration(1),
     mBeaconMuteRefCount(0),
     mBeaconPlayingRefCount(0),
@@ -3065,7 +2916,7 @@
         mForceUse[i] = AUDIO_POLICY_FORCE_NONE;
     }
 
-    mDefaultOutputDevice = new DeviceDescriptor(String8(""), AUDIO_DEVICE_OUT_SPEAKER);
+    mDefaultOutputDevice = new DeviceDescriptor(String8("Speaker"), AUDIO_DEVICE_OUT_SPEAKER);
     if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) {
         if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) {
             ALOGE("could not load audio policy configuration file, setting defaults");
@@ -3148,9 +2999,8 @@
                     ssize_t index =
                             mAvailableOutputDevices.indexOf(outProfile->mSupportedDevices[k]);
                     // give a valid ID to an attached device once confirmed it is reachable
-                    if ((index >= 0) && (mAvailableOutputDevices[index]->mId == 0)) {
-                        mAvailableOutputDevices[index]->mId = nextUniqueId();
-                        mAvailableOutputDevices[index]->mModule = mHwModules[i];
+                    if (index >= 0 && !mAvailableOutputDevices[index]->isAttached()) {
+                        mAvailableOutputDevices[index]->attach(mHwModules[i]);
                     }
                 }
                 if (mPrimaryOutput == 0 &&
@@ -3217,9 +3067,8 @@
                     ssize_t index =
                             mAvailableInputDevices.indexOf(inProfile->mSupportedDevices[k]);
                     // give a valid ID to an attached device once confirmed it is reachable
-                    if ((index >= 0) && (mAvailableInputDevices[index]->mId == 0)) {
-                        mAvailableInputDevices[index]->mId = nextUniqueId();
-                        mAvailableInputDevices[index]->mModule = mHwModules[i];
+                    if (index >= 0 && !mAvailableInputDevices[index]->isAttached()) {
+                        mAvailableInputDevices[index]->attach(mHwModules[i]);
                     }
                 }
                 mpClientInterface->closeInput(input);
@@ -3232,7 +3081,7 @@
     }
     // make sure all attached devices have been allocated a unique ID
     for (size_t i = 0; i  < mAvailableOutputDevices.size();) {
-        if (mAvailableOutputDevices[i]->mId == 0) {
+        if (!mAvailableOutputDevices[i]->isAttached()) {
             ALOGW("Input device %08x unreachable", mAvailableOutputDevices[i]->mDeviceType);
             mAvailableOutputDevices.remove(mAvailableOutputDevices[i]);
             continue;
@@ -3240,7 +3089,7 @@
         i++;
     }
     for (size_t i = 0; i  < mAvailableInputDevices.size();) {
-        if (mAvailableInputDevices[i]->mId == 0) {
+        if (!mAvailableInputDevices[i]->isAttached()) {
             ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->mDeviceType);
             mAvailableInputDevices.remove(mAvailableInputDevices[i]);
             continue;
@@ -4328,7 +4177,7 @@
         return AUDIO_DEVICE_NONE;
     }
     audio_devices_t devices;
-    AudioPolicyManager::routing_strategy strategy = getStrategy(stream);
+    routing_strategy strategy = getStrategy(stream);
     devices = getDeviceForStrategy(strategy, true /*fromCache*/);
     SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(devices, mOutputs);
     for (size_t i = 0; i < outputs.size(); i++) {
@@ -4349,7 +4198,7 @@
     return devices;
 }
 
-AudioPolicyManager::routing_strategy AudioPolicyManager::getStrategy(
+routing_strategy AudioPolicyManager::getStrategy(
         audio_stream_type_t stream) {
 
     ALOG_ASSERT(stream != AUDIO_STREAM_PATCH,"getStrategy() called for AUDIO_STREAM_PATCH");
@@ -5128,7 +4977,7 @@
     return status;
 }
 
-sp<AudioPolicyManager::IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device,
+sp<IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device,
                                                    String8 address,
                                                    uint32_t& samplingRate,
                                                    audio_format_t format,
@@ -5338,305 +5187,29 @@
 }
 
 
-audio_devices_t AudioPolicyManager::getDeviceForVolume(audio_devices_t device)
-{
-    if (device == AUDIO_DEVICE_NONE) {
-        // this happens when forcing a route update and no track is active on an output.
-        // In this case the returned category is not important.
-        device =  AUDIO_DEVICE_OUT_SPEAKER;
-    } else if (popcount(device) > 1) {
-        // Multiple device selection is either:
-        //  - speaker + one other device: give priority to speaker in this case.
-        //  - one A2DP device + another device: happens with duplicated output. In this case
-        // retain the device on the A2DP output as the other must not correspond to an active
-        // selection if not the speaker.
-        //  - HDMI-CEC system audio mode only output: give priority to available item in order.
-        if (device & AUDIO_DEVICE_OUT_SPEAKER) {
-            device = AUDIO_DEVICE_OUT_SPEAKER;
-        } else if (device & AUDIO_DEVICE_OUT_HDMI_ARC) {
-            device = AUDIO_DEVICE_OUT_HDMI_ARC;
-        } else if (device & AUDIO_DEVICE_OUT_AUX_LINE) {
-            device = AUDIO_DEVICE_OUT_AUX_LINE;
-        } else if (device & AUDIO_DEVICE_OUT_SPDIF) {
-            device = AUDIO_DEVICE_OUT_SPDIF;
-        } else {
-            device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
-        }
-    }
-
-    /*SPEAKER_SAFE is an alias of SPEAKER for purposes of volume control*/
-    if (device == AUDIO_DEVICE_OUT_SPEAKER_SAFE)
-        device = AUDIO_DEVICE_OUT_SPEAKER;
-
-    ALOGW_IF(popcount(device) != 1,
-            "getDeviceForVolume() invalid device combination: %08x",
-            device);
-
-    return device;
-}
-
-AudioPolicyManager::device_category AudioPolicyManager::getDeviceCategory(audio_devices_t device)
-{
-    switch(getDeviceForVolume(device)) {
-        case AUDIO_DEVICE_OUT_EARPIECE:
-            return DEVICE_CATEGORY_EARPIECE;
-        case AUDIO_DEVICE_OUT_WIRED_HEADSET:
-        case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
-        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
-        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
-        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
-        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
-            return DEVICE_CATEGORY_HEADSET;
-        case AUDIO_DEVICE_OUT_LINE:
-        case AUDIO_DEVICE_OUT_AUX_DIGITAL:
-        /*USB?  Remote submix?*/
-            return DEVICE_CATEGORY_EXT_MEDIA;
-        case AUDIO_DEVICE_OUT_SPEAKER:
-        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
-        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
-        case AUDIO_DEVICE_OUT_USB_ACCESSORY:
-        case AUDIO_DEVICE_OUT_USB_DEVICE:
-        case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
-        default:
-            return DEVICE_CATEGORY_SPEAKER;
-    }
-}
-
-/* static */
-float AudioPolicyManager::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
-        int indexInUi)
-{
-    device_category deviceCategory = getDeviceCategory(device);
-    const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory];
-
-    // the volume index in the UI is relative to the min and max volume indices for this stream type
-    int nbSteps = 1 + curve[VOLMAX].mIndex -
-            curve[VOLMIN].mIndex;
-    int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) /
-            (streamDesc.mIndexMax - streamDesc.mIndexMin);
-
-    // find what part of the curve this index volume belongs to, or if it's out of bounds
-    int segment = 0;
-    if (volIdx < curve[VOLMIN].mIndex) {         // out of bounds
-        return 0.0f;
-    } else if (volIdx < curve[VOLKNEE1].mIndex) {
-        segment = 0;
-    } else if (volIdx < curve[VOLKNEE2].mIndex) {
-        segment = 1;
-    } else if (volIdx <= curve[VOLMAX].mIndex) {
-        segment = 2;
-    } else {                                                               // out of bounds
-        return 1.0f;
-    }
-
-    // linear interpolation in the attenuation table in dB
-    float decibels = curve[segment].mDBAttenuation +
-            ((float)(volIdx - curve[segment].mIndex)) *
-                ( (curve[segment+1].mDBAttenuation -
-                        curve[segment].mDBAttenuation) /
-                    ((float)(curve[segment+1].mIndex -
-                            curve[segment].mIndex)) );
-
-    float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
-
-    ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
-            curve[segment].mIndex, volIdx,
-            curve[segment+1].mIndex,
-            curve[segment].mDBAttenuation,
-            decibels,
-            curve[segment+1].mDBAttenuation,
-            amplification);
-
-    return amplification;
-}
-
-const AudioPolicyManager::VolumeCurvePoint
-    AudioPolicyManager::sDefaultVolumeCurve[AudioPolicyManager::VOLCNT] = {
-    {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
-    AudioPolicyManager::sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT] = {
-    {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
-    AudioPolicyManager::sExtMediaSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
-    {1, -58.0f}, {20, -40.0f}, {60, -21.0f}, {100, -10.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
-    AudioPolicyManager::sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT] = {
-    {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
-    AudioPolicyManager::sSpeakerMediaVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
-    {1, -55.0f}, {20, -43.0f}, {86, -12.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
-    AudioPolicyManager::sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT] = {
-    {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
-    AudioPolicyManager::sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
-    {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f}
-};
-
-// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks
-// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets.
-// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java).
-// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset.
-
-const AudioPolicyManager::VolumeCurvePoint
-    AudioPolicyManager::sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
-    {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
-    AudioPolicyManager::sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
-    {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
-    AudioPolicyManager::sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
-    {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
-    AudioPolicyManager::sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = {
-    {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
-    AudioPolicyManager::sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = {
-    {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
-    AudioPolicyManager::sLinearVolumeCurve[AudioPolicyManager::VOLCNT] = {
-    {0, -96.0f}, {33, -68.0f}, {66, -34.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
-    AudioPolicyManager::sSilentVolumeCurve[AudioPolicyManager::VOLCNT] = {
-    {0, -96.0f}, {1, -96.0f}, {2, -96.0f}, {100, -96.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
-    AudioPolicyManager::sFullScaleVolumeCurve[AudioPolicyManager::VOLCNT] = {
-    {0, 0.0f}, {1, 0.0f}, {2, 0.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
-            *AudioPolicyManager::sVolumeProfiles[AUDIO_STREAM_CNT]
-                                                   [AudioPolicyManager::DEVICE_CATEGORY_CNT] = {
-    { // AUDIO_STREAM_VOICE_CALL
-        sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
-        sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
-    },
-    { // AUDIO_STREAM_SYSTEM
-        sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultSystemVolumeCurve,  // DEVICE_CATEGORY_EARPIECE
-        sExtMediaSystemVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
-    },
-    { // AUDIO_STREAM_RING
-        sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultVolumeCurve,  // DEVICE_CATEGORY_EARPIECE
-        sExtMediaSystemVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
-    },
-    { // AUDIO_STREAM_MUSIC
-        sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
-        sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
-    },
-    { // AUDIO_STREAM_ALARM
-        sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultVolumeCurve,  // DEVICE_CATEGORY_EARPIECE
-        sExtMediaSystemVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
-    },
-    { // AUDIO_STREAM_NOTIFICATION
-        sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultVolumeCurve,  // DEVICE_CATEGORY_EARPIECE
-        sExtMediaSystemVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
-    },
-    { // AUDIO_STREAM_BLUETOOTH_SCO
-        sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
-        sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
-    },
-    { // AUDIO_STREAM_ENFORCED_AUDIBLE
-        sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
-        sExtMediaSystemVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
-    },
-    {  // AUDIO_STREAM_DTMF
-        sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
-        sExtMediaSystemVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
-    },
-    { // AUDIO_STREAM_TTS
-      // "Transmitted Through Speaker": always silent except on DEVICE_CATEGORY_SPEAKER
-        sSilentVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sLinearVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sSilentVolumeCurve, // DEVICE_CATEGORY_EARPIECE
-        sSilentVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
-    },
-    { // AUDIO_STREAM_ACCESSIBILITY
-        sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
-        sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
-    },
-    { // AUDIO_STREAM_REROUTING
-        sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
-        sFullScaleVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
-    },
-    { // AUDIO_STREAM_PATCH
-        sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
-        sFullScaleVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
-    },
-};
-
 void AudioPolicyManager::initializeVolumeCurves()
 {
     for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
-        for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+        for (int j = 0; j < ApmGains::DEVICE_CATEGORY_CNT; j++) {
             mStreams[i].mVolumeCurve[j] =
-                    sVolumeProfiles[i][j];
+                    ApmGains::sVolumeProfiles[i][j];
         }
     }
 
     // Check availability of DRC on speaker path: if available, override some of the speaker curves
     if (mSpeakerDrcEnabled) {
-        mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
-                sDefaultSystemVolumeCurveDrc;
-        mStreams[AUDIO_STREAM_RING].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
-                sSpeakerSonificationVolumeCurveDrc;
-        mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
-                sSpeakerSonificationVolumeCurveDrc;
-        mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
-                sSpeakerSonificationVolumeCurveDrc;
-        mStreams[AUDIO_STREAM_MUSIC].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
-                sSpeakerMediaVolumeCurveDrc;
-        mStreams[AUDIO_STREAM_ACCESSIBILITY].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
-                sSpeakerMediaVolumeCurveDrc;
+        mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+                ApmGains::sDefaultSystemVolumeCurveDrc;
+        mStreams[AUDIO_STREAM_RING].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+                ApmGains::sSpeakerSonificationVolumeCurveDrc;
+        mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+                ApmGains::sSpeakerSonificationVolumeCurveDrc;
+        mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+                ApmGains::sSpeakerSonificationVolumeCurveDrc;
+        mStreams[AUDIO_STREAM_MUSIC].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+                ApmGains::sSpeakerMediaVolumeCurveDrc;
+        mStreams[AUDIO_STREAM_ACCESSIBILITY].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+                ApmGains::sSpeakerMediaVolumeCurveDrc;
     }
 }
 
@@ -5653,7 +5226,7 @@
         device = outputDesc->device();
     }
 
-    volume = volIndexToAmpl(device, streamDesc, index);
+    volume = ApmGains::volIndexToAmpl(device, streamDesc, index);
 
     // if a headset is connected, apply the following rules to ring tones and notifications
     // to avoid sound level bursts in user's ears:
@@ -5909,319 +5482,6 @@
 }
 
 
-// --- AudioOutputDescriptor class implementation
-
-AudioPolicyManager::AudioOutputDescriptor::AudioOutputDescriptor(
-        const sp<IOProfile>& profile)
-    : mId(0), mIoHandle(0), mLatency(0),
-    mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL),
-    mPatchHandle(0),
-    mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
-{
-    // clear usage count for all stream types
-    for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
-        mRefCount[i] = 0;
-        mCurVolume[i] = -1.0;
-        mMuteCount[i] = 0;
-        mStopTime[i] = 0;
-    }
-    for (int i = 0; i < NUM_STRATEGIES; i++) {
-        mStrategyMutedByDevice[i] = false;
-    }
-    if (profile != NULL) {
-        mFlags = (audio_output_flags_t)profile->mFlags;
-        mSamplingRate = profile->pickSamplingRate();
-        mFormat = profile->pickFormat();
-        mChannelMask = profile->pickChannelMask();
-        if (profile->mGains.size() > 0) {
-            profile->mGains[0]->getDefaultConfig(&mGain);
-        }
-    }
-}
-
-audio_devices_t AudioPolicyManager::AudioOutputDescriptor::device() const
-{
-    if (isDuplicated()) {
-        return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
-    } else {
-        return mDevice;
-    }
-}
-
-uint32_t AudioPolicyManager::AudioOutputDescriptor::latency()
-{
-    if (isDuplicated()) {
-        return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
-    } else {
-        return mLatency;
-    }
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::sharesHwModuleWith(
-        const sp<AudioOutputDescriptor> outputDesc)
-{
-    if (isDuplicated()) {
-        return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
-    } else if (outputDesc->isDuplicated()){
-        return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
-    } else {
-        return (mProfile->mModule == outputDesc->mProfile->mModule);
-    }
-}
-
-void AudioPolicyManager::AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream,
-                                                                   int delta)
-{
-    // forward usage count change to attached outputs
-    if (isDuplicated()) {
-        mOutput1->changeRefCount(stream, delta);
-        mOutput2->changeRefCount(stream, delta);
-    }
-    if ((delta + (int)mRefCount[stream]) < 0) {
-        ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d",
-              delta, stream, mRefCount[stream]);
-        mRefCount[stream] = 0;
-        return;
-    }
-    mRefCount[stream] += delta;
-    ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
-}
-
-audio_devices_t AudioPolicyManager::AudioOutputDescriptor::supportedDevices()
-{
-    if (isDuplicated()) {
-        return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
-    } else {
-        return mProfile->mSupportedDevices.types() ;
-    }
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::isActive(uint32_t inPastMs) const
-{
-    return isStrategyActive(NUM_STRATEGIES, inPastMs);
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::isStrategyActive(routing_strategy strategy,
-                                                                       uint32_t inPastMs,
-                                                                       nsecs_t sysTime) const
-{
-    if ((sysTime == 0) && (inPastMs != 0)) {
-        sysTime = systemTime();
-    }
-    for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
-        if (i == AUDIO_STREAM_PATCH) {
-            continue;
-        }
-        if (((getStrategy((audio_stream_type_t)i) == strategy) ||
-                (NUM_STRATEGIES == strategy)) &&
-                isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) {
-            return true;
-        }
-    }
-    return false;
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream,
-                                                                       uint32_t inPastMs,
-                                                                       nsecs_t sysTime) const
-{
-    if (mRefCount[stream] != 0) {
-        return true;
-    }
-    if (inPastMs == 0) {
-        return false;
-    }
-    if (sysTime == 0) {
-        sysTime = systemTime();
-    }
-    if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) {
-        return true;
-    }
-    return false;
-}
-
-void AudioPolicyManager::AudioOutputDescriptor::toAudioPortConfig(
-                                                 struct audio_port_config *dstConfig,
-                                                 const struct audio_port_config *srcConfig) const
-{
-    ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle);
-
-    dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
-                            AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
-    if (srcConfig != NULL) {
-        dstConfig->config_mask |= srcConfig->config_mask;
-    }
-    AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
-
-    dstConfig->id = mId;
-    dstConfig->role = AUDIO_PORT_ROLE_SOURCE;
-    dstConfig->type = AUDIO_PORT_TYPE_MIX;
-    dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
-    dstConfig->ext.mix.handle = mIoHandle;
-    dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
-}
-
-void AudioPolicyManager::AudioOutputDescriptor::toAudioPort(
-                                                    struct audio_port *port) const
-{
-    ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle);
-    mProfile->toAudioPort(port);
-    port->id = mId;
-    toAudioPortConfig(&port->active_config);
-    port->ext.mix.hw_module = mProfile->mModule->mHandle;
-    port->ext.mix.handle = mIoHandle;
-    port->ext.mix.latency_class =
-            mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL;
-}
-
-status_t AudioPolicyManager::AudioOutputDescriptor::dump(int fd)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, " ID: %d\n", mId);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Format: %08x\n", mFormat);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Devices %08x\n", device());
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
-    result.append(buffer);
-    for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
-        snprintf(buffer, SIZE, " %02d     %.03f     %02d       %02d\n",
-                 i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
-        result.append(buffer);
-    }
-    write(fd, result.string(), result.size());
-
-    return NO_ERROR;
-}
-
-// --- AudioInputDescriptor class implementation
-
-AudioPolicyManager::AudioInputDescriptor::AudioInputDescriptor(const sp<IOProfile>& profile)
-    : mId(0), mIoHandle(0),
-      mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0), mRefCount(0),
-      mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false)
-{
-    if (profile != NULL) {
-        mSamplingRate = profile->pickSamplingRate();
-        mFormat = profile->pickFormat();
-        mChannelMask = profile->pickChannelMask();
-        if (profile->mGains.size() > 0) {
-            profile->mGains[0]->getDefaultConfig(&mGain);
-        }
-    }
-}
-
-void AudioPolicyManager::AudioInputDescriptor::toAudioPortConfig(
-                                                   struct audio_port_config *dstConfig,
-                                                   const struct audio_port_config *srcConfig) const
-{
-    ALOG_ASSERT(mProfile != 0,
-                "toAudioPortConfig() called on input with null profile %d", mIoHandle);
-    dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
-                            AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
-    if (srcConfig != NULL) {
-        dstConfig->config_mask |= srcConfig->config_mask;
-    }
-
-    AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
-
-    dstConfig->id = mId;
-    dstConfig->role = AUDIO_PORT_ROLE_SINK;
-    dstConfig->type = AUDIO_PORT_TYPE_MIX;
-    dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
-    dstConfig->ext.mix.handle = mIoHandle;
-    dstConfig->ext.mix.usecase.source = mInputSource;
-}
-
-void AudioPolicyManager::AudioInputDescriptor::toAudioPort(
-                                                    struct audio_port *port) const
-{
-    ALOG_ASSERT(mProfile != 0, "toAudioPort() called on input with null profile %d", mIoHandle);
-
-    mProfile->toAudioPort(port);
-    port->id = mId;
-    toAudioPortConfig(&port->active_config);
-    port->ext.mix.hw_module = mProfile->mModule->mHandle;
-    port->ext.mix.handle = mIoHandle;
-    port->ext.mix.latency_class = AUDIO_LATENCY_NORMAL;
-}
-
-status_t AudioPolicyManager::AudioInputDescriptor::dump(int fd)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, " ID: %d\n", mId);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Format: %d\n", mFormat);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Open Ref Count %d\n", mOpenRefCount);
-    result.append(buffer);
-
-    write(fd, result.string(), result.size());
-
-    return NO_ERROR;
-}
-
-// --- StreamDescriptor class implementation
-
-AudioPolicyManager::StreamDescriptor::StreamDescriptor()
-    :   mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
-{
-    mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
-}
-
-int AudioPolicyManager::StreamDescriptor::getVolumeIndex(audio_devices_t device)
-{
-    device = AudioPolicyManager::getDeviceForVolume(device);
-    // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT
-    if (mIndexCur.indexOfKey(device) < 0) {
-        device = AUDIO_DEVICE_OUT_DEFAULT;
-    }
-    return mIndexCur.valueFor(device);
-}
-
-void AudioPolicyManager::StreamDescriptor::dump(int fd)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, "%s         %02d         %02d         ",
-             mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax);
-    result.append(buffer);
-    for (size_t i = 0; i < mIndexCur.size(); i++) {
-        snprintf(buffer, SIZE, "%04x : %02d, ",
-                 mIndexCur.keyAt(i),
-                 mIndexCur.valueAt(i));
-        result.append(buffer);
-    }
-    result.append("\n");
-
-    write(fd, result.string(), result.size());
-}
-
 // --- EffectDescriptor class implementation
 
 status_t AudioPolicyManager::EffectDescriptor::dump(int fd)
@@ -6245,1601 +5505,9 @@
     return NO_ERROR;
 }
 
-// --- HwModule class implementation
-
-AudioPolicyManager::HwModule::HwModule(const char *name)
-    : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)),
-      mHalVersion(AUDIO_DEVICE_API_VERSION_MIN), mHandle(0)
-{
-}
-
-AudioPolicyManager::HwModule::~HwModule()
-{
-    for (size_t i = 0; i < mOutputProfiles.size(); i++) {
-        mOutputProfiles[i]->mSupportedDevices.clear();
-    }
-    for (size_t i = 0; i < mInputProfiles.size(); i++) {
-        mInputProfiles[i]->mSupportedDevices.clear();
-    }
-    free((void *)mName);
-}
-
-status_t AudioPolicyManager::HwModule::loadInput(cnode *root)
-{
-    cnode *node = root->first_child;
-
-    sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SINK, this);
-
-    while (node) {
-        if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
-            profile->loadSamplingRates((char *)node->value);
-        } else if (strcmp(node->name, FORMATS_TAG) == 0) {
-            profile->loadFormats((char *)node->value);
-        } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
-            profile->loadInChannels((char *)node->value);
-        } else if (strcmp(node->name, DEVICES_TAG) == 0) {
-            profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
-                                                           mDeclaredDevices);
-        } else if (strcmp(node->name, FLAGS_TAG) == 0) {
-            profile->mFlags = parseInputFlagNames((char *)node->value);
-        } else if (strcmp(node->name, GAINS_TAG) == 0) {
-            profile->loadGains(node);
-        }
-        node = node->next;
-    }
-    ALOGW_IF(profile->mSupportedDevices.isEmpty(),
-            "loadInput() invalid supported devices");
-    ALOGW_IF(profile->mChannelMasks.size() == 0,
-            "loadInput() invalid supported channel masks");
-    ALOGW_IF(profile->mSamplingRates.size() == 0,
-            "loadInput() invalid supported sampling rates");
-    ALOGW_IF(profile->mFormats.size() == 0,
-            "loadInput() invalid supported formats");
-    if (!profile->mSupportedDevices.isEmpty() &&
-            (profile->mChannelMasks.size() != 0) &&
-            (profile->mSamplingRates.size() != 0) &&
-            (profile->mFormats.size() != 0)) {
-
-        ALOGV("loadInput() adding input Supported Devices %04x",
-              profile->mSupportedDevices.types());
-
-        mInputProfiles.add(profile);
-        return NO_ERROR;
-    } else {
-        return BAD_VALUE;
-    }
-}
-
-status_t AudioPolicyManager::HwModule::loadOutput(cnode *root)
-{
-    cnode *node = root->first_child;
-
-    sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SOURCE, this);
-
-    while (node) {
-        if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
-            profile->loadSamplingRates((char *)node->value);
-        } else if (strcmp(node->name, FORMATS_TAG) == 0) {
-            profile->loadFormats((char *)node->value);
-        } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
-            profile->loadOutChannels((char *)node->value);
-        } else if (strcmp(node->name, DEVICES_TAG) == 0) {
-            profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
-                                                           mDeclaredDevices);
-        } else if (strcmp(node->name, FLAGS_TAG) == 0) {
-            profile->mFlags = parseOutputFlagNames((char *)node->value);
-        } else if (strcmp(node->name, GAINS_TAG) == 0) {
-            profile->loadGains(node);
-        }
-        node = node->next;
-    }
-    ALOGW_IF(profile->mSupportedDevices.isEmpty(),
-            "loadOutput() invalid supported devices");
-    ALOGW_IF(profile->mChannelMasks.size() == 0,
-            "loadOutput() invalid supported channel masks");
-    ALOGW_IF(profile->mSamplingRates.size() == 0,
-            "loadOutput() invalid supported sampling rates");
-    ALOGW_IF(profile->mFormats.size() == 0,
-            "loadOutput() invalid supported formats");
-    if (!profile->mSupportedDevices.isEmpty() &&
-            (profile->mChannelMasks.size() != 0) &&
-            (profile->mSamplingRates.size() != 0) &&
-            (profile->mFormats.size() != 0)) {
-
-        ALOGV("loadOutput() adding output Supported Devices %04x, mFlags %04x",
-              profile->mSupportedDevices.types(), profile->mFlags);
-
-        mOutputProfiles.add(profile);
-        return NO_ERROR;
-    } else {
-        return BAD_VALUE;
-    }
-}
-
-status_t AudioPolicyManager::HwModule::loadDevice(cnode *root)
-{
-    cnode *node = root->first_child;
-
-    audio_devices_t type = AUDIO_DEVICE_NONE;
-    while (node) {
-        if (strcmp(node->name, DEVICE_TYPE) == 0) {
-            type = parseDeviceNames((char *)node->value);
-            break;
-        }
-        node = node->next;
-    }
-    if (type == AUDIO_DEVICE_NONE ||
-            (!audio_is_input_device(type) && !audio_is_output_device(type))) {
-        ALOGW("loadDevice() bad type %08x", type);
-        return BAD_VALUE;
-    }
-    sp<DeviceDescriptor> deviceDesc = new DeviceDescriptor(String8(root->name), type);
-    deviceDesc->mModule = this;
-
-    node = root->first_child;
-    while (node) {
-        if (strcmp(node->name, DEVICE_ADDRESS) == 0) {
-            deviceDesc->mAddress = String8((char *)node->value);
-        } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
-            if (audio_is_input_device(type)) {
-                deviceDesc->loadInChannels((char *)node->value);
-            } else {
-                deviceDesc->loadOutChannels((char *)node->value);
-            }
-        } else if (strcmp(node->name, GAINS_TAG) == 0) {
-            deviceDesc->loadGains(node);
-        }
-        node = node->next;
-    }
-
-    ALOGV("loadDevice() adding device name %s type %08x address %s",
-          deviceDesc->mName.string(), type, deviceDesc->mAddress.string());
-
-    mDeclaredDevices.add(deviceDesc);
-
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManager::HwModule::addOutputProfile(String8 name, const audio_config_t *config,
-                                                  audio_devices_t device, String8 address)
-{
-    sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SOURCE, this);
-
-    profile->mSamplingRates.add(config->sample_rate);
-    profile->mChannelMasks.add(config->channel_mask);
-    profile->mFormats.add(config->format);
-
-    sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device);
-    devDesc->mAddress = address;
-    profile->mSupportedDevices.add(devDesc);
-
-    mOutputProfiles.add(profile);
-
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManager::HwModule::removeOutputProfile(String8 name)
-{
-    for (size_t i = 0; i < mOutputProfiles.size(); i++) {
-        if (mOutputProfiles[i]->mName == name) {
-            mOutputProfiles.removeAt(i);
-            break;
-        }
-    }
-
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManager::HwModule::addInputProfile(String8 name, const audio_config_t *config,
-                                                  audio_devices_t device, String8 address)
-{
-    sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SINK, this);
-
-    profile->mSamplingRates.add(config->sample_rate);
-    profile->mChannelMasks.add(config->channel_mask);
-    profile->mFormats.add(config->format);
-
-    sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device);
-    devDesc->mAddress = address;
-    profile->mSupportedDevices.add(devDesc);
-
-    ALOGV("addInputProfile() name %s rate %d mask 0x08", name.string(), config->sample_rate, config->channel_mask);
-
-    mInputProfiles.add(profile);
-
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManager::HwModule::removeInputProfile(String8 name)
-{
-    for (size_t i = 0; i < mInputProfiles.size(); i++) {
-        if (mInputProfiles[i]->mName == name) {
-            mInputProfiles.removeAt(i);
-            break;
-        }
-    }
-
-    return NO_ERROR;
-}
-
-
-void AudioPolicyManager::HwModule::dump(int fd)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, "  - name: %s\n", mName);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "  - handle: %d\n", mHandle);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "  - version: %u.%u\n", mHalVersion >> 8, mHalVersion & 0xFF);
-    result.append(buffer);
-    write(fd, result.string(), result.size());
-    if (mOutputProfiles.size()) {
-        write(fd, "  - outputs:\n", strlen("  - outputs:\n"));
-        for (size_t i = 0; i < mOutputProfiles.size(); i++) {
-            snprintf(buffer, SIZE, "    output %zu:\n", i);
-            write(fd, buffer, strlen(buffer));
-            mOutputProfiles[i]->dump(fd);
-        }
-    }
-    if (mInputProfiles.size()) {
-        write(fd, "  - inputs:\n", strlen("  - inputs:\n"));
-        for (size_t i = 0; i < mInputProfiles.size(); i++) {
-            snprintf(buffer, SIZE, "    input %zu:\n", i);
-            write(fd, buffer, strlen(buffer));
-            mInputProfiles[i]->dump(fd);
-        }
-    }
-    if (mDeclaredDevices.size()) {
-        write(fd, "  - devices:\n", strlen("  - devices:\n"));
-        for (size_t i = 0; i < mDeclaredDevices.size(); i++) {
-            mDeclaredDevices[i]->dump(fd, 4, i);
-        }
-    }
-}
-
-// --- AudioPort class implementation
-
-
-AudioPolicyManager::AudioPort::AudioPort(const String8& name, audio_port_type_t type,
-          audio_port_role_t role, const sp<HwModule>& module) :
-    mName(name), mType(type), mRole(role), mModule(module), mFlags(0)
-{
-    mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) ||
-                    ((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK));
-}
-
-void AudioPolicyManager::AudioPort::toAudioPort(struct audio_port *port) const
-{
-    port->role = mRole;
-    port->type = mType;
-    unsigned int i;
-    for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) {
-        if (mSamplingRates[i] != 0) {
-            port->sample_rates[i] = mSamplingRates[i];
-        }
-    }
-    port->num_sample_rates = i;
-    for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) {
-        if (mChannelMasks[i] != 0) {
-            port->channel_masks[i] = mChannelMasks[i];
-        }
-    }
-    port->num_channel_masks = i;
-    for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) {
-        if (mFormats[i] != 0) {
-            port->formats[i] = mFormats[i];
-        }
-    }
-    port->num_formats = i;
-
-    ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
-
-    for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
-        port->gains[i] = mGains[i]->mGain;
-    }
-    port->num_gains = i;
-}
-
-void AudioPolicyManager::AudioPort::importAudioPort(const sp<AudioPort> port) {
-    for (size_t k = 0 ; k < port->mSamplingRates.size() ; k++) {
-        const uint32_t rate = port->mSamplingRates.itemAt(k);
-        if (rate != 0) { // skip "dynamic" rates
-            bool hasRate = false;
-            for (size_t l = 0 ; l < mSamplingRates.size() ; l++) {
-                if (rate == mSamplingRates.itemAt(l)) {
-                    hasRate = true;
-                    break;
-                }
-            }
-            if (!hasRate) { // never import a sampling rate twice
-                mSamplingRates.add(rate);
-            }
-        }
-    }
-    for (size_t k = 0 ; k < port->mChannelMasks.size() ; k++) {
-        const audio_channel_mask_t mask = port->mChannelMasks.itemAt(k);
-        if (mask != 0) { // skip "dynamic" masks
-            bool hasMask = false;
-            for (size_t l = 0 ; l < mChannelMasks.size() ; l++) {
-                if (mask == mChannelMasks.itemAt(l)) {
-                    hasMask = true;
-                    break;
-                }
-            }
-            if (!hasMask) { // never import a channel mask twice
-                mChannelMasks.add(mask);
-            }
-        }
-    }
-    for (size_t k = 0 ; k < port->mFormats.size() ; k++) {
-        const audio_format_t format = port->mFormats.itemAt(k);
-        if (format != 0) { // skip "dynamic" formats
-            bool hasFormat = false;
-            for (size_t l = 0 ; l < mFormats.size() ; l++) {
-                if (format == mFormats.itemAt(l)) {
-                    hasFormat = true;
-                    break;
-                }
-            }
-            if (!hasFormat) { // never import a channel mask twice
-                mFormats.add(format);
-            }
-        }
-    }
-    for (size_t k = 0 ; k < port->mGains.size() ; k++) {
-        sp<AudioGain> gain = port->mGains.itemAt(k);
-        if (gain != 0) {
-            bool hasGain = false;
-            for (size_t l = 0 ; l < mGains.size() ; l++) {
-                if (gain == mGains.itemAt(l)) {
-                    hasGain = true;
-                    break;
-                }
-            }
-            if (!hasGain) { // never import a gain twice
-                mGains.add(gain);
-            }
-        }
-    }
-}
-
-void AudioPolicyManager::AudioPort::clearCapabilities() {
-    mChannelMasks.clear();
-    mFormats.clear();
-    mSamplingRates.clear();
-    mGains.clear();
-}
-
-void AudioPolicyManager::AudioPort::loadSamplingRates(char *name)
-{
-    char *str = strtok(name, "|");
-
-    // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
-    // rates should be read from the output stream after it is opened for the first time
-    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
-        mSamplingRates.add(0);
-        return;
-    }
-
-    while (str != NULL) {
-        uint32_t rate = atoi(str);
-        if (rate != 0) {
-            ALOGV("loadSamplingRates() adding rate %d", rate);
-            mSamplingRates.add(rate);
-        }
-        str = strtok(NULL, "|");
-    }
-}
-
-void AudioPolicyManager::AudioPort::loadFormats(char *name)
-{
-    char *str = strtok(name, "|");
-
-    // by convention, "0' in the first entry in mFormats indicates the supported formats
-    // should be read from the output stream after it is opened for the first time
-    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
-        mFormats.add(AUDIO_FORMAT_DEFAULT);
-        return;
-    }
-
-    while (str != NULL) {
-        audio_format_t format = (audio_format_t)stringToEnum(sFormatNameToEnumTable,
-                                                             ARRAY_SIZE(sFormatNameToEnumTable),
-                                                             str);
-        if (format != AUDIO_FORMAT_DEFAULT) {
-            mFormats.add(format);
-        }
-        str = strtok(NULL, "|");
-    }
-}
-
-void AudioPolicyManager::AudioPort::loadInChannels(char *name)
-{
-    const char *str = strtok(name, "|");
-
-    ALOGV("loadInChannels() %s", name);
-
-    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
-        mChannelMasks.add(0);
-        return;
-    }
-
-    while (str != NULL) {
-        audio_channel_mask_t channelMask =
-                (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
-                                                   ARRAY_SIZE(sInChannelsNameToEnumTable),
-                                                   str);
-        if (channelMask != 0) {
-            ALOGV("loadInChannels() adding channelMask %04x", channelMask);
-            mChannelMasks.add(channelMask);
-        }
-        str = strtok(NULL, "|");
-    }
-}
-
-void AudioPolicyManager::AudioPort::loadOutChannels(char *name)
-{
-    const char *str = strtok(name, "|");
-
-    ALOGV("loadOutChannels() %s", name);
-
-    // by convention, "0' in the first entry in mChannelMasks indicates the supported channel
-    // masks should be read from the output stream after it is opened for the first time
-    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
-        mChannelMasks.add(0);
-        return;
-    }
-
-    while (str != NULL) {
-        audio_channel_mask_t channelMask =
-                (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
-                                                   ARRAY_SIZE(sOutChannelsNameToEnumTable),
-                                                   str);
-        if (channelMask != 0) {
-            mChannelMasks.add(channelMask);
-        }
-        str = strtok(NULL, "|");
-    }
-    return;
-}
-
-audio_gain_mode_t AudioPolicyManager::AudioPort::loadGainMode(char *name)
-{
-    const char *str = strtok(name, "|");
-
-    ALOGV("loadGainMode() %s", name);
-    audio_gain_mode_t mode = 0;
-    while (str != NULL) {
-        mode |= (audio_gain_mode_t)stringToEnum(sGainModeNameToEnumTable,
-                                                ARRAY_SIZE(sGainModeNameToEnumTable),
-                                                str);
-        str = strtok(NULL, "|");
-    }
-    return mode;
-}
-
-void AudioPolicyManager::AudioPort::loadGain(cnode *root, int index)
-{
-    cnode *node = root->first_child;
-
-    sp<AudioGain> gain = new AudioGain(index, mUseInChannelMask);
-
-    while (node) {
-        if (strcmp(node->name, GAIN_MODE) == 0) {
-            gain->mGain.mode = loadGainMode((char *)node->value);
-        } else if (strcmp(node->name, GAIN_CHANNELS) == 0) {
-            if (mUseInChannelMask) {
-                gain->mGain.channel_mask =
-                        (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
-                                                           ARRAY_SIZE(sInChannelsNameToEnumTable),
-                                                           (char *)node->value);
-            } else {
-                gain->mGain.channel_mask =
-                        (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
-                                                           ARRAY_SIZE(sOutChannelsNameToEnumTable),
-                                                           (char *)node->value);
-            }
-        } else if (strcmp(node->name, GAIN_MIN_VALUE) == 0) {
-            gain->mGain.min_value = atoi((char *)node->value);
-        } else if (strcmp(node->name, GAIN_MAX_VALUE) == 0) {
-            gain->mGain.max_value = atoi((char *)node->value);
-        } else if (strcmp(node->name, GAIN_DEFAULT_VALUE) == 0) {
-            gain->mGain.default_value = atoi((char *)node->value);
-        } else if (strcmp(node->name, GAIN_STEP_VALUE) == 0) {
-            gain->mGain.step_value = atoi((char *)node->value);
-        } else if (strcmp(node->name, GAIN_MIN_RAMP_MS) == 0) {
-            gain->mGain.min_ramp_ms = atoi((char *)node->value);
-        } else if (strcmp(node->name, GAIN_MAX_RAMP_MS) == 0) {
-            gain->mGain.max_ramp_ms = atoi((char *)node->value);
-        }
-        node = node->next;
-    }
-
-    ALOGV("loadGain() adding new gain mode %08x channel mask %08x min mB %d max mB %d",
-          gain->mGain.mode, gain->mGain.channel_mask, gain->mGain.min_value, gain->mGain.max_value);
-
-    if (gain->mGain.mode == 0) {
-        return;
-    }
-    mGains.add(gain);
-}
-
-void AudioPolicyManager::AudioPort::loadGains(cnode *root)
-{
-    cnode *node = root->first_child;
-    int index = 0;
-    while (node) {
-        ALOGV("loadGains() loading gain %s", node->name);
-        loadGain(node, index++);
-        node = node->next;
-    }
-}
-
-status_t AudioPolicyManager::AudioPort::checkExactSamplingRate(uint32_t samplingRate) const
-{
-    if (mSamplingRates.isEmpty()) {
-        return NO_ERROR;
-    }
-
-    for (size_t i = 0; i < mSamplingRates.size(); i ++) {
-        if (mSamplingRates[i] == samplingRate) {
-            return NO_ERROR;
-        }
-    }
-    return BAD_VALUE;
-}
-
-status_t AudioPolicyManager::AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate,
-        uint32_t *updatedSamplingRate) const
-{
-    if (mSamplingRates.isEmpty()) {
-        return NO_ERROR;
-    }
-
-    // Search for the closest supported sampling rate that is above (preferred)
-    // or below (acceptable) the desired sampling rate, within a permitted ratio.
-    // The sampling rates do not need to be sorted in ascending order.
-    ssize_t maxBelow = -1;
-    ssize_t minAbove = -1;
-    uint32_t candidate;
-    for (size_t i = 0; i < mSamplingRates.size(); i++) {
-        candidate = mSamplingRates[i];
-        if (candidate == samplingRate) {
-            if (updatedSamplingRate != NULL) {
-                *updatedSamplingRate = candidate;
-            }
-            return NO_ERROR;
-        }
-        // candidate < desired
-        if (candidate < samplingRate) {
-            if (maxBelow < 0 || candidate > mSamplingRates[maxBelow]) {
-                maxBelow = i;
-            }
-        // candidate > desired
-        } else {
-            if (minAbove < 0 || candidate < mSamplingRates[minAbove]) {
-                minAbove = i;
-            }
-        }
-    }
-    // This uses hard-coded knowledge about AudioFlinger resampling ratios.
-    // TODO Move these assumptions out.
-    static const uint32_t kMaxDownSampleRatio = 6;  // beyond this aliasing occurs
-    static const uint32_t kMaxUpSampleRatio = 256;  // beyond this sample rate inaccuracies occur
-                                                    // due to approximation by an int32_t of the
-                                                    // phase increments
-    // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
-    if (minAbove >= 0) {
-        candidate = mSamplingRates[minAbove];
-        if (candidate / kMaxDownSampleRatio <= samplingRate) {
-            if (updatedSamplingRate != NULL) {
-                *updatedSamplingRate = candidate;
-            }
-            return NO_ERROR;
-        }
-    }
-    // But if we have to up-sample from a lower sampling rate, that's OK.
-    if (maxBelow >= 0) {
-        candidate = mSamplingRates[maxBelow];
-        if (candidate * kMaxUpSampleRatio >= samplingRate) {
-            if (updatedSamplingRate != NULL) {
-                *updatedSamplingRate = candidate;
-            }
-            return NO_ERROR;
-        }
-    }
-    // leave updatedSamplingRate unmodified
-    return BAD_VALUE;
-}
-
-status_t AudioPolicyManager::AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) const
-{
-    if (mChannelMasks.isEmpty()) {
-        return NO_ERROR;
-    }
-
-    for (size_t i = 0; i < mChannelMasks.size(); i++) {
-        if (mChannelMasks[i] == channelMask) {
-            return NO_ERROR;
-        }
-    }
-    return BAD_VALUE;
-}
-
-status_t AudioPolicyManager::AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask)
-        const
-{
-    if (mChannelMasks.isEmpty()) {
-        return NO_ERROR;
-    }
-
-    const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
-    for (size_t i = 0; i < mChannelMasks.size(); i ++) {
-        // FIXME Does not handle multi-channel automatic conversions yet
-        audio_channel_mask_t supported = mChannelMasks[i];
-        if (supported == channelMask) {
-            return NO_ERROR;
-        }
-        if (isRecordThread) {
-            // This uses hard-coded knowledge that AudioFlinger can silently down-mix and up-mix.
-            // FIXME Abstract this out to a table.
-            if (((supported == AUDIO_CHANNEL_IN_FRONT_BACK || supported == AUDIO_CHANNEL_IN_STEREO)
-                    && channelMask == AUDIO_CHANNEL_IN_MONO) ||
-                (supported == AUDIO_CHANNEL_IN_MONO && (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK
-                    || channelMask == AUDIO_CHANNEL_IN_STEREO))) {
-                return NO_ERROR;
-            }
-        }
-    }
-    return BAD_VALUE;
-}
-
-status_t AudioPolicyManager::AudioPort::checkFormat(audio_format_t format) const
-{
-    if (mFormats.isEmpty()) {
-        return NO_ERROR;
-    }
-
-    for (size_t i = 0; i < mFormats.size(); i ++) {
-        if (mFormats[i] == format) {
-            return NO_ERROR;
-        }
-    }
-    return BAD_VALUE;
-}
-
-
-uint32_t AudioPolicyManager::AudioPort::pickSamplingRate() const
-{
-    // special case for uninitialized dynamic profile
-    if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) {
-        return 0;
-    }
-
-    // For direct outputs, pick minimum sampling rate: this helps ensuring that the
-    // channel count / sampling rate combination chosen will be supported by the connected
-    // sink
-    if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
-            (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
-        uint32_t samplingRate = UINT_MAX;
-        for (size_t i = 0; i < mSamplingRates.size(); i ++) {
-            if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) {
-                samplingRate = mSamplingRates[i];
-            }
-        }
-        return (samplingRate == UINT_MAX) ? 0 : samplingRate;
-    }
-
-    uint32_t samplingRate = 0;
-    uint32_t maxRate = MAX_MIXER_SAMPLING_RATE;
-
-    // For mixed output and inputs, use max mixer sampling rates. Do not
-    // limit sampling rate otherwise
-    if (mType != AUDIO_PORT_TYPE_MIX) {
-        maxRate = UINT_MAX;
-    }
-    for (size_t i = 0; i < mSamplingRates.size(); i ++) {
-        if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) {
-            samplingRate = mSamplingRates[i];
-        }
-    }
-    return samplingRate;
-}
-
-audio_channel_mask_t AudioPolicyManager::AudioPort::pickChannelMask() const
-{
-    // special case for uninitialized dynamic profile
-    if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) {
-        return AUDIO_CHANNEL_NONE;
-    }
-    audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE;
-
-    // For direct outputs, pick minimum channel count: this helps ensuring that the
-    // channel count / sampling rate combination chosen will be supported by the connected
-    // sink
-    if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
-            (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
-        uint32_t channelCount = UINT_MAX;
-        for (size_t i = 0; i < mChannelMasks.size(); i ++) {
-            uint32_t cnlCount;
-            if (mUseInChannelMask) {
-                cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
-            } else {
-                cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
-            }
-            if ((cnlCount < channelCount) && (cnlCount > 0)) {
-                channelMask = mChannelMasks[i];
-                channelCount = cnlCount;
-            }
-        }
-        return channelMask;
-    }
-
-    uint32_t channelCount = 0;
-    uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
-
-    // For mixed output and inputs, use max mixer channel count. Do not
-    // limit channel count otherwise
-    if (mType != AUDIO_PORT_TYPE_MIX) {
-        maxCount = UINT_MAX;
-    }
-    for (size_t i = 0; i < mChannelMasks.size(); i ++) {
-        uint32_t cnlCount;
-        if (mUseInChannelMask) {
-            cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
-        } else {
-            cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
-        }
-        if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
-            channelMask = mChannelMasks[i];
-            channelCount = cnlCount;
-        }
-    }
-    return channelMask;
-}
-
-/* format in order of increasing preference */
-const audio_format_t AudioPolicyManager::AudioPort::sPcmFormatCompareTable[] = {
-        AUDIO_FORMAT_DEFAULT,
-        AUDIO_FORMAT_PCM_16_BIT,
-        AUDIO_FORMAT_PCM_8_24_BIT,
-        AUDIO_FORMAT_PCM_24_BIT_PACKED,
-        AUDIO_FORMAT_PCM_32_BIT,
-        AUDIO_FORMAT_PCM_FLOAT,
-};
-
-int AudioPolicyManager::AudioPort::compareFormats(audio_format_t format1,
-                                                  audio_format_t format2)
-{
-    // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any
-    // compressed format and better than any PCM format. This is by design of pickFormat()
-    if (!audio_is_linear_pcm(format1)) {
-        if (!audio_is_linear_pcm(format2)) {
-            return 0;
-        }
-        return 1;
-    }
-    if (!audio_is_linear_pcm(format2)) {
-        return -1;
-    }
-
-    int index1 = -1, index2 = -1;
-    for (size_t i = 0;
-            (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1));
-            i ++) {
-        if (sPcmFormatCompareTable[i] == format1) {
-            index1 = i;
-        }
-        if (sPcmFormatCompareTable[i] == format2) {
-            index2 = i;
-        }
-    }
-    // format1 not found => index1 < 0 => format2 > format1
-    // format2 not found => index2 < 0 => format2 < format1
-    return index1 - index2;
-}
-
-audio_format_t AudioPolicyManager::AudioPort::pickFormat() const
-{
-    // special case for uninitialized dynamic profile
-    if (mFormats.size() == 1 && mFormats[0] == 0) {
-        return AUDIO_FORMAT_DEFAULT;
-    }
-
-    audio_format_t format = AUDIO_FORMAT_DEFAULT;
-    audio_format_t bestFormat =
-            AudioPolicyManager::AudioPort::sPcmFormatCompareTable[
-                ARRAY_SIZE(AudioPolicyManager::AudioPort::sPcmFormatCompareTable) - 1];
-    // For mixed output and inputs, use best mixer output format. Do not
-    // limit format otherwise
-    if ((mType != AUDIO_PORT_TYPE_MIX) ||
-            ((mRole == AUDIO_PORT_ROLE_SOURCE) &&
-             (((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) != 0)))) {
-        bestFormat = AUDIO_FORMAT_INVALID;
-    }
-
-    for (size_t i = 0; i < mFormats.size(); i ++) {
-        if ((compareFormats(mFormats[i], format) > 0) &&
-                (compareFormats(mFormats[i], bestFormat) <= 0)) {
-            format = mFormats[i];
-        }
-    }
-    return format;
-}
-
-status_t AudioPolicyManager::AudioPort::checkGain(const struct audio_gain_config *gainConfig,
-                                                  int index) const
-{
-    if (index < 0 || (size_t)index >= mGains.size()) {
-        return BAD_VALUE;
-    }
-    return mGains[index]->checkConfig(gainConfig);
-}
-
-void AudioPolicyManager::AudioPort::dump(int fd, int spaces) const
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    if (mName.size() != 0) {
-        snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string());
-        result.append(buffer);
-    }
-
-    if (mSamplingRates.size() != 0) {
-        snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, "");
-        result.append(buffer);
-        for (size_t i = 0; i < mSamplingRates.size(); i++) {
-            if (i == 0 && mSamplingRates[i] == 0) {
-                snprintf(buffer, SIZE, "Dynamic");
-            } else {
-                snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
-            }
-            result.append(buffer);
-            result.append(i == (mSamplingRates.size() - 1) ? "" : ", ");
-        }
-        result.append("\n");
-    }
-
-    if (mChannelMasks.size() != 0) {
-        snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, "");
-        result.append(buffer);
-        for (size_t i = 0; i < mChannelMasks.size(); i++) {
-            ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]);
-
-            if (i == 0 && mChannelMasks[i] == 0) {
-                snprintf(buffer, SIZE, "Dynamic");
-            } else {
-                snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
-            }
-            result.append(buffer);
-            result.append(i == (mChannelMasks.size() - 1) ? "" : ", ");
-        }
-        result.append("\n");
-    }
-
-    if (mFormats.size() != 0) {
-        snprintf(buffer, SIZE, "%*s- formats: ", spaces, "");
-        result.append(buffer);
-        for (size_t i = 0; i < mFormats.size(); i++) {
-            const char *formatStr = enumToString(sFormatNameToEnumTable,
-                                                 ARRAY_SIZE(sFormatNameToEnumTable),
-                                                 mFormats[i]);
-            if (i == 0 && strcmp(formatStr, "") == 0) {
-                snprintf(buffer, SIZE, "Dynamic");
-            } else {
-                snprintf(buffer, SIZE, "%s", formatStr);
-            }
-            result.append(buffer);
-            result.append(i == (mFormats.size() - 1) ? "" : ", ");
-        }
-        result.append("\n");
-    }
-    write(fd, result.string(), result.size());
-    if (mGains.size() != 0) {
-        snprintf(buffer, SIZE, "%*s- gains:\n", spaces, "");
-        write(fd, buffer, strlen(buffer) + 1);
-        result.append(buffer);
-        for (size_t i = 0; i < mGains.size(); i++) {
-            mGains[i]->dump(fd, spaces + 2, i);
-        }
-    }
-}
-
-// --- AudioGain class implementation
-
-AudioPolicyManager::AudioGain::AudioGain(int index, bool useInChannelMask)
-{
-    mIndex = index;
-    mUseInChannelMask = useInChannelMask;
-    memset(&mGain, 0, sizeof(struct audio_gain));
-}
-
-void AudioPolicyManager::AudioGain::getDefaultConfig(struct audio_gain_config *config)
-{
-    config->index = mIndex;
-    config->mode = mGain.mode;
-    config->channel_mask = mGain.channel_mask;
-    if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
-        config->values[0] = mGain.default_value;
-    } else {
-        uint32_t numValues;
-        if (mUseInChannelMask) {
-            numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
-        } else {
-            numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
-        }
-        for (size_t i = 0; i < numValues; i++) {
-            config->values[i] = mGain.default_value;
-        }
-    }
-    if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
-        config->ramp_duration_ms = mGain.min_ramp_ms;
-    }
-}
-
-status_t AudioPolicyManager::AudioGain::checkConfig(const struct audio_gain_config *config)
-{
-    if ((config->mode & ~mGain.mode) != 0) {
-        return BAD_VALUE;
-    }
-    if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
-        if ((config->values[0] < mGain.min_value) ||
-                    (config->values[0] > mGain.max_value)) {
-            return BAD_VALUE;
-        }
-    } else {
-        if ((config->channel_mask & ~mGain.channel_mask) != 0) {
-            return BAD_VALUE;
-        }
-        uint32_t numValues;
-        if (mUseInChannelMask) {
-            numValues = audio_channel_count_from_in_mask(config->channel_mask);
-        } else {
-            numValues = audio_channel_count_from_out_mask(config->channel_mask);
-        }
-        for (size_t i = 0; i < numValues; i++) {
-            if ((config->values[i] < mGain.min_value) ||
-                    (config->values[i] > mGain.max_value)) {
-                return BAD_VALUE;
-            }
-        }
-    }
-    if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
-        if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
-                    (config->ramp_duration_ms > mGain.max_ramp_ms)) {
-            return BAD_VALUE;
-        }
-    }
-    return NO_ERROR;
-}
-
-void AudioPolicyManager::AudioGain::dump(int fd, int spaces, int index) const
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, "%*sGain %d:\n", spaces, "", index+1);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "%*s- mode: %08x\n", spaces, "", mGain.mode);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "%*s- min_value: %d mB\n", spaces, "", mGain.min_value);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "%*s- max_value: %d mB\n", spaces, "", mGain.max_value);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "%*s- default_value: %d mB\n", spaces, "", mGain.default_value);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "%*s- step_value: %d mB\n", spaces, "", mGain.step_value);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms);
-    result.append(buffer);
-
-    write(fd, result.string(), result.size());
-}
-
-// --- AudioPortConfig class implementation
-
-AudioPolicyManager::AudioPortConfig::AudioPortConfig()
-{
-    mSamplingRate = 0;
-    mChannelMask = AUDIO_CHANNEL_NONE;
-    mFormat = AUDIO_FORMAT_INVALID;
-    mGain.index = -1;
-}
-
-status_t AudioPolicyManager::AudioPortConfig::applyAudioPortConfig(
-                                                        const struct audio_port_config *config,
-                                                        struct audio_port_config *backupConfig)
-{
-    struct audio_port_config localBackupConfig;
-    status_t status = NO_ERROR;
-
-    localBackupConfig.config_mask = config->config_mask;
-    toAudioPortConfig(&localBackupConfig);
-
-    sp<AudioPort> audioport = getAudioPort();
-    if (audioport == 0) {
-        status = NO_INIT;
-        goto exit;
-    }
-    if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
-        status = audioport->checkExactSamplingRate(config->sample_rate);
-        if (status != NO_ERROR) {
-            goto exit;
-        }
-        mSamplingRate = config->sample_rate;
-    }
-    if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
-        status = audioport->checkExactChannelMask(config->channel_mask);
-        if (status != NO_ERROR) {
-            goto exit;
-        }
-        mChannelMask = config->channel_mask;
-    }
-    if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
-        status = audioport->checkFormat(config->format);
-        if (status != NO_ERROR) {
-            goto exit;
-        }
-        mFormat = config->format;
-    }
-    if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) {
-        status = audioport->checkGain(&config->gain, config->gain.index);
-        if (status != NO_ERROR) {
-            goto exit;
-        }
-        mGain = config->gain;
-    }
-
-exit:
-    if (status != NO_ERROR) {
-        applyAudioPortConfig(&localBackupConfig);
-    }
-    if (backupConfig != NULL) {
-        *backupConfig = localBackupConfig;
-    }
-    return status;
-}
-
-void AudioPolicyManager::AudioPortConfig::toAudioPortConfig(
-                                                    struct audio_port_config *dstConfig,
-                                                    const struct audio_port_config *srcConfig) const
-{
-    if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
-        dstConfig->sample_rate = mSamplingRate;
-        if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) {
-            dstConfig->sample_rate = srcConfig->sample_rate;
-        }
-    } else {
-        dstConfig->sample_rate = 0;
-    }
-    if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
-        dstConfig->channel_mask = mChannelMask;
-        if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) {
-            dstConfig->channel_mask = srcConfig->channel_mask;
-        }
-    } else {
-        dstConfig->channel_mask = AUDIO_CHANNEL_NONE;
-    }
-    if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
-        dstConfig->format = mFormat;
-        if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) {
-            dstConfig->format = srcConfig->format;
-        }
-    } else {
-        dstConfig->format = AUDIO_FORMAT_INVALID;
-    }
-    if (dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
-        dstConfig->gain = mGain;
-        if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)) {
-            dstConfig->gain = srcConfig->gain;
-        }
-    } else {
-        dstConfig->gain.index = -1;
-    }
-    if (dstConfig->gain.index != -1) {
-        dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
-    } else {
-        dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
-    }
-}
-
-// --- IOProfile class implementation
-
-AudioPolicyManager::IOProfile::IOProfile(const String8& name, audio_port_role_t role,
-                                         const sp<HwModule>& module)
-    : AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module)
-{
-}
-
-AudioPolicyManager::IOProfile::~IOProfile()
-{
-}
-
-// checks if the IO profile is compatible with specified parameters.
-// Sampling rate, format and channel mask must be specified in order to
-// get a valid a match
-bool AudioPolicyManager::IOProfile::isCompatibleProfile(audio_devices_t device,
-                                                        String8 address,
-                                                        uint32_t samplingRate,
-                                                        uint32_t *updatedSamplingRate,
-                                                        audio_format_t format,
-                                                        audio_channel_mask_t channelMask,
-                                                        uint32_t flags) const
-{
-    const bool isPlaybackThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SOURCE;
-    const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
-    ALOG_ASSERT(isPlaybackThread != isRecordThread);
-
-    if (device != AUDIO_DEVICE_NONE && mSupportedDevices.getDevice(device, address) == 0) {
-        return false;
-    }
-
-    if (samplingRate == 0) {
-         return false;
-    }
-    uint32_t myUpdatedSamplingRate = samplingRate;
-    if (isPlaybackThread && checkExactSamplingRate(samplingRate) != NO_ERROR) {
-         return false;
-    }
-    if (isRecordThread && checkCompatibleSamplingRate(samplingRate, &myUpdatedSamplingRate) !=
-            NO_ERROR) {
-         return false;
-    }
-
-    if (!audio_is_valid_format(format) || checkFormat(format) != NO_ERROR) {
-        return false;
-    }
-
-    if (isPlaybackThread && (!audio_is_output_channel(channelMask) ||
-            checkExactChannelMask(channelMask) != NO_ERROR)) {
-        return false;
-    }
-    if (isRecordThread && (!audio_is_input_channel(channelMask) ||
-            checkCompatibleChannelMask(channelMask) != NO_ERROR)) {
-        return false;
-    }
-
-    if (isPlaybackThread && (mFlags & flags) != flags) {
-        return false;
-    }
-    // The only input flag that is allowed to be different is the fast flag.
-    // An existing fast stream is compatible with a normal track request.
-    // An existing normal stream is compatible with a fast track request,
-    // but the fast request will be denied by AudioFlinger and converted to normal track.
-    if (isRecordThread && ((mFlags ^ flags) &
-            ~AUDIO_INPUT_FLAG_FAST)) {
-        return false;
-    }
-
-    if (updatedSamplingRate != NULL) {
-        *updatedSamplingRate = myUpdatedSamplingRate;
-    }
-    return true;
-}
-
-void AudioPolicyManager::IOProfile::dump(int fd)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    AudioPort::dump(fd, 4);
-
-    snprintf(buffer, SIZE, "    - flags: 0x%04x\n", mFlags);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "    - devices:\n");
-    result.append(buffer);
-    write(fd, result.string(), result.size());
-    for (size_t i = 0; i < mSupportedDevices.size(); i++) {
-        mSupportedDevices[i]->dump(fd, 6, i);
-    }
-}
-
-void AudioPolicyManager::IOProfile::log()
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    ALOGV("    - sampling rates: ");
-    for (size_t i = 0; i < mSamplingRates.size(); i++) {
-        ALOGV("  %d", mSamplingRates[i]);
-    }
-
-    ALOGV("    - channel masks: ");
-    for (size_t i = 0; i < mChannelMasks.size(); i++) {
-        ALOGV("  0x%04x", mChannelMasks[i]);
-    }
-
-    ALOGV("    - formats: ");
-    for (size_t i = 0; i < mFormats.size(); i++) {
-        ALOGV("  0x%08x", mFormats[i]);
-    }
-
-    ALOGV("    - devices: 0x%04x\n", mSupportedDevices.types());
-    ALOGV("    - flags: 0x%04x\n", mFlags);
-}
-
-
-// --- DeviceDescriptor implementation
-
-
-AudioPolicyManager::DeviceDescriptor::DeviceDescriptor(const String8& name, audio_devices_t type) :
-                     AudioPort(name, AUDIO_PORT_TYPE_DEVICE,
-                               audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
-                                                              AUDIO_PORT_ROLE_SOURCE,
-                             NULL),
-                     mDeviceType(type), mAddress(""), mId(0)
-{
-}
-
-bool AudioPolicyManager::DeviceDescriptor::equals(const sp<DeviceDescriptor>& other) const
-{
-    // Devices are considered equal if they:
-    // - are of the same type (a device type cannot be AUDIO_DEVICE_NONE)
-    // - have the same address or one device does not specify the address
-    // - have the same channel mask or one device does not specify the channel mask
-    return (mDeviceType == other->mDeviceType) &&
-           (mAddress == "" || other->mAddress == "" || mAddress == other->mAddress) &&
-           (mChannelMask == 0 || other->mChannelMask == 0 ||
-                mChannelMask == other->mChannelMask);
-}
-
-void AudioPolicyManager::DeviceDescriptor::loadGains(cnode *root)
-{
-    AudioPort::loadGains(root);
-    if (mGains.size() > 0) {
-        mGains[0]->getDefaultConfig(&mGain);
-    }
-}
-
-
-void AudioPolicyManager::DeviceVector::refreshTypes()
-{
-    mDeviceTypes = AUDIO_DEVICE_NONE;
-    for(size_t i = 0; i < size(); i++) {
-        mDeviceTypes |= itemAt(i)->mDeviceType;
-    }
-    ALOGV("DeviceVector::refreshTypes() mDeviceTypes %08x", mDeviceTypes);
-}
-
-ssize_t AudioPolicyManager::DeviceVector::indexOf(const sp<DeviceDescriptor>& item) const
-{
-    for(size_t i = 0; i < size(); i++) {
-        if (item->equals(itemAt(i))) {
-            return i;
-        }
-    }
-    return -1;
-}
-
-ssize_t AudioPolicyManager::DeviceVector::add(const sp<DeviceDescriptor>& item)
-{
-    ssize_t ret = indexOf(item);
-
-    if (ret < 0) {
-        ret = SortedVector::add(item);
-        if (ret >= 0) {
-            refreshTypes();
-        }
-    } else {
-        ALOGW("DeviceVector::add device %08x already in", item->mDeviceType);
-        ret = -1;
-    }
-    return ret;
-}
-
-ssize_t AudioPolicyManager::DeviceVector::remove(const sp<DeviceDescriptor>& item)
-{
-    size_t i;
-    ssize_t ret = indexOf(item);
-
-    if (ret < 0) {
-        ALOGW("DeviceVector::remove device %08x not in", item->mDeviceType);
-    } else {
-        ret = SortedVector::removeAt(ret);
-        if (ret >= 0) {
-            refreshTypes();
-        }
-    }
-    return ret;
-}
-
-void AudioPolicyManager::DeviceVector::loadDevicesFromType(audio_devices_t types)
-{
-    DeviceVector deviceList;
-
-    uint32_t role_bit = AUDIO_DEVICE_BIT_IN & types;
-    types &= ~role_bit;
-
-    while (types) {
-        uint32_t i = 31 - __builtin_clz(types);
-        uint32_t type = 1 << i;
-        types &= ~type;
-        add(new DeviceDescriptor(String8(""), type | role_bit));
-    }
-}
-
-void AudioPolicyManager::DeviceVector::loadDevicesFromName(char *name,
-                                                           const DeviceVector& declaredDevices)
-{
-    char *devName = strtok(name, "|");
-    while (devName != NULL) {
-        if (strlen(devName) != 0) {
-            audio_devices_t type = stringToEnum(sDeviceNameToEnumTable,
-                                 ARRAY_SIZE(sDeviceNameToEnumTable),
-                                 devName);
-            if (type != AUDIO_DEVICE_NONE) {
-                sp<DeviceDescriptor> dev = new DeviceDescriptor(String8(""), type);
-                if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX ||
-                        type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) {
-                    dev->mAddress = String8("0");
-                }
-                add(dev);
-            } else {
-                sp<DeviceDescriptor> deviceDesc =
-                        declaredDevices.getDeviceFromName(String8(devName));
-                if (deviceDesc != 0) {
-                    add(deviceDesc);
-                }
-            }
-         }
-         devName = strtok(NULL, "|");
-     }
-}
-
-sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDevice(
-                                                        audio_devices_t type, String8 address) const
-{
-    sp<DeviceDescriptor> device;
-    for (size_t i = 0; i < size(); i++) {
-        if (itemAt(i)->mDeviceType == type) {
-            if (address == "" || itemAt(i)->mAddress == address) {
-                device = itemAt(i);
-                if (itemAt(i)->mAddress == address) {
-                    break;
-                }
-            }
-        }
-    }
-    ALOGV("DeviceVector::getDevice() for type %08x address %s found %p",
-          type, address.string(), device.get());
-    return device;
-}
-
-sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDeviceFromId(
-                                                                    audio_port_handle_t id) const
-{
-    sp<DeviceDescriptor> device;
-    for (size_t i = 0; i < size(); i++) {
-        ALOGV("DeviceVector::getDeviceFromId(%d) itemAt(%zu)->mId %d", id, i, itemAt(i)->mId);
-        if (itemAt(i)->mId == id) {
-            device = itemAt(i);
-            break;
-        }
-    }
-    return device;
-}
-
-AudioPolicyManager::DeviceVector AudioPolicyManager::DeviceVector::getDevicesFromType(
-                                                                        audio_devices_t type) const
-{
-    DeviceVector devices;
-    for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) {
-        if (itemAt(i)->mDeviceType & type & ~AUDIO_DEVICE_BIT_IN) {
-            devices.add(itemAt(i));
-            type &= ~itemAt(i)->mDeviceType;
-            ALOGV("DeviceVector::getDevicesFromType() for type %x found %p",
-                  itemAt(i)->mDeviceType, itemAt(i).get());
-        }
-    }
-    return devices;
-}
-
-AudioPolicyManager::DeviceVector AudioPolicyManager::DeviceVector::getDevicesFromTypeAddr(
-        audio_devices_t type, String8 address) const
-{
-    DeviceVector devices;
-    for (size_t i = 0; i < size(); i++) {
-        if (itemAt(i)->mDeviceType == type) {
-            if (itemAt(i)->mAddress == address) {
-                devices.add(itemAt(i));
-            }
-        }
-    }
-    return devices;
-}
-
-sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDeviceFromName(
-        const String8& name) const
-{
-    sp<DeviceDescriptor> device;
-    for (size_t i = 0; i < size(); i++) {
-        if (itemAt(i)->mName == name) {
-            device = itemAt(i);
-            break;
-        }
-    }
-    return device;
-}
-
-void AudioPolicyManager::DeviceDescriptor::toAudioPortConfig(
-                                                    struct audio_port_config *dstConfig,
-                                                    const struct audio_port_config *srcConfig) const
-{
-    dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK|AUDIO_PORT_CONFIG_GAIN;
-    if (srcConfig != NULL) {
-        dstConfig->config_mask |= srcConfig->config_mask;
-    }
-
-    AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
-
-    dstConfig->id = mId;
-    dstConfig->role = audio_is_output_device(mDeviceType) ?
-                        AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
-    dstConfig->type = AUDIO_PORT_TYPE_DEVICE;
-    dstConfig->ext.device.type = mDeviceType;
-    dstConfig->ext.device.hw_module = mModule->mHandle;
-    strncpy(dstConfig->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
-}
-
-void AudioPolicyManager::DeviceDescriptor::toAudioPort(struct audio_port *port) const
-{
-    ALOGV("DeviceDescriptor::toAudioPort() handle %d type %x", mId, mDeviceType);
-    AudioPort::toAudioPort(port);
-    port->id = mId;
-    toAudioPortConfig(&port->active_config);
-    port->ext.device.type = mDeviceType;
-    port->ext.device.hw_module = mModule->mHandle;
-    strncpy(port->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
-}
-
-status_t AudioPolicyManager::DeviceDescriptor::dump(int fd, int spaces, int index) const
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, "%*sDevice %d:\n", spaces, "", index+1);
-    result.append(buffer);
-    if (mId != 0) {
-        snprintf(buffer, SIZE, "%*s- id: %2d\n", spaces, "", mId);
-        result.append(buffer);
-    }
-    snprintf(buffer, SIZE, "%*s- type: %-48s\n", spaces, "",
-                                              enumToString(sDeviceNameToEnumTable,
-                                                           ARRAY_SIZE(sDeviceNameToEnumTable),
-                                                           mDeviceType));
-    result.append(buffer);
-    if (mAddress.size() != 0) {
-        snprintf(buffer, SIZE, "%*s- address: %-32s\n", spaces, "", mAddress.string());
-        result.append(buffer);
-    }
-    write(fd, result.string(), result.size());
-    AudioPort::dump(fd, spaces);
-
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManager::AudioPatch::dump(int fd, int spaces, int index) const
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-
-    snprintf(buffer, SIZE, "%*sAudio patch %d:\n", spaces, "", index+1);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "%*s- handle: %2d\n", spaces, "", mHandle);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "%*s- audio flinger handle: %2d\n", spaces, "", mAfPatchHandle);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "%*s- owner uid: %2d\n", spaces, "", mUid);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "%*s- %d sources:\n", spaces, "", mPatch.num_sources);
-    result.append(buffer);
-    for (size_t i = 0; i < mPatch.num_sources; i++) {
-        if (mPatch.sources[i].type == AUDIO_PORT_TYPE_DEVICE) {
-            snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
-                     mPatch.sources[i].id, enumToString(sDeviceNameToEnumTable,
-                                                        ARRAY_SIZE(sDeviceNameToEnumTable),
-                                                        mPatch.sources[i].ext.device.type));
-        } else {
-            snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
-                     mPatch.sources[i].id, mPatch.sources[i].ext.mix.handle);
-        }
-        result.append(buffer);
-    }
-    snprintf(buffer, SIZE, "%*s- %d sinks:\n", spaces, "", mPatch.num_sinks);
-    result.append(buffer);
-    for (size_t i = 0; i < mPatch.num_sinks; i++) {
-        if (mPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE) {
-            snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
-                     mPatch.sinks[i].id, enumToString(sDeviceNameToEnumTable,
-                                                        ARRAY_SIZE(sDeviceNameToEnumTable),
-                                                        mPatch.sinks[i].ext.device.type));
-        } else {
-            snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
-                     mPatch.sinks[i].id, mPatch.sinks[i].ext.mix.handle);
-        }
-        result.append(buffer);
-    }
-
-    write(fd, result.string(), result.size());
-    return NO_ERROR;
-}
 
 // --- audio_policy.conf file parsing
-
-uint32_t AudioPolicyManager::parseOutputFlagNames(char *name)
-{
-    uint32_t flag = 0;
-
-    // it is OK to cast name to non const here as we are not going to use it after
-    // strtok() modifies it
-    char *flagName = strtok(name, "|");
-    while (flagName != NULL) {
-        if (strlen(flagName) != 0) {
-            flag |= stringToEnum(sOutputFlagNameToEnumTable,
-                               ARRAY_SIZE(sOutputFlagNameToEnumTable),
-                               flagName);
-        }
-        flagName = strtok(NULL, "|");
-    }
-    //force direct flag if offload flag is set: offloading implies a direct output stream
-    // and all common behaviors are driven by checking only the direct flag
-    // this should normally be set appropriately in the policy configuration file
-    if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
-        flag |= AUDIO_OUTPUT_FLAG_DIRECT;
-    }
-
-    return flag;
-}
-
-uint32_t AudioPolicyManager::parseInputFlagNames(char *name)
-{
-    uint32_t flag = 0;
-
-    // it is OK to cast name to non const here as we are not going to use it after
-    // strtok() modifies it
-    char *flagName = strtok(name, "|");
-    while (flagName != NULL) {
-        if (strlen(flagName) != 0) {
-            flag |= stringToEnum(sInputFlagNameToEnumTable,
-                               ARRAY_SIZE(sInputFlagNameToEnumTable),
-                               flagName);
-        }
-        flagName = strtok(NULL, "|");
-    }
-    return flag;
-}
-
-audio_devices_t AudioPolicyManager::parseDeviceNames(char *name)
-{
-    uint32_t device = 0;
-
-    char *devName = strtok(name, "|");
-    while (devName != NULL) {
-        if (strlen(devName) != 0) {
-            device |= stringToEnum(sDeviceNameToEnumTable,
-                                 ARRAY_SIZE(sDeviceNameToEnumTable),
-                                 devName);
-         }
-        devName = strtok(NULL, "|");
-     }
-    return device;
-}
-
+// TODO candidate to be moved to ConfigParsingUtils
 void AudioPolicyManager::loadHwModule(cnode *root)
 {
     status_t status = NAME_NOT_FOUND;
@@ -7889,6 +5557,7 @@
     }
 }
 
+// TODO candidate to be moved to ConfigParsingUtils
 void AudioPolicyManager::loadHwModules(cnode *root)
 {
     cnode *node = config_find(root, AUDIO_HW_MODULE_TAG);
@@ -7904,6 +5573,7 @@
     }
 }
 
+// TODO candidate to be moved to ConfigParsingUtils
 void AudioPolicyManager::loadGlobalConfig(cnode *root, const sp<HwModule>& module)
 {
     cnode *node = config_find(root, GLOBAL_CONFIG_TAG);
@@ -7924,11 +5594,12 @@
             ALOGV("loadGlobalConfig() Attached Output Devices %08x",
                   mAvailableOutputDevices.types());
         } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) {
-            audio_devices_t device = (audio_devices_t)stringToEnum(sDeviceNameToEnumTable,
-                                              ARRAY_SIZE(sDeviceNameToEnumTable),
-                                              (char *)node->value);
+            audio_devices_t device = (audio_devices_t)ConfigParsingUtils::stringToEnum(
+                    sDeviceNameToEnumTable,
+                    ARRAY_SIZE(sDeviceNameToEnumTable),
+                    (char *)node->value);
             if (device != AUDIO_DEVICE_NONE) {
-                mDefaultOutputDevice = new DeviceDescriptor(String8(""), device);
+                mDefaultOutputDevice = new DeviceDescriptor(String8("default-output"), device);
             } else {
                 ALOGW("loadGlobalConfig() default device not specified");
             }
@@ -7938,7 +5609,7 @@
                                                        declaredDevices);
             ALOGV("loadGlobalConfig() Available InputDevices %08x", mAvailableInputDevices.types());
         } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) {
-            mSpeakerDrcEnabled = stringToBool((char *)node->value);
+            mSpeakerDrcEnabled = ConfigParsingUtils::stringToBool((char *)node->value);
             ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled);
         } else if (strcmp(AUDIO_HAL_VERSION_TAG, node->name) == 0) {
             uint32_t major, minor;
@@ -7951,6 +5622,7 @@
     }
 }
 
+// TODO candidate to be moved to ConfigParsingUtils
 status_t AudioPolicyManager::loadAudioPolicyConfig(const char *path)
 {
     cnode *root;
@@ -7979,8 +5651,8 @@
 {
     sp<HwModule> module;
     sp<IOProfile> profile;
-    sp<DeviceDescriptor> defaultInputDevice = new DeviceDescriptor(String8(""),
-                                                                   AUDIO_DEVICE_IN_BUILTIN_MIC);
+    sp<DeviceDescriptor> defaultInputDevice =
+                    new DeviceDescriptor(String8("builtin-mic"), AUDIO_DEVICE_IN_BUILTIN_MIC);
     mAvailableOutputDevices.add(mDefaultOutputDevice);
     mAvailableInputDevices.add(defaultInputDevice);
 
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
new file mode 100644
index 0000000..61ea6f2
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -0,0 +1,560 @@
+/*
+ * Copyright (C) 2009 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 <stdint.h>
+#include <sys/types.h>
+#include <cutils/config_utils.h>
+#include <cutils/misc.h>
+#include <utils/Timers.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <utils/SortedVector.h>
+#include <media/AudioPolicy.h>
+#include "AudioPolicyInterface.h"
+
+#include "Gains.h"
+#include "Ports.h"
+#include "ConfigParsingUtils.h"
+#include "Devices.h"
+#include "IOProfile.h"
+#include "HwModule.h"
+#include "AudioInputDescriptor.h"
+#include "AudioOutputDescriptor.h"
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
+#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5
+// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB
+#define SONIFICATION_HEADSET_VOLUME_MIN  0.016
+// Time in milliseconds during which we consider that music is still active after a music
+// track was stopped - see computeVolume()
+#define SONIFICATION_HEADSET_MUSIC_DELAY  5000
+// Time in milliseconds after media stopped playing during which we consider that the
+// sonification should be as unobtrusive as during the time media was playing.
+#define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000
+// Time in milliseconds during witch some streams are muted while the audio path
+// is switched
+#define MUTE_TIME_MS 2000
+
+#define NUM_TEST_OUTPUTS 5
+
+#define NUM_VOL_CURVE_KNEES 2
+
+// Default minimum length allowed for offloading a compressed track
+// Can be overridden by the audio.offload.min.duration.secs property
+#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60
+
+#define MAX_MIXER_SAMPLING_RATE 48000
+#define MAX_MIXER_CHANNEL_COUNT 8
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManager implements audio policy manager behavior common to all platforms.
+// ----------------------------------------------------------------------------
+
+class AudioPolicyManager: public AudioPolicyInterface
+#ifdef AUDIO_POLICY_TEST
+    , public Thread
+#endif //AUDIO_POLICY_TEST
+{
+
+public:
+                AudioPolicyManager(AudioPolicyClientInterface *clientInterface);
+        virtual ~AudioPolicyManager();
+
+        // AudioPolicyInterface
+        virtual status_t setDeviceConnectionState(audio_devices_t device,
+                                                          audio_policy_dev_state_t state,
+                                                          const char *device_address,
+                                                          const char *device_name);
+        virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
+                                                                              const char *device_address);
+        virtual void setPhoneState(audio_mode_t state);
+        virtual void setForceUse(audio_policy_force_use_t usage,
+                                 audio_policy_forced_cfg_t config);
+        virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
+        virtual void setSystemProperty(const char* property, const char* value);
+        virtual status_t initCheck();
+        virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
+                                            uint32_t samplingRate,
+                                            audio_format_t format,
+                                            audio_channel_mask_t channelMask,
+                                            audio_output_flags_t flags,
+                                            const audio_offload_info_t *offloadInfo);
+        virtual status_t getOutputForAttr(const audio_attributes_t *attr,
+                                          audio_io_handle_t *output,
+                                          audio_session_t session,
+                                          audio_stream_type_t *stream,
+                                          uint32_t samplingRate,
+                                          audio_format_t format,
+                                          audio_channel_mask_t channelMask,
+                                          audio_output_flags_t flags,
+                                          const audio_offload_info_t *offloadInfo);
+        virtual status_t startOutput(audio_io_handle_t output,
+                                     audio_stream_type_t stream,
+                                     audio_session_t session);
+        virtual status_t stopOutput(audio_io_handle_t output,
+                                    audio_stream_type_t stream,
+                                    audio_session_t session);
+        virtual void releaseOutput(audio_io_handle_t output,
+                                   audio_stream_type_t stream,
+                                   audio_session_t session);
+        virtual status_t getInputForAttr(const audio_attributes_t *attr,
+                                         audio_io_handle_t *input,
+                                         audio_session_t session,
+                                         uint32_t samplingRate,
+                                         audio_format_t format,
+                                         audio_channel_mask_t channelMask,
+                                         audio_input_flags_t flags,
+                                         input_type_t *inputType);
+
+        // indicates to the audio policy manager that the input starts being used.
+        virtual status_t startInput(audio_io_handle_t input,
+                                    audio_session_t session);
+
+        // indicates to the audio policy manager that the input stops being used.
+        virtual status_t stopInput(audio_io_handle_t input,
+                                   audio_session_t session);
+        virtual void releaseInput(audio_io_handle_t input,
+                                  audio_session_t session);
+        virtual void closeAllInputs();
+        virtual void initStreamVolume(audio_stream_type_t stream,
+                                                    int indexMin,
+                                                    int indexMax);
+        virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
+                                              int index,
+                                              audio_devices_t device);
+        virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
+                                              int *index,
+                                              audio_devices_t device);
+
+        // return the strategy corresponding to a given stream type
+        virtual uint32_t getStrategyForStream(audio_stream_type_t stream);
+        // return the strategy corresponding to the given audio attributes
+        virtual uint32_t getStrategyForAttr(const audio_attributes_t *attr);
+
+        // return the enabled output devices for the given stream type
+        virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream);
+
+        virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL);
+        virtual status_t registerEffect(const effect_descriptor_t *desc,
+                                        audio_io_handle_t io,
+                                        uint32_t strategy,
+                                        int session,
+                                        int id);
+        virtual status_t unregisterEffect(int id);
+        virtual status_t setEffectEnabled(int id, bool enabled);
+
+        virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
+        // return whether a stream is playing remotely, override to change the definition of
+        //   local/remote playback, used for instance by notification manager to not make
+        //   media players lose audio focus when not playing locally
+        //   For the base implementation, "remotely" means playing during screen mirroring which
+        //   uses an output for playback with a non-empty, non "0" address.
+        virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
+        virtual bool isSourceActive(audio_source_t source) const;
+
+        virtual status_t dump(int fd);
+
+        virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
+
+        virtual status_t listAudioPorts(audio_port_role_t role,
+                                        audio_port_type_t type,
+                                        unsigned int *num_ports,
+                                        struct audio_port *ports,
+                                        unsigned int *generation);
+        virtual status_t getAudioPort(struct audio_port *port);
+        virtual status_t createAudioPatch(const struct audio_patch *patch,
+                                           audio_patch_handle_t *handle,
+                                           uid_t uid);
+        virtual status_t releaseAudioPatch(audio_patch_handle_t handle,
+                                              uid_t uid);
+        virtual status_t listAudioPatches(unsigned int *num_patches,
+                                          struct audio_patch *patches,
+                                          unsigned int *generation);
+        virtual status_t setAudioPortConfig(const struct audio_port_config *config);
+        virtual void clearAudioPatches(uid_t uid);
+
+        virtual status_t acquireSoundTriggerSession(audio_session_t *session,
+                                               audio_io_handle_t *ioHandle,
+                                               audio_devices_t *device);
+
+        virtual status_t releaseSoundTriggerSession(audio_session_t session);
+
+        virtual status_t registerPolicyMixes(Vector<AudioMix> mixes);
+        virtual status_t unregisterPolicyMixes(Vector<AudioMix> mixes);
+
+        // Audio policy configuration file parsing (audio_policy.conf)
+        // TODO candidates to be moved to ConfigParsingUtils
+                void loadHwModule(cnode *root);
+                void loadHwModules(cnode *root);
+                void loadGlobalConfig(cnode *root, const sp<HwModule>& module);
+                status_t loadAudioPolicyConfig(const char *path);
+                void defaultAudioPolicyConfig(void);
+
+                // return the strategy corresponding to a given stream type
+                static routing_strategy getStrategy(audio_stream_type_t stream);
+
+                static uint32_t nextUniqueId();
+protected:
+
+        class EffectDescriptor : public RefBase
+        {
+        public:
+
+            status_t dump(int fd);
+
+            int mIo;                // io the effect is attached to
+            routing_strategy mStrategy; // routing strategy the effect is associated to
+            int mSession;               // audio session the effect is on
+            effect_descriptor_t mDesc;  // effect descriptor
+            bool mEnabled;              // enabled state: CPU load being used or not
+        };
+
+        void addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc);
+        void addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc);
+
+        // return appropriate device for streams handled by the specified strategy according to current
+        // phone state, connected devices...
+        // if fromCache is true, the device is returned from mDeviceForStrategy[],
+        // otherwise it is determine by current state
+        // (device connected,phone state, force use, a2dp output...)
+        // This allows to:
+        //  1 speed up process when the state is stable (when starting or stopping an output)
+        //  2 access to either current device selection (fromCache == true) or
+        // "future" device selection (fromCache == false) when called from a context
+        //  where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
+        //  before updateDevicesAndOutputs() is called.
+        virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy,
+                                                     bool fromCache);
+
+        // change the route of the specified output. Returns the number of ms we have slept to
+        // allow new routing to take effect in certain cases.
+        virtual uint32_t setOutputDevice(audio_io_handle_t output,
+                             audio_devices_t device,
+                             bool force = false,
+                             int delayMs = 0,
+                             audio_patch_handle_t *patchHandle = NULL,
+                             const char* address = NULL);
+        status_t resetOutputDevice(audio_io_handle_t output,
+                                   int delayMs = 0,
+                                   audio_patch_handle_t *patchHandle = NULL);
+        status_t setInputDevice(audio_io_handle_t input,
+                                audio_devices_t device,
+                                bool force = false,
+                                audio_patch_handle_t *patchHandle = NULL);
+        status_t resetInputDevice(audio_io_handle_t input,
+                                  audio_patch_handle_t *patchHandle = NULL);
+
+        // select input device corresponding to requested audio source
+        virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource);
+
+        // return io handle of active input or 0 if no input is active
+        //    Only considers inputs from physical devices (e.g. main mic, headset mic) when
+        //    ignoreVirtualInputs is true.
+        audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true);
+
+        uint32_t activeInputsCount() const;
+
+        // initialize volume curves for each strategy and device category
+        void initializeVolumeCurves();
+
+        // compute the actual volume for a given stream according to the requested index and a particular
+        // device
+        virtual float computeVolume(audio_stream_type_t stream, int index,
+                                    audio_io_handle_t output, audio_devices_t device);
+
+        // check that volume change is permitted, compute and send new volume to audio hardware
+        virtual status_t checkAndSetVolume(audio_stream_type_t stream, int index,
+                                           audio_io_handle_t output,
+                                           audio_devices_t device,
+                                           int delayMs = 0, bool force = false);
+
+        // apply all stream volumes to the specified output and device
+        void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
+
+        // Mute or unmute all streams handled by the specified strategy on the specified output
+        void setStrategyMute(routing_strategy strategy,
+                             bool on,
+                             audio_io_handle_t output,
+                             int delayMs = 0,
+                             audio_devices_t device = (audio_devices_t)0);
+
+        // Mute or unmute the stream on the specified output
+        void setStreamMute(audio_stream_type_t stream,
+                           bool on,
+                           audio_io_handle_t output,
+                           int delayMs = 0,
+                           audio_devices_t device = (audio_devices_t)0);
+
+        // handle special cases for sonification strategy while in call: mute streams or replace by
+        // a special tone in the device used for communication
+        void handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange);
+
+        // true if device is in a telephony or VoIP call
+        virtual bool isInCall();
+
+        // true if given state represents a device in a telephony or VoIP call
+        virtual bool isStateInCall(int state);
+
+        // when a device is connected, checks if an open output can be routed
+        // to this device. If none is open, tries to open one of the available outputs.
+        // Returns an output suitable to this device or 0.
+        // when a device is disconnected, checks if an output is not used any more and
+        // returns its handle if any.
+        // transfers the audio tracks and effects from one output thread to another accordingly.
+        status_t checkOutputsForDevice(const sp<DeviceDescriptor> devDesc,
+                                       audio_policy_dev_state_t state,
+                                       SortedVector<audio_io_handle_t>& outputs,
+                                       const String8 address);
+
+        status_t checkInputsForDevice(audio_devices_t device,
+                                      audio_policy_dev_state_t state,
+                                      SortedVector<audio_io_handle_t>& inputs,
+                                      const String8 address);
+
+        // close an output and its companion duplicating output.
+        void closeOutput(audio_io_handle_t output);
+
+        // close an input.
+        void closeInput(audio_io_handle_t input);
+
+        // checks and if necessary changes outputs used for all strategies.
+        // must be called every time a condition that affects the output choice for a given strategy
+        // changes: connected device, phone state, force use...
+        // Must be called before updateDevicesAndOutputs()
+        void checkOutputForStrategy(routing_strategy strategy);
+
+        // Same as checkOutputForStrategy() but for a all strategies in order of priority
+        void checkOutputForAllStrategies();
+
+        // manages A2DP output suspend/restore according to phone state and BT SCO usage
+        void checkA2dpSuspend();
+
+        // returns the A2DP output handle if it is open or 0 otherwise
+        audio_io_handle_t getA2dpOutput();
+
+        // selects the most appropriate device on output for current state
+        // must be called every time a condition that affects the device choice for a given output is
+        // changed: connected device, phone state, force use, output start, output stop..
+        // see getDeviceForStrategy() for the use of fromCache parameter
+        audio_devices_t getNewOutputDevice(audio_io_handle_t output, bool fromCache);
+
+        // updates cache of device used by all strategies (mDeviceForStrategy[])
+        // must be called every time a condition that affects the device choice for a given strategy is
+        // changed: connected device, phone state, force use...
+        // cached values are used by getDeviceForStrategy() if parameter fromCache is true.
+         // Must be called after checkOutputForAllStrategies()
+        void updateDevicesAndOutputs();
+
+        // selects the most appropriate device on input for current state
+        audio_devices_t getNewInputDevice(audio_io_handle_t input);
+
+        virtual uint32_t getMaxEffectsCpuLoad();
+        virtual uint32_t getMaxEffectsMemory();
+#ifdef AUDIO_POLICY_TEST
+        virtual     bool        threadLoop();
+                    void        exit();
+        int testOutputIndex(audio_io_handle_t output);
+#endif //AUDIO_POLICY_TEST
+
+        status_t setEffectEnabled(const sp<EffectDescriptor>& effectDesc, bool enabled);
+
+        SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
+                        DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > openOutputs);
+        bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
+                                           SortedVector<audio_io_handle_t>& outputs2);
+
+        // mute/unmute strategies using an incompatible device combination
+        // if muting, wait for the audio in pcm buffer to be drained before proceeding
+        // if unmuting, unmute only after the specified delay
+        // Returns the number of ms waited
+        virtual uint32_t  checkDeviceMuteStrategies(sp<AudioOutputDescriptor> outputDesc,
+                                            audio_devices_t prevDevice,
+                                            uint32_t delayMs);
+
+        audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
+                                       audio_output_flags_t flags,
+                                       audio_format_t format);
+        // samplingRate parameter is an in/out and so may be modified
+        sp<IOProfile> getInputProfile(audio_devices_t device,
+                                      String8 address,
+                                      uint32_t& samplingRate,
+                                      audio_format_t format,
+                                      audio_channel_mask_t channelMask,
+                                      audio_input_flags_t flags);
+        sp<IOProfile> getProfileForDirectOutput(audio_devices_t device,
+                                                       uint32_t samplingRate,
+                                                       audio_format_t format,
+                                                       audio_channel_mask_t channelMask,
+                                                       audio_output_flags_t flags);
+
+        audio_io_handle_t selectOutputForEffects(const SortedVector<audio_io_handle_t>& outputs);
+
+        bool isNonOffloadableEffectEnabled();
+
+        virtual status_t addAudioPatch(audio_patch_handle_t handle,
+                               const sp<AudioPatch>& patch);
+        virtual status_t removeAudioPatch(audio_patch_handle_t handle);
+
+        sp<AudioOutputDescriptor> getOutputFromId(audio_port_handle_t id) const;
+        sp<AudioInputDescriptor> getInputFromId(audio_port_handle_t id) const;
+        sp<HwModule> getModuleForDevice(audio_devices_t device) const;
+        sp<HwModule> getModuleFromName(const char *name) const;
+        audio_devices_t availablePrimaryOutputDevices();
+        audio_devices_t availablePrimaryInputDevices();
+
+        void updateCallRouting(audio_devices_t rxDevice, int delayMs = 0);
+
+
+        uid_t mUidCached;
+        AudioPolicyClientInterface *mpClientInterface;  // audio policy client interface
+        audio_io_handle_t mPrimaryOutput;              // primary output handle
+        // list of descriptors for outputs currently opened
+        DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mOutputs;
+        // copy of mOutputs before setDeviceConnectionState() opens new outputs
+        // reset to mOutputs when updateDevicesAndOutputs() is called.
+        DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mPreviousOutputs;
+        DefaultKeyedVector<audio_io_handle_t, sp<AudioInputDescriptor> > mInputs;     // list of input descriptors
+        DeviceVector  mAvailableOutputDevices; // all available output devices
+        DeviceVector  mAvailableInputDevices;  // all available input devices
+        int mPhoneState;                                                    // current phone state
+        audio_policy_forced_cfg_t mForceUse[AUDIO_POLICY_FORCE_USE_CNT];   // current forced use configuration
+
+        StreamDescriptor mStreams[AUDIO_STREAM_CNT];           // stream descriptors for volume control
+        bool    mLimitRingtoneVolume;                                       // limit ringtone volume to music volume if headset connected
+        audio_devices_t mDeviceForStrategy[NUM_STRATEGIES];
+        float   mLastVoiceVolume;                                           // last voice volume value sent to audio HAL
+
+        // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units
+        static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000;
+        // Maximum memory allocated to audio effects in KB
+        static const uint32_t MAX_EFFECTS_MEMORY = 512;
+        uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects
+        uint32_t mTotalEffectsMemory;  // current memory used by effects
+        KeyedVector<int, sp<EffectDescriptor> > mEffects;  // list of registered audio effects
+        bool    mA2dpSuspended;  // true if A2DP output is suspended
+        sp<DeviceDescriptor> mDefaultOutputDevice; // output device selected by default at boot time
+        bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path
+                                // to boost soft sounds, used to adjust volume curves accordingly
+
+        Vector < sp<HwModule> > mHwModules;
+        static volatile int32_t mNextUniqueId;
+        volatile int32_t mAudioPortGeneration;
+
+        DefaultKeyedVector<audio_patch_handle_t, sp<AudioPatch> > mAudioPatches;
+
+        DefaultKeyedVector<audio_session_t, audio_io_handle_t> mSoundTriggerSessions;
+
+        sp<AudioPatch> mCallTxPatch;
+        sp<AudioPatch> mCallRxPatch;
+
+        // for supporting "beacon" streams, i.e. streams that only play on speaker, and never
+        // when something other than STREAM_TTS (a.k.a. "Transmitted Through Speaker") is playing
+        enum {
+            STARTING_OUTPUT,
+            STARTING_BEACON,
+            STOPPING_OUTPUT,
+            STOPPING_BEACON
+        };
+        uint32_t mBeaconMuteRefCount;   // ref count for stream that would mute beacon
+        uint32_t mBeaconPlayingRefCount;// ref count for the playing beacon streams
+        bool mBeaconMuted;              // has STREAM_TTS been muted
+
+        // custom mix entry in mPolicyMixes
+        class AudioPolicyMix : public RefBase {
+        public:
+            AudioPolicyMix() {}
+
+            AudioMix    mMix;                   // Audio policy mix descriptor
+            sp<AudioOutputDescriptor> mOutput;  // Corresponding output stream
+        };
+        DefaultKeyedVector<String8, sp<AudioPolicyMix> > mPolicyMixes; // list of registered mixes
+
+
+#ifdef AUDIO_POLICY_TEST
+        Mutex   mLock;
+        Condition mWaitWorkCV;
+
+        int             mCurOutput;
+        bool            mDirectOutput;
+        audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS];
+        int             mTestInput;
+        uint32_t        mTestDevice;
+        uint32_t        mTestSamplingRate;
+        uint32_t        mTestFormat;
+        uint32_t        mTestChannels;
+        uint32_t        mTestLatencyMs;
+#endif //AUDIO_POLICY_TEST
+
+        static bool isVirtualInputDevice(audio_devices_t device);
+
+        uint32_t nextAudioPortGeneration();
+private:
+        // updates device caching and output for streams that can influence the
+        //    routing of notifications
+        void handleNotificationRoutingForStream(audio_stream_type_t stream);
+        static bool deviceDistinguishesOnAddress(audio_devices_t device);
+        // find the outputs on a given output descriptor that have the given address.
+        // to be called on an AudioOutputDescriptor whose supported devices (as defined
+        //   in mProfile->mSupportedDevices) matches the device whose address is to be matched.
+        // see deviceDistinguishesOnAddress(audio_devices_t) for whether the device type is one
+        //   where addresses are used to distinguish between one connected device and another.
+        void findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/,
+                const audio_devices_t device /*in*/,
+                const String8 address /*in*/,
+                SortedVector<audio_io_handle_t>& outputs /*out*/);
+        uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; }
+        // internal method to return the output handle for the given device and format
+        audio_io_handle_t getOutputForDevice(
+                audio_devices_t device,
+                audio_session_t session,
+                audio_stream_type_t stream,
+                uint32_t samplingRate,
+                audio_format_t format,
+                audio_channel_mask_t channelMask,
+                audio_output_flags_t flags,
+                const audio_offload_info_t *offloadInfo);
+        // internal function to derive a stream type value from audio attributes
+        audio_stream_type_t streamTypefromAttributesInt(const audio_attributes_t *attr);
+        // return true if any output is playing anything besides the stream to ignore
+        bool isAnyOutputActive(audio_stream_type_t streamToIgnore);
+        // event is one of STARTING_OUTPUT, STARTING_BEACON, STOPPING_OUTPUT, STOPPING_BEACON
+        // returns 0 if no mute/unmute event happened, the largest latency of the device where
+        //   the mute/unmute happened
+        uint32_t handleEventForBeacon(int event);
+        uint32_t setBeaconMute(bool mute);
+        bool     isValidAttributes(const audio_attributes_t *paa);
+
+        // select input device corresponding to requested audio source and return associated policy
+        // mix if any. Calls getDeviceForInputSource().
+        audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource,
+                                                        AudioMix **policyMix = NULL);
+
+        // Called by setDeviceConnectionState().
+        status_t setDeviceConnectionStateInt(audio_devices_t device,
+                                                          audio_policy_dev_state_t state,
+                                                          const char *device_address,
+                                                          const char *device_name);
+        sp<DeviceDescriptor>  getDeviceDescriptor(const audio_devices_t device,
+                                                  const char *device_address,
+                                                  const char *device_name);
+};
+
+};
diff --git a/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp b/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp
new file mode 100644
index 0000000..1afd487
--- /dev/null
+++ b/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp
@@ -0,0 +1,121 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::ConfigParsingUtils"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+//static
+uint32_t ConfigParsingUtils::stringToEnum(const struct StringToEnum *table,
+                                              size_t size,
+                                              const char *name)
+{
+    for (size_t i = 0; i < size; i++) {
+        if (strcmp(table[i].name, name) == 0) {
+            ALOGV("stringToEnum() found %s", table[i].name);
+            return table[i].value;
+        }
+    }
+    return 0;
+}
+
+//static
+const char *ConfigParsingUtils::enumToString(const struct StringToEnum *table,
+                                              size_t size,
+                                              uint32_t value)
+{
+    for (size_t i = 0; i < size; i++) {
+        if (table[i].value == value) {
+            return table[i].name;
+        }
+    }
+    return "";
+}
+
+//static
+bool ConfigParsingUtils::stringToBool(const char *value)
+{
+    return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0));
+}
+
+
+// --- audio_policy.conf file parsing
+//static
+uint32_t ConfigParsingUtils::parseOutputFlagNames(char *name)
+{
+    uint32_t flag = 0;
+
+    // it is OK to cast name to non const here as we are not going to use it after
+    // strtok() modifies it
+    char *flagName = strtok(name, "|");
+    while (flagName != NULL) {
+        if (strlen(flagName) != 0) {
+            flag |= ConfigParsingUtils::stringToEnum(sOutputFlagNameToEnumTable,
+                               ARRAY_SIZE(sOutputFlagNameToEnumTable),
+                               flagName);
+        }
+        flagName = strtok(NULL, "|");
+    }
+    //force direct flag if offload flag is set: offloading implies a direct output stream
+    // and all common behaviors are driven by checking only the direct flag
+    // this should normally be set appropriately in the policy configuration file
+    if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+        flag |= AUDIO_OUTPUT_FLAG_DIRECT;
+    }
+
+    return flag;
+}
+
+//static
+uint32_t ConfigParsingUtils::parseInputFlagNames(char *name)
+{
+    uint32_t flag = 0;
+
+    // it is OK to cast name to non const here as we are not going to use it after
+    // strtok() modifies it
+    char *flagName = strtok(name, "|");
+    while (flagName != NULL) {
+        if (strlen(flagName) != 0) {
+            flag |= stringToEnum(sInputFlagNameToEnumTable,
+                               ARRAY_SIZE(sInputFlagNameToEnumTable),
+                               flagName);
+        }
+        flagName = strtok(NULL, "|");
+    }
+    return flag;
+}
+
+//static
+audio_devices_t ConfigParsingUtils::parseDeviceNames(char *name)
+{
+    uint32_t device = 0;
+
+    char *devName = strtok(name, "|");
+    while (devName != NULL) {
+        if (strlen(devName) != 0) {
+            device |= stringToEnum(sDeviceNameToEnumTable,
+                                 ARRAY_SIZE(sDeviceNameToEnumTable),
+                                 devName);
+         }
+        devName = strtok(NULL, "|");
+     }
+    return device;
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/ConfigParsingUtils.h b/services/audiopolicy/managerdefault/ConfigParsingUtils.h
new file mode 100644
index 0000000..7969661
--- /dev/null
+++ b/services/audiopolicy/managerdefault/ConfigParsingUtils.h
@@ -0,0 +1,159 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+// Definitions for audio_policy.conf file parsing
+// ----------------------------------------------------------------------------
+
+struct StringToEnum {
+    const char *name;
+    uint32_t value;
+};
+
+#define STRING_TO_ENUM(string) { #string, string }
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+
+const StringToEnum sDeviceNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER_SAFE),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_TELEPHONY_TX),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_LINE),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI_ARC),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPDIF),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_FM),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_LINE),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_AMBIENT),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_ALL_SCO),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_HDMI),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_TELEPHONY_RX),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_DEVICE),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_FM_TUNER),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_TV_TUNER),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_LINE),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_SPDIF),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_LOOPBACK),
+};
+
+const StringToEnum sOutputFlagNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_HW_AV_SYNC),
+};
+
+const StringToEnum sInputFlagNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_INPUT_FLAG_FAST),
+    STRING_TO_ENUM(AUDIO_INPUT_FLAG_HW_HOTWORD),
+};
+
+const StringToEnum sFormatNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
+    STRING_TO_ENUM(AUDIO_FORMAT_MP3),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_MAIN),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_LC),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_SSR),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_LTP),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V1),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_SCALABLE),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_ERLC),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_LD),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V2),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_ELD),
+    STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
+    STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1),
+    STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2),
+    STRING_TO_ENUM(AUDIO_FORMAT_OPUS),
+    STRING_TO_ENUM(AUDIO_FORMAT_AC3),
+    STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
+};
+
+const StringToEnum sOutChannelsNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
+};
+
+const StringToEnum sInChannelsNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
+};
+
+const StringToEnum sGainModeNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_GAIN_MODE_JOINT),
+    STRING_TO_ENUM(AUDIO_GAIN_MODE_CHANNELS),
+    STRING_TO_ENUM(AUDIO_GAIN_MODE_RAMP),
+};
+
+class ConfigParsingUtils
+{
+public:
+    static uint32_t stringToEnum(const struct StringToEnum *table,
+            size_t size,
+            const char *name);
+    static const char *enumToString(const struct StringToEnum *table,
+            size_t size,
+            uint32_t value);
+    static bool stringToBool(const char *value);
+    static uint32_t parseOutputFlagNames(char *name);
+    static uint32_t parseInputFlagNames(char *name);
+    static audio_devices_t parseDeviceNames(char *name);
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Devices.cpp b/services/audiopolicy/managerdefault/Devices.cpp
new file mode 100644
index 0000000..13c8bbc
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Devices.cpp
@@ -0,0 +1,282 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::Devices"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+String8 DeviceDescriptor::emptyNameStr = String8("");
+
+DeviceDescriptor::DeviceDescriptor(const String8& name, audio_devices_t type) :
+                     AudioPort(name, AUDIO_PORT_TYPE_DEVICE,
+                               audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
+                                                              AUDIO_PORT_ROLE_SOURCE,
+                             NULL),
+                     mDeviceType(type), mAddress("")
+{
+
+}
+
+bool DeviceDescriptor::equals(const sp<DeviceDescriptor>& other) const
+{
+    // Devices are considered equal if they:
+    // - are of the same type (a device type cannot be AUDIO_DEVICE_NONE)
+    // - have the same address or one device does not specify the address
+    // - have the same channel mask or one device does not specify the channel mask
+    return (mDeviceType == other->mDeviceType) &&
+           (mAddress == "" || other->mAddress == "" || mAddress == other->mAddress) &&
+           (mChannelMask == 0 || other->mChannelMask == 0 ||
+                mChannelMask == other->mChannelMask);
+}
+
+void DeviceDescriptor::loadGains(cnode *root)
+{
+    AudioPort::loadGains(root);
+    if (mGains.size() > 0) {
+        mGains[0]->getDefaultConfig(&mGain);
+    }
+}
+
+void DeviceVector::refreshTypes()
+{
+    mDeviceTypes = AUDIO_DEVICE_NONE;
+    for(size_t i = 0; i < size(); i++) {
+        mDeviceTypes |= itemAt(i)->mDeviceType;
+    }
+    ALOGV("DeviceVector::refreshTypes() mDeviceTypes %08x", mDeviceTypes);
+}
+
+ssize_t DeviceVector::indexOf(const sp<DeviceDescriptor>& item) const
+{
+    for(size_t i = 0; i < size(); i++) {
+        if (item->equals(itemAt(i))) {
+            return i;
+        }
+    }
+    return -1;
+}
+
+ssize_t DeviceVector::add(const sp<DeviceDescriptor>& item)
+{
+    ssize_t ret = indexOf(item);
+
+    if (ret < 0) {
+        ret = SortedVector::add(item);
+        if (ret >= 0) {
+            refreshTypes();
+        }
+    } else {
+        ALOGW("DeviceVector::add device %08x already in", item->mDeviceType);
+        ret = -1;
+    }
+    return ret;
+}
+
+ssize_t DeviceVector::remove(const sp<DeviceDescriptor>& item)
+{
+    size_t i;
+    ssize_t ret = indexOf(item);
+
+    if (ret < 0) {
+        ALOGW("DeviceVector::remove device %08x not in", item->mDeviceType);
+    } else {
+        ret = SortedVector::removeAt(ret);
+        if (ret >= 0) {
+            refreshTypes();
+        }
+    }
+    return ret;
+}
+
+void DeviceVector::loadDevicesFromType(audio_devices_t types)
+{
+    DeviceVector deviceList;
+
+    uint32_t role_bit = AUDIO_DEVICE_BIT_IN & types;
+    types &= ~role_bit;
+
+    while (types) {
+        uint32_t i = 31 - __builtin_clz(types);
+        uint32_t type = 1 << i;
+        types &= ~type;
+        add(new DeviceDescriptor(String8("device_type"), type | role_bit));
+    }
+}
+
+void DeviceVector::loadDevicesFromName(char *name,
+                                       const DeviceVector& declaredDevices)
+{
+    char *devName = strtok(name, "|");
+    while (devName != NULL) {
+        if (strlen(devName) != 0) {
+            audio_devices_t type = ConfigParsingUtils::stringToEnum(sDeviceNameToEnumTable,
+                                 ARRAY_SIZE(sDeviceNameToEnumTable),
+                                 devName);
+            if (type != AUDIO_DEVICE_NONE) {
+                sp<DeviceDescriptor> dev = new DeviceDescriptor(String8(name), type);
+                if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX ||
+                        type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) {
+                    dev->mAddress = String8("0");
+                }
+                add(dev);
+            } else {
+                sp<DeviceDescriptor> deviceDesc =
+                        declaredDevices.getDeviceFromName(String8(devName));
+                if (deviceDesc != 0) {
+                    add(deviceDesc);
+                }
+            }
+         }
+         devName = strtok(NULL, "|");
+     }
+}
+
+sp<DeviceDescriptor> DeviceVector::getDevice(audio_devices_t type, String8 address) const
+{
+    sp<DeviceDescriptor> device;
+    for (size_t i = 0; i < size(); i++) {
+        if (itemAt(i)->mDeviceType == type) {
+            if (address == "" || itemAt(i)->mAddress == address) {
+                device = itemAt(i);
+                if (itemAt(i)->mAddress == address) {
+                    break;
+                }
+            }
+        }
+    }
+    ALOGV("DeviceVector::getDevice() for type %08x address %s found %p",
+          type, address.string(), device.get());
+    return device;
+}
+
+sp<DeviceDescriptor> DeviceVector::getDeviceFromId(audio_port_handle_t id) const
+{
+    sp<DeviceDescriptor> device;
+    for (size_t i = 0; i < size(); i++) {
+        if (itemAt(i)->getHandle() == id) {
+            device = itemAt(i);
+            break;
+        }
+    }
+    return device;
+}
+
+DeviceVector DeviceVector::getDevicesFromType(audio_devices_t type) const
+{
+    DeviceVector devices;
+    for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) {
+        if (itemAt(i)->mDeviceType & type & ~AUDIO_DEVICE_BIT_IN) {
+            devices.add(itemAt(i));
+            type &= ~itemAt(i)->mDeviceType;
+            ALOGV("DeviceVector::getDevicesFromType() for type %x found %p",
+                  itemAt(i)->mDeviceType, itemAt(i).get());
+        }
+    }
+    return devices;
+}
+
+DeviceVector DeviceVector::getDevicesFromTypeAddr(
+        audio_devices_t type, String8 address) const
+{
+    DeviceVector devices;
+    for (size_t i = 0; i < size(); i++) {
+        if (itemAt(i)->mDeviceType == type) {
+            if (itemAt(i)->mAddress == address) {
+                devices.add(itemAt(i));
+            }
+        }
+    }
+    return devices;
+}
+
+sp<DeviceDescriptor> DeviceVector::getDeviceFromName(const String8& name) const
+{
+    sp<DeviceDescriptor> device;
+    for (size_t i = 0; i < size(); i++) {
+        if (itemAt(i)->mName == name) {
+            device = itemAt(i);
+            break;
+        }
+    }
+    return device;
+}
+
+void DeviceDescriptor::toAudioPortConfig(struct audio_port_config *dstConfig,
+                                         const struct audio_port_config *srcConfig) const
+{
+    dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK|AUDIO_PORT_CONFIG_GAIN;
+    if (srcConfig != NULL) {
+        dstConfig->config_mask |= srcConfig->config_mask;
+    }
+
+    AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+    dstConfig->id = mId;
+    dstConfig->role = audio_is_output_device(mDeviceType) ?
+                        AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
+    dstConfig->type = AUDIO_PORT_TYPE_DEVICE;
+    dstConfig->ext.device.type = mDeviceType;
+
+    //TODO Understand why this test is necessary. i.e. why at boot time does it crash
+    // without the test?
+    // This has been demonstrated to NOT be true (at start up)
+    // ALOG_ASSERT(mModule != NULL);
+    dstConfig->ext.device.hw_module = mModule != NULL ? mModule->mHandle : NULL;
+    strncpy(dstConfig->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
+}
+
+void DeviceDescriptor::toAudioPort(struct audio_port *port) const
+{
+    ALOGV("DeviceDescriptor::toAudioPort() handle %d type %x", mId, mDeviceType);
+    AudioPort::toAudioPort(port);
+    port->id = mId;
+    toAudioPortConfig(&port->active_config);
+    port->ext.device.type = mDeviceType;
+    port->ext.device.hw_module = mModule->mHandle;
+    strncpy(port->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
+}
+
+status_t DeviceDescriptor::dump(int fd, int spaces, int index) const
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "%*sDevice %d:\n", spaces, "", index+1);
+    result.append(buffer);
+    if (mId != 0) {
+        snprintf(buffer, SIZE, "%*s- id: %2d\n", spaces, "", mId);
+        result.append(buffer);
+    }
+    snprintf(buffer, SIZE, "%*s- type: %-48s\n", spaces, "",
+            ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
+                    ARRAY_SIZE(sDeviceNameToEnumTable),
+                    mDeviceType));
+    result.append(buffer);
+    if (mAddress.size() != 0) {
+        snprintf(buffer, SIZE, "%*s- address: %-32s\n", spaces, "", mAddress.string());
+        result.append(buffer);
+    }
+    write(fd, result.string(), result.size());
+    AudioPort::dump(fd, spaces);
+
+    return NO_ERROR;
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Devices.h b/services/audiopolicy/managerdefault/Devices.h
new file mode 100644
index 0000000..65e1416
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Devices.h
@@ -0,0 +1,75 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+namespace android {
+
+class AudioPort;
+class AudioPortConfig;
+
+class DeviceDescriptor: public AudioPort, public AudioPortConfig
+{
+public:
+    DeviceDescriptor(const String8& name, audio_devices_t type);
+
+    virtual ~DeviceDescriptor() {}
+
+    bool equals(const sp<DeviceDescriptor>& other) const;
+
+    // AudioPortConfig
+    virtual sp<AudioPort> getAudioPort() const { return (AudioPort*) this; }
+    virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+            const struct audio_port_config *srcConfig = NULL) const;
+
+    // AudioPort
+    virtual void loadGains(cnode *root);
+    virtual void toAudioPort(struct audio_port *port) const;
+
+    status_t dump(int fd, int spaces, int index) const;
+
+    audio_devices_t mDeviceType;
+    String8 mAddress;
+    audio_port_handle_t mId;
+
+    static String8  emptyNameStr;
+};
+
+class DeviceVector : public SortedVector< sp<DeviceDescriptor> >
+{
+public:
+    DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {}
+
+    ssize_t         add(const sp<DeviceDescriptor>& item);
+    ssize_t         remove(const sp<DeviceDescriptor>& item);
+    ssize_t         indexOf(const sp<DeviceDescriptor>& item) const;
+
+    audio_devices_t types() const { return mDeviceTypes; }
+
+    void loadDevicesFromType(audio_devices_t types);
+    void loadDevicesFromName(char *name, const DeviceVector& declaredDevices);
+
+    sp<DeviceDescriptor> getDevice(audio_devices_t type, String8 address) const;
+    DeviceVector getDevicesFromType(audio_devices_t types) const;
+    sp<DeviceDescriptor> getDeviceFromId(audio_port_handle_t id) const;
+    sp<DeviceDescriptor> getDeviceFromName(const String8& name) const;
+    DeviceVector getDevicesFromTypeAddr(audio_devices_t type, String8 address)
+    const;
+
+private:
+    void refreshTypes();
+    audio_devices_t mDeviceTypes;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Gains.cpp b/services/audiopolicy/managerdefault/Gains.cpp
new file mode 100644
index 0000000..4aca26d
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Gains.cpp
@@ -0,0 +1,446 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::Gains"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#include "AudioPolicyManager.h"
+
+#include <math.h>
+
+namespace android {
+
+const VolumeCurvePoint
+ApmGains::sDefaultVolumeCurve[ApmGains::VOLCNT] = {
+    {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
+};
+
+
+const VolumeCurvePoint
+ApmGains::sDefaultMediaVolumeCurve[ApmGains::VOLCNT] = {
+    {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sExtMediaSystemVolumeCurve[ApmGains::VOLCNT] = {
+    {1, -58.0f}, {20, -40.0f}, {60, -21.0f}, {100, -10.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerMediaVolumeCurve[ApmGains::VOLCNT] = {
+    {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerMediaVolumeCurveDrc[ApmGains::VOLCNT] = {
+    {1, -55.0f}, {20, -43.0f}, {86, -12.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerSonificationVolumeCurve[ApmGains::VOLCNT] = {
+    {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerSonificationVolumeCurveDrc[ApmGains::VOLCNT] = {
+    {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f}
+};
+
+// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks
+// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets.
+// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java).
+// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset.
+
+const VolumeCurvePoint
+ApmGains::sDefaultSystemVolumeCurve[ApmGains::VOLCNT] = {
+    {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sDefaultSystemVolumeCurveDrc[ApmGains::VOLCNT] = {
+    {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sHeadsetSystemVolumeCurve[ApmGains::VOLCNT] = {
+    {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sDefaultVoiceVolumeCurve[ApmGains::VOLCNT] = {
+    {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerVoiceVolumeCurve[ApmGains::VOLCNT] = {
+    {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sLinearVolumeCurve[ApmGains::VOLCNT] = {
+    {0, -96.0f}, {33, -68.0f}, {66, -34.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSilentVolumeCurve[ApmGains::VOLCNT] = {
+    {0, -96.0f}, {1, -96.0f}, {2, -96.0f}, {100, -96.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sFullScaleVolumeCurve[ApmGains::VOLCNT] = {
+    {0, 0.0f}, {1, 0.0f}, {2, 0.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint *ApmGains::sVolumeProfiles[AUDIO_STREAM_CNT]
+                                                  [ApmGains::DEVICE_CATEGORY_CNT] = {
+    { // AUDIO_STREAM_VOICE_CALL
+        ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_SYSTEM
+        ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sDefaultSystemVolumeCurve,  // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sExtMediaSystemVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_RING
+        ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sDefaultVolumeCurve,  // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sExtMediaSystemVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_MUSIC
+        ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_ALARM
+        ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sDefaultVolumeCurve,  // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sExtMediaSystemVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_NOTIFICATION
+        ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sDefaultVolumeCurve,  // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sExtMediaSystemVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_BLUETOOTH_SCO
+        ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_ENFORCED_AUDIBLE
+        ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sExtMediaSystemVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    {  // AUDIO_STREAM_DTMF
+        ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sExtMediaSystemVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_TTS
+      // "Transmitted Through Speaker": always silent except on DEVICE_CATEGORY_SPEAKER
+        ApmGains::sSilentVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sLinearVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sSilentVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sSilentVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_ACCESSIBILITY
+        ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_REROUTING
+        ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sFullScaleVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_PATCH
+        ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sFullScaleVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+};
+
+//static
+audio_devices_t ApmGains::getDeviceForVolume(audio_devices_t device)
+{
+    if (device == AUDIO_DEVICE_NONE) {
+        // this happens when forcing a route update and no track is active on an output.
+        // In this case the returned category is not important.
+        device =  AUDIO_DEVICE_OUT_SPEAKER;
+    } else if (popcount(device) > 1) {
+        // Multiple device selection is either:
+        //  - speaker + one other device: give priority to speaker in this case.
+        //  - one A2DP device + another device: happens with duplicated output. In this case
+        // retain the device on the A2DP output as the other must not correspond to an active
+        // selection if not the speaker.
+        //  - HDMI-CEC system audio mode only output: give priority to available item in order.
+        if (device & AUDIO_DEVICE_OUT_SPEAKER) {
+            device = AUDIO_DEVICE_OUT_SPEAKER;
+        } else if (device & AUDIO_DEVICE_OUT_HDMI_ARC) {
+            device = AUDIO_DEVICE_OUT_HDMI_ARC;
+        } else if (device & AUDIO_DEVICE_OUT_AUX_LINE) {
+            device = AUDIO_DEVICE_OUT_AUX_LINE;
+        } else if (device & AUDIO_DEVICE_OUT_SPDIF) {
+            device = AUDIO_DEVICE_OUT_SPDIF;
+        } else {
+            device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
+        }
+    }
+
+    /*SPEAKER_SAFE is an alias of SPEAKER for purposes of volume control*/
+    if (device == AUDIO_DEVICE_OUT_SPEAKER_SAFE)
+        device = AUDIO_DEVICE_OUT_SPEAKER;
+
+    ALOGW_IF(popcount(device) != 1,
+            "getDeviceForVolume() invalid device combination: %08x",
+            device);
+
+    return device;
+}
+
+//static
+ApmGains::device_category ApmGains::getDeviceCategory(audio_devices_t device)
+{
+    switch(getDeviceForVolume(device)) {
+        case AUDIO_DEVICE_OUT_EARPIECE:
+            return ApmGains::DEVICE_CATEGORY_EARPIECE;
+        case AUDIO_DEVICE_OUT_WIRED_HEADSET:
+        case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
+            return ApmGains::DEVICE_CATEGORY_HEADSET;
+        case AUDIO_DEVICE_OUT_LINE:
+        case AUDIO_DEVICE_OUT_AUX_DIGITAL:
+        /*USB?  Remote submix?*/
+            return ApmGains::DEVICE_CATEGORY_EXT_MEDIA;
+        case AUDIO_DEVICE_OUT_SPEAKER:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
+        case AUDIO_DEVICE_OUT_USB_ACCESSORY:
+        case AUDIO_DEVICE_OUT_USB_DEVICE:
+        case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
+        default:
+            return ApmGains::DEVICE_CATEGORY_SPEAKER;
+    }
+}
+
+//static
+float ApmGains::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+        int indexInUi)
+{
+    ApmGains::device_category deviceCategory = ApmGains::getDeviceCategory(device);
+    const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory];
+
+    // the volume index in the UI is relative to the min and max volume indices for this stream type
+    int nbSteps = 1 + curve[ApmGains::VOLMAX].mIndex -
+            curve[ApmGains::VOLMIN].mIndex;
+    int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) /
+            (streamDesc.mIndexMax - streamDesc.mIndexMin);
+
+    // find what part of the curve this index volume belongs to, or if it's out of bounds
+    int segment = 0;
+    if (volIdx < curve[ApmGains::VOLMIN].mIndex) {         // out of bounds
+        return 0.0f;
+    } else if (volIdx < curve[ApmGains::VOLKNEE1].mIndex) {
+        segment = 0;
+    } else if (volIdx < curve[ApmGains::VOLKNEE2].mIndex) {
+        segment = 1;
+    } else if (volIdx <= curve[ApmGains::VOLMAX].mIndex) {
+        segment = 2;
+    } else {                                                               // out of bounds
+        return 1.0f;
+    }
+
+    // linear interpolation in the attenuation table in dB
+    float decibels = curve[segment].mDBAttenuation +
+            ((float)(volIdx - curve[segment].mIndex)) *
+                ( (curve[segment+1].mDBAttenuation -
+                        curve[segment].mDBAttenuation) /
+                    ((float)(curve[segment+1].mIndex -
+                            curve[segment].mIndex)) );
+
+    float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
+
+    ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
+            curve[segment].mIndex, volIdx,
+            curve[segment+1].mIndex,
+            curve[segment].mDBAttenuation,
+            decibels,
+            curve[segment+1].mDBAttenuation,
+            amplification);
+
+    return amplification;
+}
+
+
+
+AudioGain::AudioGain(int index, bool useInChannelMask)
+{
+    mIndex = index;
+    mUseInChannelMask = useInChannelMask;
+    memset(&mGain, 0, sizeof(struct audio_gain));
+}
+
+void AudioGain::getDefaultConfig(struct audio_gain_config *config)
+{
+    config->index = mIndex;
+    config->mode = mGain.mode;
+    config->channel_mask = mGain.channel_mask;
+    if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+        config->values[0] = mGain.default_value;
+    } else {
+        uint32_t numValues;
+        if (mUseInChannelMask) {
+            numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
+        } else {
+            numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
+        }
+        for (size_t i = 0; i < numValues; i++) {
+            config->values[i] = mGain.default_value;
+        }
+    }
+    if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+        config->ramp_duration_ms = mGain.min_ramp_ms;
+    }
+}
+
+status_t AudioGain::checkConfig(const struct audio_gain_config *config)
+{
+    if ((config->mode & ~mGain.mode) != 0) {
+        return BAD_VALUE;
+    }
+    if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+        if ((config->values[0] < mGain.min_value) ||
+                    (config->values[0] > mGain.max_value)) {
+            return BAD_VALUE;
+        }
+    } else {
+        if ((config->channel_mask & ~mGain.channel_mask) != 0) {
+            return BAD_VALUE;
+        }
+        uint32_t numValues;
+        if (mUseInChannelMask) {
+            numValues = audio_channel_count_from_in_mask(config->channel_mask);
+        } else {
+            numValues = audio_channel_count_from_out_mask(config->channel_mask);
+        }
+        for (size_t i = 0; i < numValues; i++) {
+            if ((config->values[i] < mGain.min_value) ||
+                    (config->values[i] > mGain.max_value)) {
+                return BAD_VALUE;
+            }
+        }
+    }
+    if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+        if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
+                    (config->ramp_duration_ms > mGain.max_ramp_ms)) {
+            return BAD_VALUE;
+        }
+    }
+    return NO_ERROR;
+}
+
+void AudioGain::dump(int fd, int spaces, int index) const
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "%*sGain %d:\n", spaces, "", index+1);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- mode: %08x\n", spaces, "", mGain.mode);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- min_value: %d mB\n", spaces, "", mGain.min_value);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- max_value: %d mB\n", spaces, "", mGain.max_value);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- default_value: %d mB\n", spaces, "", mGain.default_value);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- step_value: %d mB\n", spaces, "", mGain.step_value);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms);
+    result.append(buffer);
+
+    write(fd, result.string(), result.size());
+}
+
+
+// --- StreamDescriptor class implementation
+
+StreamDescriptor::StreamDescriptor()
+    :   mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
+{
+    mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
+}
+
+int StreamDescriptor::getVolumeIndex(audio_devices_t device)
+{
+    device = ApmGains::getDeviceForVolume(device);
+    // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT
+    if (mIndexCur.indexOfKey(device) < 0) {
+        device = AUDIO_DEVICE_OUT_DEFAULT;
+    }
+    return mIndexCur.valueFor(device);
+}
+
+void StreamDescriptor::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "%s         %02d         %02d         ",
+             mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax);
+    result.append(buffer);
+    for (size_t i = 0; i < mIndexCur.size(); i++) {
+        snprintf(buffer, SIZE, "%04x : %02d, ",
+                 mIndexCur.keyAt(i),
+                 mIndexCur.valueAt(i));
+        result.append(buffer);
+    }
+    result.append("\n");
+
+    write(fd, result.string(), result.size());
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Gains.h b/services/audiopolicy/managerdefault/Gains.h
new file mode 100644
index 0000000..b4ab129
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Gains.h
@@ -0,0 +1,112 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+namespace android {
+
+class VolumeCurvePoint
+{
+public:
+    int mIndex;
+    float mDBAttenuation;
+};
+
+class StreamDescriptor;
+
+class ApmGains
+{
+public :
+    // 4 points to define the volume attenuation curve, each characterized by the volume
+    // index (from 0 to 100) at which they apply, and the attenuation in dB at that index.
+    // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl()
+    enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4};
+
+    // device categories used for volume curve management.
+    enum device_category {
+        DEVICE_CATEGORY_HEADSET,
+        DEVICE_CATEGORY_SPEAKER,
+        DEVICE_CATEGORY_EARPIECE,
+        DEVICE_CATEGORY_EXT_MEDIA,
+        DEVICE_CATEGORY_CNT
+    };
+
+    // returns the category the device belongs to with regard to volume curve management
+    static ApmGains::device_category getDeviceCategory(audio_devices_t device);
+
+    // extract one device relevant for volume control from multiple device selection
+    static audio_devices_t getDeviceForVolume(audio_devices_t device);
+
+    static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+                    int indexInUi);
+
+    // default volume curve
+    static const VolumeCurvePoint sDefaultVolumeCurve[ApmGains::VOLCNT];
+    // default volume curve for media strategy
+    static const VolumeCurvePoint sDefaultMediaVolumeCurve[ApmGains::VOLCNT];
+    // volume curve for non-media audio on ext media outputs (HDMI, Line, etc)
+    static const VolumeCurvePoint sExtMediaSystemVolumeCurve[ApmGains::VOLCNT];
+    // volume curve for media strategy on speakers
+    static const VolumeCurvePoint sSpeakerMediaVolumeCurve[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sSpeakerMediaVolumeCurveDrc[ApmGains::VOLCNT];
+    // volume curve for sonification strategy on speakers
+    static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sDefaultSystemVolumeCurve[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sHeadsetSystemVolumeCurve[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sDefaultVoiceVolumeCurve[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sLinearVolumeCurve[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sSilentVolumeCurve[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sFullScaleVolumeCurve[ApmGains::VOLCNT];
+    // default volume curves per stream and device category. See initializeVolumeCurves()
+    static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][ApmGains::DEVICE_CATEGORY_CNT];
+};
+
+
+class AudioGain: public RefBase
+{
+public:
+    AudioGain(int index, bool useInChannelMask);
+    virtual ~AudioGain() {}
+
+    void dump(int fd, int spaces, int index) const;
+
+    void getDefaultConfig(struct audio_gain_config *config);
+    status_t checkConfig(const struct audio_gain_config *config);
+    int               mIndex;
+    struct audio_gain mGain;
+    bool              mUseInChannelMask;
+};
+
+
+// stream descriptor used for volume control
+class StreamDescriptor
+{
+public:
+    StreamDescriptor();
+
+    int getVolumeIndex(audio_devices_t device);
+    void dump(int fd);
+
+    int mIndexMin;      // min volume index
+    int mIndexMax;      // max volume index
+    KeyedVector<audio_devices_t, int> mIndexCur;   // current volume index per device
+    bool mCanBeMuted;   // true is the stream can be muted
+
+    const VolumeCurvePoint *mVolumeCurve[ApmGains::DEVICE_CATEGORY_CNT];
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/HwModule.cpp b/services/audiopolicy/managerdefault/HwModule.cpp
new file mode 100644
index 0000000..a04bdc8
--- /dev/null
+++ b/services/audiopolicy/managerdefault/HwModule.cpp
@@ -0,0 +1,279 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::HwModule"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+#include "audio_policy_conf.h"
+#include <hardware/audio.h>
+
+namespace android {
+
+HwModule::HwModule(const char *name)
+    : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)),
+      mHalVersion(AUDIO_DEVICE_API_VERSION_MIN), mHandle(0)
+{
+}
+
+HwModule::~HwModule()
+{
+    for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+        mOutputProfiles[i]->mSupportedDevices.clear();
+    }
+    for (size_t i = 0; i < mInputProfiles.size(); i++) {
+        mInputProfiles[i]->mSupportedDevices.clear();
+    }
+    free((void *)mName);
+}
+
+status_t HwModule::loadInput(cnode *root)
+{
+    cnode *node = root->first_child;
+
+    sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SINK, this);
+
+    while (node) {
+        if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
+            profile->loadSamplingRates((char *)node->value);
+        } else if (strcmp(node->name, FORMATS_TAG) == 0) {
+            profile->loadFormats((char *)node->value);
+        } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+            profile->loadInChannels((char *)node->value);
+        } else if (strcmp(node->name, DEVICES_TAG) == 0) {
+            profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
+                                                           mDeclaredDevices);
+        } else if (strcmp(node->name, FLAGS_TAG) == 0) {
+            profile->mFlags = ConfigParsingUtils::parseInputFlagNames((char *)node->value);
+        } else if (strcmp(node->name, GAINS_TAG) == 0) {
+            profile->loadGains(node);
+        }
+        node = node->next;
+    }
+    ALOGW_IF(profile->mSupportedDevices.isEmpty(),
+            "loadInput() invalid supported devices");
+    ALOGW_IF(profile->mChannelMasks.size() == 0,
+            "loadInput() invalid supported channel masks");
+    ALOGW_IF(profile->mSamplingRates.size() == 0,
+            "loadInput() invalid supported sampling rates");
+    ALOGW_IF(profile->mFormats.size() == 0,
+            "loadInput() invalid supported formats");
+    if (!profile->mSupportedDevices.isEmpty() &&
+            (profile->mChannelMasks.size() != 0) &&
+            (profile->mSamplingRates.size() != 0) &&
+            (profile->mFormats.size() != 0)) {
+
+        ALOGV("loadInput() adding input Supported Devices %04x",
+              profile->mSupportedDevices.types());
+
+        mInputProfiles.add(profile);
+        return NO_ERROR;
+    } else {
+        return BAD_VALUE;
+    }
+}
+
+status_t HwModule::loadOutput(cnode *root)
+{
+    cnode *node = root->first_child;
+
+    sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SOURCE, this);
+
+    while (node) {
+        if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
+            profile->loadSamplingRates((char *)node->value);
+        } else if (strcmp(node->name, FORMATS_TAG) == 0) {
+            profile->loadFormats((char *)node->value);
+        } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+            profile->loadOutChannels((char *)node->value);
+        } else if (strcmp(node->name, DEVICES_TAG) == 0) {
+            profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
+                                                           mDeclaredDevices);
+        } else if (strcmp(node->name, FLAGS_TAG) == 0) {
+            profile->mFlags = ConfigParsingUtils::parseOutputFlagNames((char *)node->value);
+        } else if (strcmp(node->name, GAINS_TAG) == 0) {
+            profile->loadGains(node);
+        }
+        node = node->next;
+    }
+    ALOGW_IF(profile->mSupportedDevices.isEmpty(),
+            "loadOutput() invalid supported devices");
+    ALOGW_IF(profile->mChannelMasks.size() == 0,
+            "loadOutput() invalid supported channel masks");
+    ALOGW_IF(profile->mSamplingRates.size() == 0,
+            "loadOutput() invalid supported sampling rates");
+    ALOGW_IF(profile->mFormats.size() == 0,
+            "loadOutput() invalid supported formats");
+    if (!profile->mSupportedDevices.isEmpty() &&
+            (profile->mChannelMasks.size() != 0) &&
+            (profile->mSamplingRates.size() != 0) &&
+            (profile->mFormats.size() != 0)) {
+
+        ALOGV("loadOutput() adding output Supported Devices %04x, mFlags %04x",
+              profile->mSupportedDevices.types(), profile->mFlags);
+
+        mOutputProfiles.add(profile);
+        return NO_ERROR;
+    } else {
+        return BAD_VALUE;
+    }
+}
+
+status_t HwModule::loadDevice(cnode *root)
+{
+    cnode *node = root->first_child;
+
+    audio_devices_t type = AUDIO_DEVICE_NONE;
+    while (node) {
+        if (strcmp(node->name, DEVICE_TYPE) == 0) {
+            type = ConfigParsingUtils::parseDeviceNames((char *)node->value);
+            break;
+        }
+        node = node->next;
+    }
+    if (type == AUDIO_DEVICE_NONE ||
+            (!audio_is_input_device(type) && !audio_is_output_device(type))) {
+        ALOGW("loadDevice() bad type %08x", type);
+        return BAD_VALUE;
+    }
+    sp<DeviceDescriptor> deviceDesc = new DeviceDescriptor(String8(root->name), type);
+    deviceDesc->mModule = this;
+
+    node = root->first_child;
+    while (node) {
+        if (strcmp(node->name, DEVICE_ADDRESS) == 0) {
+            deviceDesc->mAddress = String8((char *)node->value);
+        } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+            if (audio_is_input_device(type)) {
+                deviceDesc->loadInChannels((char *)node->value);
+            } else {
+                deviceDesc->loadOutChannels((char *)node->value);
+            }
+        } else if (strcmp(node->name, GAINS_TAG) == 0) {
+            deviceDesc->loadGains(node);
+        }
+        node = node->next;
+    }
+
+    ALOGV("loadDevice() adding device name %s type %08x address %s",
+          deviceDesc->mName.string(), type, deviceDesc->mAddress.string());
+
+    mDeclaredDevices.add(deviceDesc);
+
+    return NO_ERROR;
+}
+
+status_t HwModule::addOutputProfile(String8 name, const audio_config_t *config,
+                                                  audio_devices_t device, String8 address)
+{
+    sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SOURCE, this);
+
+    profile->mSamplingRates.add(config->sample_rate);
+    profile->mChannelMasks.add(config->channel_mask);
+    profile->mFormats.add(config->format);
+
+    sp<DeviceDescriptor> devDesc = new DeviceDescriptor(name, device);
+    devDesc->mAddress = address;
+    profile->mSupportedDevices.add(devDesc);
+
+    mOutputProfiles.add(profile);
+
+    return NO_ERROR;
+}
+
+status_t HwModule::removeOutputProfile(String8 name)
+{
+    for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+        if (mOutputProfiles[i]->mName == name) {
+            mOutputProfiles.removeAt(i);
+            break;
+        }
+    }
+
+    return NO_ERROR;
+}
+
+status_t HwModule::addInputProfile(String8 name, const audio_config_t *config,
+                                                  audio_devices_t device, String8 address)
+{
+    sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SINK, this);
+
+    profile->mSamplingRates.add(config->sample_rate);
+    profile->mChannelMasks.add(config->channel_mask);
+    profile->mFormats.add(config->format);
+
+    sp<DeviceDescriptor> devDesc = new DeviceDescriptor(name, device);
+    devDesc->mAddress = address;
+    profile->mSupportedDevices.add(devDesc);
+
+    ALOGV("addInputProfile() name %s rate %d mask 0x08", name.string(), config->sample_rate, config->channel_mask);
+
+    mInputProfiles.add(profile);
+
+    return NO_ERROR;
+}
+
+status_t HwModule::removeInputProfile(String8 name)
+{
+    for (size_t i = 0; i < mInputProfiles.size(); i++) {
+        if (mInputProfiles[i]->mName == name) {
+            mInputProfiles.removeAt(i);
+            break;
+        }
+    }
+
+    return NO_ERROR;
+}
+
+
+void HwModule::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "  - name: %s\n", mName);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "  - handle: %d\n", mHandle);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "  - version: %u.%u\n", mHalVersion >> 8, mHalVersion & 0xFF);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+    if (mOutputProfiles.size()) {
+        write(fd, "  - outputs:\n", strlen("  - outputs:\n"));
+        for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+            snprintf(buffer, SIZE, "    output %zu:\n", i);
+            write(fd, buffer, strlen(buffer));
+            mOutputProfiles[i]->dump(fd);
+        }
+    }
+    if (mInputProfiles.size()) {
+        write(fd, "  - inputs:\n", strlen("  - inputs:\n"));
+        for (size_t i = 0; i < mInputProfiles.size(); i++) {
+            snprintf(buffer, SIZE, "    input %zu:\n", i);
+            write(fd, buffer, strlen(buffer));
+            mInputProfiles[i]->dump(fd);
+        }
+    }
+    if (mDeclaredDevices.size()) {
+        write(fd, "  - devices:\n", strlen("  - devices:\n"));
+        for (size_t i = 0; i < mDeclaredDevices.size(); i++) {
+            mDeclaredDevices[i]->dump(fd, 4, i);
+        }
+    }
+}
+
+} //namespace android
diff --git a/services/audiopolicy/managerdefault/HwModule.h b/services/audiopolicy/managerdefault/HwModule.h
new file mode 100644
index 0000000..f814dd9
--- /dev/null
+++ b/services/audiopolicy/managerdefault/HwModule.h
@@ -0,0 +1,46 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+namespace android {
+
+class HwModule : public RefBase
+{
+public:
+    HwModule(const char *name);
+    ~HwModule();
+
+    status_t loadOutput(cnode *root);
+    status_t loadInput(cnode *root);
+    status_t loadDevice(cnode *root);
+
+    status_t addOutputProfile(String8 name, const audio_config_t *config,
+            audio_devices_t device, String8 address);
+    status_t removeOutputProfile(String8 name);
+    status_t addInputProfile(String8 name, const audio_config_t *config,
+            audio_devices_t device, String8 address);
+    status_t removeInputProfile(String8 name);
+
+    void dump(int fd);
+
+    const char *const        mName; // base name of the audio HW module (primary, a2dp ...)
+    uint32_t                 mHalVersion; // audio HAL API version
+    audio_module_handle_t    mHandle;
+    Vector < sp<IOProfile> > mOutputProfiles; // output profiles exposed by this module
+    Vector < sp<IOProfile> > mInputProfiles;  // input profiles exposed by this module
+    DeviceVector             mDeclaredDevices; // devices declared in audio_policy.conf
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/IOProfile.cpp b/services/audiopolicy/managerdefault/IOProfile.cpp
new file mode 100644
index 0000000..538ac1a
--- /dev/null
+++ b/services/audiopolicy/managerdefault/IOProfile.cpp
@@ -0,0 +1,139 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::IOProfile"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+IOProfile::IOProfile(const String8& name, audio_port_role_t role,
+                                         const sp<HwModule>& module)
+    : AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module)
+{
+}
+
+IOProfile::~IOProfile()
+{
+}
+
+// checks if the IO profile is compatible with specified parameters.
+// Sampling rate, format and channel mask must be specified in order to
+// get a valid a match
+bool IOProfile::isCompatibleProfile(audio_devices_t device,
+                                                        String8 address,
+                                                        uint32_t samplingRate,
+                                                        uint32_t *updatedSamplingRate,
+                                                        audio_format_t format,
+                                                        audio_channel_mask_t channelMask,
+                                                        uint32_t flags) const
+{
+    const bool isPlaybackThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SOURCE;
+    const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
+    ALOG_ASSERT(isPlaybackThread != isRecordThread);
+
+    if (device != AUDIO_DEVICE_NONE && mSupportedDevices.getDevice(device, address) == 0) {
+        return false;
+    }
+
+    if (samplingRate == 0) {
+         return false;
+    }
+    uint32_t myUpdatedSamplingRate = samplingRate;
+    if (isPlaybackThread && checkExactSamplingRate(samplingRate) != NO_ERROR) {
+         return false;
+    }
+    if (isRecordThread && checkCompatibleSamplingRate(samplingRate, &myUpdatedSamplingRate) !=
+            NO_ERROR) {
+         return false;
+    }
+
+    if (!audio_is_valid_format(format) || checkFormat(format) != NO_ERROR) {
+        return false;
+    }
+
+    if (isPlaybackThread && (!audio_is_output_channel(channelMask) ||
+            checkExactChannelMask(channelMask) != NO_ERROR)) {
+        return false;
+    }
+    if (isRecordThread && (!audio_is_input_channel(channelMask) ||
+            checkCompatibleChannelMask(channelMask) != NO_ERROR)) {
+        return false;
+    }
+
+    if (isPlaybackThread && (mFlags & flags) != flags) {
+        return false;
+    }
+    // The only input flag that is allowed to be different is the fast flag.
+    // An existing fast stream is compatible with a normal track request.
+    // An existing normal stream is compatible with a fast track request,
+    // but the fast request will be denied by AudioFlinger and converted to normal track.
+    if (isRecordThread && ((mFlags ^ flags) &
+            ~AUDIO_INPUT_FLAG_FAST)) {
+        return false;
+    }
+
+    if (updatedSamplingRate != NULL) {
+        *updatedSamplingRate = myUpdatedSamplingRate;
+    }
+    return true;
+}
+
+void IOProfile::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    AudioPort::dump(fd, 4);
+
+    snprintf(buffer, SIZE, "    - flags: 0x%04x\n", mFlags);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "    - devices:\n");
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+    for (size_t i = 0; i < mSupportedDevices.size(); i++) {
+        mSupportedDevices[i]->dump(fd, 6, i);
+    }
+}
+
+void IOProfile::log()
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    ALOGV("    - sampling rates: ");
+    for (size_t i = 0; i < mSamplingRates.size(); i++) {
+        ALOGV("  %d", mSamplingRates[i]);
+    }
+
+    ALOGV("    - channel masks: ");
+    for (size_t i = 0; i < mChannelMasks.size(); i++) {
+        ALOGV("  0x%04x", mChannelMasks[i]);
+    }
+
+    ALOGV("    - formats: ");
+    for (size_t i = 0; i < mFormats.size(); i++) {
+        ALOGV("  0x%08x", mFormats[i]);
+    }
+
+    ALOGV("    - devices: 0x%04x\n", mSupportedDevices.types());
+    ALOGV("    - flags: 0x%04x\n", mFlags);
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/IOProfile.h b/services/audiopolicy/managerdefault/IOProfile.h
new file mode 100644
index 0000000..3317969
--- /dev/null
+++ b/services/audiopolicy/managerdefault/IOProfile.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+namespace android {
+
+class HwModule;
+
+// the IOProfile class describes the capabilities of an output or input stream.
+// It is currently assumed that all combination of listed parameters are supported.
+// It is used by the policy manager to determine if an output or input is suitable for
+// a given use case,  open/close it accordingly and connect/disconnect audio tracks
+// to/from it.
+class IOProfile : public AudioPort
+{
+public:
+    IOProfile(const String8& name, audio_port_role_t role, const sp<HwModule>& module);
+    virtual ~IOProfile();
+
+    // This method is used for both output and input.
+    // If parameter updatedSamplingRate is non-NULL, it is assigned the actual sample rate.
+    // For input, flags is interpreted as audio_input_flags_t.
+    // TODO: merge audio_output_flags_t and audio_input_flags_t.
+    bool isCompatibleProfile(audio_devices_t device,
+                             String8 address,
+                             uint32_t samplingRate,
+                             uint32_t *updatedSamplingRate,
+                             audio_format_t format,
+                             audio_channel_mask_t channelMask,
+                             uint32_t flags) const;
+
+    void dump(int fd);
+    void log();
+
+    DeviceVector  mSupportedDevices; // supported devices
+                                     // (devices this output can be routed to)
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Ports.cpp b/services/audiopolicy/managerdefault/Ports.cpp
new file mode 100644
index 0000000..3e55cee
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Ports.cpp
@@ -0,0 +1,844 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::Ports"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+#include "audio_policy_conf.h"
+
+namespace android {
+
+// --- AudioPort class implementation
+
+AudioPort::AudioPort(const String8& name, audio_port_type_t type,
+          audio_port_role_t role, const sp<HwModule>& module) :
+    mName(name), mType(type), mRole(role), mModule(module), mFlags(0), mId(0)
+{
+    mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) ||
+                    ((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK));
+}
+
+void AudioPort::attach(const sp<HwModule>& module) {
+    mId = AudioPolicyManager::nextUniqueId();
+    mModule = module;
+}
+
+void AudioPort::toAudioPort(struct audio_port *port) const
+{
+    port->role = mRole;
+    port->type = mType;
+    strlcpy(port->name, mName, AUDIO_PORT_MAX_NAME_LEN);
+    unsigned int i;
+    for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) {
+        if (mSamplingRates[i] != 0) {
+            port->sample_rates[i] = mSamplingRates[i];
+        }
+    }
+    port->num_sample_rates = i;
+    for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) {
+        if (mChannelMasks[i] != 0) {
+            port->channel_masks[i] = mChannelMasks[i];
+        }
+    }
+    port->num_channel_masks = i;
+    for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) {
+        if (mFormats[i] != 0) {
+            port->formats[i] = mFormats[i];
+        }
+    }
+    port->num_formats = i;
+
+    ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
+
+    for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
+        port->gains[i] = mGains[i]->mGain;
+    }
+    port->num_gains = i;
+}
+
+void AudioPort::importAudioPort(const sp<AudioPort> port) {
+    for (size_t k = 0 ; k < port->mSamplingRates.size() ; k++) {
+        const uint32_t rate = port->mSamplingRates.itemAt(k);
+        if (rate != 0) { // skip "dynamic" rates
+            bool hasRate = false;
+            for (size_t l = 0 ; l < mSamplingRates.size() ; l++) {
+                if (rate == mSamplingRates.itemAt(l)) {
+                    hasRate = true;
+                    break;
+                }
+            }
+            if (!hasRate) { // never import a sampling rate twice
+                mSamplingRates.add(rate);
+            }
+        }
+    }
+    for (size_t k = 0 ; k < port->mChannelMasks.size() ; k++) {
+        const audio_channel_mask_t mask = port->mChannelMasks.itemAt(k);
+        if (mask != 0) { // skip "dynamic" masks
+            bool hasMask = false;
+            for (size_t l = 0 ; l < mChannelMasks.size() ; l++) {
+                if (mask == mChannelMasks.itemAt(l)) {
+                    hasMask = true;
+                    break;
+                }
+            }
+            if (!hasMask) { // never import a channel mask twice
+                mChannelMasks.add(mask);
+            }
+        }
+    }
+    for (size_t k = 0 ; k < port->mFormats.size() ; k++) {
+        const audio_format_t format = port->mFormats.itemAt(k);
+        if (format != 0) { // skip "dynamic" formats
+            bool hasFormat = false;
+            for (size_t l = 0 ; l < mFormats.size() ; l++) {
+                if (format == mFormats.itemAt(l)) {
+                    hasFormat = true;
+                    break;
+                }
+            }
+            if (!hasFormat) { // never import a channel mask twice
+                mFormats.add(format);
+            }
+        }
+    }
+    for (size_t k = 0 ; k < port->mGains.size() ; k++) {
+        sp<AudioGain> gain = port->mGains.itemAt(k);
+        if (gain != 0) {
+            bool hasGain = false;
+            for (size_t l = 0 ; l < mGains.size() ; l++) {
+                if (gain == mGains.itemAt(l)) {
+                    hasGain = true;
+                    break;
+                }
+            }
+            if (!hasGain) { // never import a gain twice
+                mGains.add(gain);
+            }
+        }
+    }
+}
+
+void AudioPort::clearCapabilities() {
+    mChannelMasks.clear();
+    mFormats.clear();
+    mSamplingRates.clear();
+    mGains.clear();
+}
+
+void AudioPort::loadSamplingRates(char *name)
+{
+    char *str = strtok(name, "|");
+
+    // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
+    // rates should be read from the output stream after it is opened for the first time
+    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+        mSamplingRates.add(0);
+        return;
+    }
+
+    while (str != NULL) {
+        uint32_t rate = atoi(str);
+        if (rate != 0) {
+            ALOGV("loadSamplingRates() adding rate %d", rate);
+            mSamplingRates.add(rate);
+        }
+        str = strtok(NULL, "|");
+    }
+}
+
+void AudioPort::loadFormats(char *name)
+{
+    char *str = strtok(name, "|");
+
+    // by convention, "0' in the first entry in mFormats indicates the supported formats
+    // should be read from the output stream after it is opened for the first time
+    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+        mFormats.add(AUDIO_FORMAT_DEFAULT);
+        return;
+    }
+
+    while (str != NULL) {
+        audio_format_t format = (audio_format_t)ConfigParsingUtils::stringToEnum(sFormatNameToEnumTable,
+                                                             ARRAY_SIZE(sFormatNameToEnumTable),
+                                                             str);
+        if (format != AUDIO_FORMAT_DEFAULT) {
+            mFormats.add(format);
+        }
+        str = strtok(NULL, "|");
+    }
+}
+
+void AudioPort::loadInChannels(char *name)
+{
+    const char *str = strtok(name, "|");
+
+    ALOGV("loadInChannels() %s", name);
+
+    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+        mChannelMasks.add(0);
+        return;
+    }
+
+    while (str != NULL) {
+        audio_channel_mask_t channelMask =
+                (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable,
+                                                   ARRAY_SIZE(sInChannelsNameToEnumTable),
+                                                   str);
+        if (channelMask != 0) {
+            ALOGV("loadInChannels() adding channelMask %04x", channelMask);
+            mChannelMasks.add(channelMask);
+        }
+        str = strtok(NULL, "|");
+    }
+}
+
+void AudioPort::loadOutChannels(char *name)
+{
+    const char *str = strtok(name, "|");
+
+    ALOGV("loadOutChannels() %s", name);
+
+    // by convention, "0' in the first entry in mChannelMasks indicates the supported channel
+    // masks should be read from the output stream after it is opened for the first time
+    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+        mChannelMasks.add(0);
+        return;
+    }
+
+    while (str != NULL) {
+        audio_channel_mask_t channelMask =
+                (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable,
+                                                   ARRAY_SIZE(sOutChannelsNameToEnumTable),
+                                                   str);
+        if (channelMask != 0) {
+            mChannelMasks.add(channelMask);
+        }
+        str = strtok(NULL, "|");
+    }
+    return;
+}
+
+audio_gain_mode_t AudioPort::loadGainMode(char *name)
+{
+    const char *str = strtok(name, "|");
+
+    ALOGV("loadGainMode() %s", name);
+    audio_gain_mode_t mode = 0;
+    while (str != NULL) {
+        mode |= (audio_gain_mode_t)ConfigParsingUtils::stringToEnum(sGainModeNameToEnumTable,
+                                                ARRAY_SIZE(sGainModeNameToEnumTable),
+                                                str);
+        str = strtok(NULL, "|");
+    }
+    return mode;
+}
+
+void AudioPort::loadGain(cnode *root, int index)
+{
+    cnode *node = root->first_child;
+
+    sp<AudioGain> gain = new AudioGain(index, mUseInChannelMask);
+
+    while (node) {
+        if (strcmp(node->name, GAIN_MODE) == 0) {
+            gain->mGain.mode = loadGainMode((char *)node->value);
+        } else if (strcmp(node->name, GAIN_CHANNELS) == 0) {
+            if (mUseInChannelMask) {
+                gain->mGain.channel_mask =
+                        (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable,
+                                                           ARRAY_SIZE(sInChannelsNameToEnumTable),
+                                                           (char *)node->value);
+            } else {
+                gain->mGain.channel_mask =
+                        (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable,
+                                                           ARRAY_SIZE(sOutChannelsNameToEnumTable),
+                                                           (char *)node->value);
+            }
+        } else if (strcmp(node->name, GAIN_MIN_VALUE) == 0) {
+            gain->mGain.min_value = atoi((char *)node->value);
+        } else if (strcmp(node->name, GAIN_MAX_VALUE) == 0) {
+            gain->mGain.max_value = atoi((char *)node->value);
+        } else if (strcmp(node->name, GAIN_DEFAULT_VALUE) == 0) {
+            gain->mGain.default_value = atoi((char *)node->value);
+        } else if (strcmp(node->name, GAIN_STEP_VALUE) == 0) {
+            gain->mGain.step_value = atoi((char *)node->value);
+        } else if (strcmp(node->name, GAIN_MIN_RAMP_MS) == 0) {
+            gain->mGain.min_ramp_ms = atoi((char *)node->value);
+        } else if (strcmp(node->name, GAIN_MAX_RAMP_MS) == 0) {
+            gain->mGain.max_ramp_ms = atoi((char *)node->value);
+        }
+        node = node->next;
+    }
+
+    ALOGV("loadGain() adding new gain mode %08x channel mask %08x min mB %d max mB %d",
+          gain->mGain.mode, gain->mGain.channel_mask, gain->mGain.min_value, gain->mGain.max_value);
+
+    if (gain->mGain.mode == 0) {
+        return;
+    }
+    mGains.add(gain);
+}
+
+void AudioPort::loadGains(cnode *root)
+{
+    cnode *node = root->first_child;
+    int index = 0;
+    while (node) {
+        ALOGV("loadGains() loading gain %s", node->name);
+        loadGain(node, index++);
+        node = node->next;
+    }
+}
+
+status_t AudioPort::checkExactSamplingRate(uint32_t samplingRate) const
+{
+    if (mSamplingRates.isEmpty()) {
+        return NO_ERROR;
+    }
+
+    for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+        if (mSamplingRates[i] == samplingRate) {
+            return NO_ERROR;
+        }
+    }
+    return BAD_VALUE;
+}
+
+status_t AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate,
+        uint32_t *updatedSamplingRate) const
+{
+    if (mSamplingRates.isEmpty()) {
+        return NO_ERROR;
+    }
+
+    // Search for the closest supported sampling rate that is above (preferred)
+    // or below (acceptable) the desired sampling rate, within a permitted ratio.
+    // The sampling rates do not need to be sorted in ascending order.
+    ssize_t maxBelow = -1;
+    ssize_t minAbove = -1;
+    uint32_t candidate;
+    for (size_t i = 0; i < mSamplingRates.size(); i++) {
+        candidate = mSamplingRates[i];
+        if (candidate == samplingRate) {
+            if (updatedSamplingRate != NULL) {
+                *updatedSamplingRate = candidate;
+            }
+            return NO_ERROR;
+        }
+        // candidate < desired
+        if (candidate < samplingRate) {
+            if (maxBelow < 0 || candidate > mSamplingRates[maxBelow]) {
+                maxBelow = i;
+            }
+        // candidate > desired
+        } else {
+            if (minAbove < 0 || candidate < mSamplingRates[minAbove]) {
+                minAbove = i;
+            }
+        }
+    }
+    // This uses hard-coded knowledge about AudioFlinger resampling ratios.
+    // TODO Move these assumptions out.
+    static const uint32_t kMaxDownSampleRatio = 6;  // beyond this aliasing occurs
+    static const uint32_t kMaxUpSampleRatio = 256;  // beyond this sample rate inaccuracies occur
+                                                    // due to approximation by an int32_t of the
+                                                    // phase increments
+    // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
+    if (minAbove >= 0) {
+        candidate = mSamplingRates[minAbove];
+        if (candidate / kMaxDownSampleRatio <= samplingRate) {
+            if (updatedSamplingRate != NULL) {
+                *updatedSamplingRate = candidate;
+            }
+            return NO_ERROR;
+        }
+    }
+    // But if we have to up-sample from a lower sampling rate, that's OK.
+    if (maxBelow >= 0) {
+        candidate = mSamplingRates[maxBelow];
+        if (candidate * kMaxUpSampleRatio >= samplingRate) {
+            if (updatedSamplingRate != NULL) {
+                *updatedSamplingRate = candidate;
+            }
+            return NO_ERROR;
+        }
+    }
+    // leave updatedSamplingRate unmodified
+    return BAD_VALUE;
+}
+
+status_t AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) const
+{
+    if (mChannelMasks.isEmpty()) {
+        return NO_ERROR;
+    }
+
+    for (size_t i = 0; i < mChannelMasks.size(); i++) {
+        if (mChannelMasks[i] == channelMask) {
+            return NO_ERROR;
+        }
+    }
+    return BAD_VALUE;
+}
+
+status_t AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask)
+        const
+{
+    if (mChannelMasks.isEmpty()) {
+        return NO_ERROR;
+    }
+
+    const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
+    for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+        // FIXME Does not handle multi-channel automatic conversions yet
+        audio_channel_mask_t supported = mChannelMasks[i];
+        if (supported == channelMask) {
+            return NO_ERROR;
+        }
+        if (isRecordThread) {
+            // This uses hard-coded knowledge that AudioFlinger can silently down-mix and up-mix.
+            // FIXME Abstract this out to a table.
+            if (((supported == AUDIO_CHANNEL_IN_FRONT_BACK || supported == AUDIO_CHANNEL_IN_STEREO)
+                    && channelMask == AUDIO_CHANNEL_IN_MONO) ||
+                (supported == AUDIO_CHANNEL_IN_MONO && (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK
+                    || channelMask == AUDIO_CHANNEL_IN_STEREO))) {
+                return NO_ERROR;
+            }
+        }
+    }
+    return BAD_VALUE;
+}
+
+status_t AudioPort::checkFormat(audio_format_t format) const
+{
+    if (mFormats.isEmpty()) {
+        return NO_ERROR;
+    }
+
+    for (size_t i = 0; i < mFormats.size(); i ++) {
+        if (mFormats[i] == format) {
+            return NO_ERROR;
+        }
+    }
+    return BAD_VALUE;
+}
+
+
+uint32_t AudioPort::pickSamplingRate() const
+{
+    // special case for uninitialized dynamic profile
+    if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) {
+        return 0;
+    }
+
+    // For direct outputs, pick minimum sampling rate: this helps ensuring that the
+    // channel count / sampling rate combination chosen will be supported by the connected
+    // sink
+    if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
+            (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
+        uint32_t samplingRate = UINT_MAX;
+        for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+            if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) {
+                samplingRate = mSamplingRates[i];
+            }
+        }
+        return (samplingRate == UINT_MAX) ? 0 : samplingRate;
+    }
+
+    uint32_t samplingRate = 0;
+    uint32_t maxRate = MAX_MIXER_SAMPLING_RATE;
+
+    // For mixed output and inputs, use max mixer sampling rates. Do not
+    // limit sampling rate otherwise
+    if (mType != AUDIO_PORT_TYPE_MIX) {
+        maxRate = UINT_MAX;
+    }
+    for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+        if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) {
+            samplingRate = mSamplingRates[i];
+        }
+    }
+    return samplingRate;
+}
+
+audio_channel_mask_t AudioPort::pickChannelMask() const
+{
+    // special case for uninitialized dynamic profile
+    if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) {
+        return AUDIO_CHANNEL_NONE;
+    }
+    audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE;
+
+    // For direct outputs, pick minimum channel count: this helps ensuring that the
+    // channel count / sampling rate combination chosen will be supported by the connected
+    // sink
+    if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
+            (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
+        uint32_t channelCount = UINT_MAX;
+        for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+            uint32_t cnlCount;
+            if (mUseInChannelMask) {
+                cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
+            } else {
+                cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
+            }
+            if ((cnlCount < channelCount) && (cnlCount > 0)) {
+                channelMask = mChannelMasks[i];
+                channelCount = cnlCount;
+            }
+        }
+        return channelMask;
+    }
+
+    uint32_t channelCount = 0;
+    uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
+
+    // For mixed output and inputs, use max mixer channel count. Do not
+    // limit channel count otherwise
+    if (mType != AUDIO_PORT_TYPE_MIX) {
+        maxCount = UINT_MAX;
+    }
+    for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+        uint32_t cnlCount;
+        if (mUseInChannelMask) {
+            cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
+        } else {
+            cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
+        }
+        if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
+            channelMask = mChannelMasks[i];
+            channelCount = cnlCount;
+        }
+    }
+    return channelMask;
+}
+
+/* format in order of increasing preference */
+const audio_format_t AudioPort::sPcmFormatCompareTable[] = {
+        AUDIO_FORMAT_DEFAULT,
+        AUDIO_FORMAT_PCM_16_BIT,
+        AUDIO_FORMAT_PCM_8_24_BIT,
+        AUDIO_FORMAT_PCM_24_BIT_PACKED,
+        AUDIO_FORMAT_PCM_32_BIT,
+        AUDIO_FORMAT_PCM_FLOAT,
+};
+
+int AudioPort::compareFormats(audio_format_t format1,
+                                                  audio_format_t format2)
+{
+    // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any
+    // compressed format and better than any PCM format. This is by design of pickFormat()
+    if (!audio_is_linear_pcm(format1)) {
+        if (!audio_is_linear_pcm(format2)) {
+            return 0;
+        }
+        return 1;
+    }
+    if (!audio_is_linear_pcm(format2)) {
+        return -1;
+    }
+
+    int index1 = -1, index2 = -1;
+    for (size_t i = 0;
+            (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1));
+            i ++) {
+        if (sPcmFormatCompareTable[i] == format1) {
+            index1 = i;
+        }
+        if (sPcmFormatCompareTable[i] == format2) {
+            index2 = i;
+        }
+    }
+    // format1 not found => index1 < 0 => format2 > format1
+    // format2 not found => index2 < 0 => format2 < format1
+    return index1 - index2;
+}
+
+audio_format_t AudioPort::pickFormat() const
+{
+    // special case for uninitialized dynamic profile
+    if (mFormats.size() == 1 && mFormats[0] == 0) {
+        return AUDIO_FORMAT_DEFAULT;
+    }
+
+    audio_format_t format = AUDIO_FORMAT_DEFAULT;
+    audio_format_t bestFormat =
+            AudioPort::sPcmFormatCompareTable[
+                ARRAY_SIZE(AudioPort::sPcmFormatCompareTable) - 1];
+    // For mixed output and inputs, use best mixer output format. Do not
+    // limit format otherwise
+    if ((mType != AUDIO_PORT_TYPE_MIX) ||
+            ((mRole == AUDIO_PORT_ROLE_SOURCE) &&
+             (((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) != 0)))) {
+        bestFormat = AUDIO_FORMAT_INVALID;
+    }
+
+    for (size_t i = 0; i < mFormats.size(); i ++) {
+        if ((compareFormats(mFormats[i], format) > 0) &&
+                (compareFormats(mFormats[i], bestFormat) <= 0)) {
+            format = mFormats[i];
+        }
+    }
+    return format;
+}
+
+status_t AudioPort::checkGain(const struct audio_gain_config *gainConfig,
+                                                  int index) const
+{
+    if (index < 0 || (size_t)index >= mGains.size()) {
+        return BAD_VALUE;
+    }
+    return mGains[index]->checkConfig(gainConfig);
+}
+
+void AudioPort::dump(int fd, int spaces) const
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    if (mName.size() != 0) {
+        snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string());
+        result.append(buffer);
+    }
+
+    if (mSamplingRates.size() != 0) {
+        snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, "");
+        result.append(buffer);
+        for (size_t i = 0; i < mSamplingRates.size(); i++) {
+            if (i == 0 && mSamplingRates[i] == 0) {
+                snprintf(buffer, SIZE, "Dynamic");
+            } else {
+                snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
+            }
+            result.append(buffer);
+            result.append(i == (mSamplingRates.size() - 1) ? "" : ", ");
+        }
+        result.append("\n");
+    }
+
+    if (mChannelMasks.size() != 0) {
+        snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, "");
+        result.append(buffer);
+        for (size_t i = 0; i < mChannelMasks.size(); i++) {
+            ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]);
+
+            if (i == 0 && mChannelMasks[i] == 0) {
+                snprintf(buffer, SIZE, "Dynamic");
+            } else {
+                snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
+            }
+            result.append(buffer);
+            result.append(i == (mChannelMasks.size() - 1) ? "" : ", ");
+        }
+        result.append("\n");
+    }
+
+    if (mFormats.size() != 0) {
+        snprintf(buffer, SIZE, "%*s- formats: ", spaces, "");
+        result.append(buffer);
+        for (size_t i = 0; i < mFormats.size(); i++) {
+            const char *formatStr = ConfigParsingUtils::enumToString(sFormatNameToEnumTable,
+                                                 ARRAY_SIZE(sFormatNameToEnumTable),
+                                                 mFormats[i]);
+            if (i == 0 && strcmp(formatStr, "") == 0) {
+                snprintf(buffer, SIZE, "Dynamic");
+            } else {
+                snprintf(buffer, SIZE, "%s", formatStr);
+            }
+            result.append(buffer);
+            result.append(i == (mFormats.size() - 1) ? "" : ", ");
+        }
+        result.append("\n");
+    }
+    write(fd, result.string(), result.size());
+    if (mGains.size() != 0) {
+        snprintf(buffer, SIZE, "%*s- gains:\n", spaces, "");
+        write(fd, buffer, strlen(buffer) + 1);
+        result.append(buffer);
+        for (size_t i = 0; i < mGains.size(); i++) {
+            mGains[i]->dump(fd, spaces + 2, i);
+        }
+    }
+}
+
+
+// --- AudioPortConfig class implementation
+
+AudioPortConfig::AudioPortConfig()
+{
+    mSamplingRate = 0;
+    mChannelMask = AUDIO_CHANNEL_NONE;
+    mFormat = AUDIO_FORMAT_INVALID;
+    mGain.index = -1;
+}
+
+status_t AudioPortConfig::applyAudioPortConfig(
+                                                        const struct audio_port_config *config,
+                                                        struct audio_port_config *backupConfig)
+{
+    struct audio_port_config localBackupConfig;
+    status_t status = NO_ERROR;
+
+    localBackupConfig.config_mask = config->config_mask;
+    toAudioPortConfig(&localBackupConfig);
+
+    sp<AudioPort> audioport = getAudioPort();
+    if (audioport == 0) {
+        status = NO_INIT;
+        goto exit;
+    }
+    if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+        status = audioport->checkExactSamplingRate(config->sample_rate);
+        if (status != NO_ERROR) {
+            goto exit;
+        }
+        mSamplingRate = config->sample_rate;
+    }
+    if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+        status = audioport->checkExactChannelMask(config->channel_mask);
+        if (status != NO_ERROR) {
+            goto exit;
+        }
+        mChannelMask = config->channel_mask;
+    }
+    if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+        status = audioport->checkFormat(config->format);
+        if (status != NO_ERROR) {
+            goto exit;
+        }
+        mFormat = config->format;
+    }
+    if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) {
+        status = audioport->checkGain(&config->gain, config->gain.index);
+        if (status != NO_ERROR) {
+            goto exit;
+        }
+        mGain = config->gain;
+    }
+
+exit:
+    if (status != NO_ERROR) {
+        applyAudioPortConfig(&localBackupConfig);
+    }
+    if (backupConfig != NULL) {
+        *backupConfig = localBackupConfig;
+    }
+    return status;
+}
+
+void AudioPortConfig::toAudioPortConfig(struct audio_port_config *dstConfig,
+                                        const struct audio_port_config *srcConfig) const
+{
+    if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+        dstConfig->sample_rate = mSamplingRate;
+        if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) {
+            dstConfig->sample_rate = srcConfig->sample_rate;
+        }
+    } else {
+        dstConfig->sample_rate = 0;
+    }
+    if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+        dstConfig->channel_mask = mChannelMask;
+        if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) {
+            dstConfig->channel_mask = srcConfig->channel_mask;
+        }
+    } else {
+        dstConfig->channel_mask = AUDIO_CHANNEL_NONE;
+    }
+    if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+        dstConfig->format = mFormat;
+        if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) {
+            dstConfig->format = srcConfig->format;
+        }
+    } else {
+        dstConfig->format = AUDIO_FORMAT_INVALID;
+    }
+    if (dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
+        dstConfig->gain = mGain;
+        if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)) {
+            dstConfig->gain = srcConfig->gain;
+        }
+    } else {
+        dstConfig->gain.index = -1;
+    }
+    if (dstConfig->gain.index != -1) {
+        dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
+    } else {
+        dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
+    }
+}
+
+
+// --- AudioPatch class implementation
+
+AudioPatch::AudioPatch(audio_patch_handle_t handle,
+            const struct audio_patch *patch, uid_t uid) :
+                mHandle(handle), mPatch(*patch), mUid(uid), mAfPatchHandle(0)
+{}
+
+status_t AudioPatch::dump(int fd, int spaces, int index) const
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "%*sAudio patch %d:\n", spaces, "", index+1);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- handle: %2d\n", spaces, "", mHandle);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- audio flinger handle: %2d\n", spaces, "", mAfPatchHandle);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- owner uid: %2d\n", spaces, "", mUid);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- %d sources:\n", spaces, "", mPatch.num_sources);
+    result.append(buffer);
+    for (size_t i = 0; i < mPatch.num_sources; i++) {
+        if (mPatch.sources[i].type == AUDIO_PORT_TYPE_DEVICE) {
+            snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
+                     mPatch.sources[i].id, ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
+                                                        ARRAY_SIZE(sDeviceNameToEnumTable),
+                                                        mPatch.sources[i].ext.device.type));
+        } else {
+            snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
+                     mPatch.sources[i].id, mPatch.sources[i].ext.mix.handle);
+        }
+        result.append(buffer);
+    }
+    snprintf(buffer, SIZE, "%*s- %d sinks:\n", spaces, "", mPatch.num_sinks);
+    result.append(buffer);
+    for (size_t i = 0; i < mPatch.num_sinks; i++) {
+        if (mPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE) {
+            snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
+                     mPatch.sinks[i].id, ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
+                                                        ARRAY_SIZE(sDeviceNameToEnumTable),
+                                                        mPatch.sinks[i].ext.device.type));
+        } else {
+            snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
+                     mPatch.sinks[i].id, mPatch.sinks[i].ext.mix.handle);
+        }
+        result.append(buffer);
+    }
+
+    write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Ports.h b/services/audiopolicy/managerdefault/Ports.h
new file mode 100644
index 0000000..f6e0e93
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Ports.h
@@ -0,0 +1,122 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+namespace android {
+
+class HwModule;
+
+class AudioPort: public virtual RefBase
+{
+public:
+    AudioPort(const String8& name, audio_port_type_t type,
+            audio_port_role_t role, const sp<HwModule>& module);
+    virtual ~AudioPort() {}
+
+    audio_port_handle_t getHandle() { return mId; }
+
+    void attach(const sp<HwModule>& module);
+    bool isAttached() { return mId != 0; }
+
+    virtual void toAudioPort(struct audio_port *port) const;
+
+    void importAudioPort(const sp<AudioPort> port);
+    void clearCapabilities();
+
+    void loadSamplingRates(char *name);
+    void loadFormats(char *name);
+    void loadOutChannels(char *name);
+    void loadInChannels(char *name);
+
+    audio_gain_mode_t loadGainMode(char *name);
+    void loadGain(cnode *root, int index);
+    virtual void loadGains(cnode *root);
+
+    // searches for an exact match
+    status_t checkExactSamplingRate(uint32_t samplingRate) const;
+    // searches for a compatible match, and returns the best match via updatedSamplingRate
+    status_t checkCompatibleSamplingRate(uint32_t samplingRate,
+            uint32_t *updatedSamplingRate) const;
+    // searches for an exact match
+    status_t checkExactChannelMask(audio_channel_mask_t channelMask) const;
+    // searches for a compatible match, currently implemented for input channel masks only
+    status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask) const;
+    status_t checkFormat(audio_format_t format) const;
+    status_t checkGain(const struct audio_gain_config *gainConfig, int index) const;
+
+    uint32_t pickSamplingRate() const;
+    audio_channel_mask_t pickChannelMask() const;
+    audio_format_t pickFormat() const;
+
+    static const audio_format_t sPcmFormatCompareTable[];
+    static int compareFormats(audio_format_t format1, audio_format_t format2);
+
+    void dump(int fd, int spaces) const;
+
+    String8           mName;
+    audio_port_type_t mType;
+    audio_port_role_t mRole;
+    bool              mUseInChannelMask;
+    // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats
+    // indicates the supported parameters should be read from the output stream
+    // after it is opened for the first time
+    Vector <uint32_t> mSamplingRates; // supported sampling rates
+    Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks
+    Vector <audio_format_t> mFormats; // supported audio formats
+    Vector < sp<AudioGain> > mGains; // gain controllers
+    sp<HwModule> mModule;                 // audio HW module exposing this I/O stream
+    uint32_t mFlags; // attribute flags (e.g primary output,
+                     // direct output...).
+
+
+protected:
+    //TODO - clarify the role of mId in this case, both an "attached" indicator
+    // and a unique ID for identifying a port to the (upcoming) selection API,
+    // and its relationship to the mId in AudioOutputDescriptor and AudioInputDescriptor.
+    audio_port_handle_t mId;
+};
+
+class AudioPortConfig: public virtual RefBase
+{
+public:
+    AudioPortConfig();
+    virtual ~AudioPortConfig() {}
+
+    status_t applyAudioPortConfig(const struct audio_port_config *config,
+            struct audio_port_config *backupConfig = NULL);
+    virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+            const struct audio_port_config *srcConfig = NULL) const = 0;
+    virtual sp<AudioPort> getAudioPort() const = 0;
+    uint32_t mSamplingRate;
+    audio_format_t mFormat;
+    audio_channel_mask_t mChannelMask;
+    struct audio_gain_config mGain;
+};
+
+
+class AudioPatch: public RefBase
+{
+public:
+    AudioPatch(audio_patch_handle_t handle, const struct audio_patch *patch, uid_t uid);
+
+    status_t dump(int fd, int spaces, int index) const;
+
+    audio_patch_handle_t mHandle;
+    struct audio_patch mPatch;
+    uid_t mUid;
+    audio_patch_handle_t mAfPatchHandle;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/audio_policy_conf.h b/services/audiopolicy/managerdefault/audio_policy_conf.h
similarity index 100%
rename from services/audiopolicy/audio_policy_conf.h
rename to services/audiopolicy/managerdefault/audio_policy_conf.h
diff --git a/services/audiopolicy/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
similarity index 100%
rename from services/audiopolicy/AudioPolicyClientImpl.cpp
rename to services/audiopolicy/service/AudioPolicyClientImpl.cpp
diff --git a/services/audiopolicy/AudioPolicyClientImplLegacy.cpp b/services/audiopolicy/service/AudioPolicyClientImplLegacy.cpp
similarity index 100%
rename from services/audiopolicy/AudioPolicyClientImplLegacy.cpp
rename to services/audiopolicy/service/AudioPolicyClientImplLegacy.cpp
diff --git a/services/audiopolicy/AudioPolicyEffects.cpp b/services/audiopolicy/service/AudioPolicyEffects.cpp
similarity index 100%
rename from services/audiopolicy/AudioPolicyEffects.cpp
rename to services/audiopolicy/service/AudioPolicyEffects.cpp
diff --git a/services/audiopolicy/AudioPolicyEffects.h b/services/audiopolicy/service/AudioPolicyEffects.h
similarity index 100%
rename from services/audiopolicy/AudioPolicyEffects.h
rename to services/audiopolicy/service/AudioPolicyEffects.h
diff --git a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
similarity index 98%
rename from services/audiopolicy/AudioPolicyInterfaceImpl.cpp
rename to services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index a45dbb3..e9ff8389 100644
--- a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -28,7 +28,8 @@
 
 status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device,
                                                   audio_policy_dev_state_t state,
-                                                  const char *device_address)
+                                                  const char *device_address,
+                                                  const char *device_name)
 {
     if (mAudioPolicyManager == NULL) {
         return NO_INIT;
@@ -46,8 +47,8 @@
 
     ALOGV("setDeviceConnectionState()");
     Mutex::Autolock _l(mLock);
-    return mAudioPolicyManager->setDeviceConnectionState(device,
-                                                      state, device_address);
+    return mAudioPolicyManager->setDeviceConnectionState(device, state,
+                                                         device_address, device_name);
 }
 
 audio_policy_dev_state_t AudioPolicyService::getDeviceConnectionState(
diff --git a/services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp
similarity index 99%
rename from services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp
rename to services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp
index b8846c6..5a91192 100644
--- a/services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp
@@ -33,7 +33,8 @@
 
 status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device,
                                                   audio_policy_dev_state_t state,
-                                                  const char *device_address)
+                                                  const char *device_address,
+                                                  const char *device_name __unused)
 {
     if (mpAudioPolicy == NULL) {
         return NO_INIT;
diff --git a/services/audiopolicy/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
similarity index 100%
rename from services/audiopolicy/AudioPolicyService.cpp
rename to services/audiopolicy/service/AudioPolicyService.cpp
diff --git a/services/audiopolicy/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
similarity index 99%
rename from services/audiopolicy/AudioPolicyService.h
rename to services/audiopolicy/service/AudioPolicyService.h
index 80284a4..0378384 100644
--- a/services/audiopolicy/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -35,7 +35,7 @@
 #include <hardware_legacy/AudioPolicyInterface.h>
 #endif
 #include "AudioPolicyEffects.h"
-#include "AudioPolicyManager.h"
+#include "managerdefault/AudioPolicyManager.h"
 
 
 namespace android {
@@ -61,7 +61,8 @@
 
     virtual status_t setDeviceConnectionState(audio_devices_t device,
                                               audio_policy_dev_state_t state,
-                                              const char *device_address);
+                                              const char *device_address,
+                                              const char *device_name);
     virtual audio_policy_dev_state_t getDeviceConnectionState(
                                                                 audio_devices_t device,
                                                                 const char *device_address);
diff --git a/services/camera/libcameraservice/Android.mk b/services/camera/libcameraservice/Android.mk
index e184d97..5d6423a 100644
--- a/services/camera/libcameraservice/Android.mk
+++ b/services/camera/libcameraservice/Android.mk
@@ -23,8 +23,10 @@
 LOCAL_SRC_FILES:=               \
     CameraService.cpp \
     CameraDeviceFactory.cpp \
+    CameraFlashlight.cpp \
     common/Camera2ClientBase.cpp \
     common/CameraDeviceBase.cpp \
+    common/CameraModule.cpp \
     common/FrameProcessorBase.cpp \
     api1/CameraClient.cpp \
     api1/Camera2Client.cpp \
diff --git a/services/camera/libcameraservice/CameraFlashlight.cpp b/services/camera/libcameraservice/CameraFlashlight.cpp
new file mode 100644
index 0000000..00a70eb
--- /dev/null
+++ b/services/camera/libcameraservice/CameraFlashlight.cpp
@@ -0,0 +1,520 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "CameraFlashlight"
+#define ATRACE_TAG ATRACE_TAG_CAMERA
+#define LOG_NDEBUG 0
+
+#include <utils/Log.h>
+#include <utils/Trace.h>
+#include <cutils/properties.h>
+
+#include "camera/CameraMetadata.h"
+#include "CameraFlashlight.h"
+#include "gui/IGraphicBufferConsumer.h"
+#include "gui/BufferQueue.h"
+#include "camera/camera2/CaptureRequest.h"
+#include "CameraDeviceFactory.h"
+
+
+namespace android {
+
+CameraFlashlight::CameraFlashlight(CameraModule& cameraModule,
+        const camera_module_callbacks_t& callbacks) :
+        mCameraModule(&cameraModule),
+        mCallbacks(&callbacks) {
+}
+
+CameraFlashlight::~CameraFlashlight() {
+}
+
+status_t CameraFlashlight::createFlashlightControl(const String16& cameraId) {
+    ALOGV("%s: creating a flash light control for camera %s", __FUNCTION__,
+            cameraId.string());
+    if (mFlashControl != NULL) {
+        return INVALID_OPERATION;
+    }
+
+    status_t res = OK;
+
+    if (mCameraModule->getRawModule()->module_api_version >=
+            CAMERA_MODULE_API_VERSION_2_4) {
+        mFlashControl = new FlashControl(*mCameraModule, *mCallbacks);
+        if (mFlashControl == NULL) {
+            ALOGV("%s: cannot create flash control for module api v2.4+",
+                     __FUNCTION__);
+            return NO_MEMORY;
+        }
+    } else {
+        uint32_t deviceVersion = CAMERA_DEVICE_API_VERSION_1_0;
+
+        if (mCameraModule->getRawModule()->module_api_version >=
+                CAMERA_MODULE_API_VERSION_2_0) {
+            camera_info info;
+            res = mCameraModule->getCameraInfo(
+                    atoi(String8(cameraId).string()), &info);
+            if (res) {
+                ALOGV("%s: failed to get camera info for camera %s",
+                        __FUNCTION__, cameraId.string());
+                return res;
+            }
+            deviceVersion = info.device_version;
+        }
+
+        if (deviceVersion >= CAMERA_DEVICE_API_VERSION_2_0) {
+            CameraDeviceClientFlashControl *flashControl =
+                    new CameraDeviceClientFlashControl(*mCameraModule,
+                                                       *mCallbacks);
+            if (!flashControl) {
+                return NO_MEMORY;
+            }
+
+            mFlashControl = flashControl;
+        }
+        else {
+            // todo: implement for device api 1
+            return INVALID_OPERATION;
+        }
+    }
+
+    return OK;
+}
+
+status_t CameraFlashlight::setTorchMode(const String16& cameraId, bool enabled) {
+    if (!mCameraModule) {
+        return NO_INIT;
+    }
+
+    ALOGV("%s: set torch mode of camera %s to %d", __FUNCTION__,
+            cameraId.string(), enabled);
+
+    status_t res = OK;
+    Mutex::Autolock l(mLock);
+
+    if (mFlashControl == NULL) {
+        res = createFlashlightControl(cameraId);
+        if (res) {
+            return res;
+        }
+        res =  mFlashControl->setTorchMode(cameraId, enabled);
+        return res;
+    }
+
+    // if flash control already exists, turning on torch mode may fail if it's
+    // tied to another camera device for module v2.3 and below.
+    res = mFlashControl->setTorchMode(cameraId, enabled);
+    if (res == BAD_INDEX) {
+        // flash control is tied to another camera device, need to close it and
+        // try again.
+        mFlashControl.clear();
+        res = createFlashlightControl(cameraId);
+        if (res) {
+            return res;
+        }
+        res = mFlashControl->setTorchMode(cameraId, enabled);
+    }
+
+    return res;
+}
+
+bool CameraFlashlight::hasFlashUnit(const String16& cameraId) {
+    status_t res;
+
+    Mutex::Autolock l(mLock);
+
+    if (mFlashControl == NULL) {
+        res = createFlashlightControl(cameraId);
+        if (res) {
+            ALOGE("%s: failed to create flash control for %s ",
+                    __FUNCTION__, cameraId.string());
+            return false;
+        }
+    }
+
+    bool flashUnit = false;
+
+    // if flash control already exists, querying if a camera device has a flash
+    // unit may fail if it's module v1
+    res = mFlashControl->hasFlashUnit(cameraId, &flashUnit);
+    if (res == BAD_INDEX) {
+        // need to close the flash control before query.
+        mFlashControl.clear();
+        res = createFlashlightControl(cameraId);
+        if (res) {
+            ALOGE("%s: failed to create flash control for %s ", __FUNCTION__,
+                    cameraId.string());
+            return false;
+        }
+        res = mFlashControl->hasFlashUnit(cameraId, &flashUnit);
+        if (res) {
+            flashUnit = false;
+        }
+    }
+
+    return flashUnit;
+}
+
+status_t CameraFlashlight::prepareDeviceOpen() {
+    ALOGV("%s: prepare for device open", __FUNCTION__);
+
+    Mutex::Autolock l(mLock);
+
+    if (mCameraModule && mCameraModule->getRawModule()->module_api_version <
+            CAMERA_MODULE_API_VERSION_2_4) {
+        // framework is going to open a camera device, all flash light control
+        // should be closed for backward compatible support.
+        if (mFlashControl != NULL) {
+            mFlashControl.clear();
+        }
+    }
+
+    return OK;
+}
+
+
+FlashControlBase::~FlashControlBase() {
+}
+
+
+FlashControl::FlashControl(CameraModule& cameraModule,
+        const camera_module_callbacks_t& callbacks) :
+    mCameraModule(&cameraModule) {
+}
+
+FlashControl::~FlashControl() {
+}
+
+status_t FlashControl::hasFlashUnit(const String16& cameraId, bool *hasFlash) {
+    if (!hasFlash) {
+        return BAD_VALUE;
+    }
+
+    *hasFlash = false;
+
+    Mutex::Autolock l(mLock);
+
+    if (!mCameraModule) {
+        return NO_INIT;
+    }
+
+    camera_info info;
+    status_t res = mCameraModule->getCameraInfo(atoi(String8(cameraId).string()),
+            &info);
+    if (res != 0) {
+        return res;
+    }
+
+    CameraMetadata metadata;
+    metadata = info.static_camera_characteristics;
+    camera_metadata_entry flashAvailable =
+            metadata.find(ANDROID_FLASH_INFO_AVAILABLE);
+    if (flashAvailable.count == 1 && flashAvailable.data.u8[0] == 1) {
+        *hasFlash = true;
+    }
+
+    return OK;
+}
+
+status_t FlashControl::setTorchMode(const String16& cameraId, bool enabled) {
+    ALOGV("%s: set camera %s torch mode to %d", __FUNCTION__,
+            cameraId.string(), enabled);
+
+    Mutex::Autolock l(mLock);
+    if (!mCameraModule) {
+        return NO_INIT;
+    }
+
+    return mCameraModule->setTorchMode(String8(cameraId).string(), enabled);
+}
+
+CameraDeviceClientFlashControl::CameraDeviceClientFlashControl(
+        CameraModule& cameraModule,
+        const camera_module_callbacks_t& callbacks) :
+        mCameraModule(&cameraModule),
+        mCallbacks(&callbacks),
+        mTorchEnabled(false),
+        mMetadata(NULL) {
+}
+
+CameraDeviceClientFlashControl::~CameraDeviceClientFlashControl() {
+    if (mDevice != NULL) {
+        mDevice->flush();
+        mDevice->deleteStream(mStreamId);
+        mDevice.clear();
+    }
+    if (mMetadata) {
+        delete mMetadata;
+    }
+
+    mAnw.clear();
+    mSurfaceTexture.clear();
+    mProducer.clear();
+    mConsumer.clear();
+
+    if (mTorchEnabled) {
+        if (mCallbacks) {
+            ALOGV("%s: notify the framework that torch was turned off",
+                    __FUNCTION__);
+            mCallbacks->torch_mode_status_change(mCallbacks,
+                    String8(mCameraId).string(), TORCH_MODE_STATUS_OFF);
+        }
+    }
+}
+
+status_t CameraDeviceClientFlashControl::initializeSurface(int32_t width,
+        int32_t height) {
+    status_t res;
+    BufferQueue::createBufferQueue(&mProducer, &mConsumer);
+
+    mSurfaceTexture = new GLConsumer(mConsumer, 0, GLConsumer::TEXTURE_EXTERNAL,
+            true, true);
+    if (mSurfaceTexture == NULL) {
+        return NO_MEMORY;
+    }
+
+    int32_t format = HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED;
+    res = mSurfaceTexture->setDefaultBufferSize(width, height);
+    if (res) {
+        return res;
+    }
+    res = mSurfaceTexture->setDefaultBufferFormat(format);
+    if (res) {
+        return res;
+    }
+
+    bool useAsync = false;
+    int32_t consumerUsage;
+    res = mProducer->query(NATIVE_WINDOW_CONSUMER_USAGE_BITS, &consumerUsage);
+    if (res) {
+        return res;
+    }
+
+    if (consumerUsage & GraphicBuffer::USAGE_HW_TEXTURE) {
+        useAsync = true;
+    }
+
+    mAnw = new Surface(mProducer, useAsync);
+    if (mAnw == NULL) {
+        return NO_MEMORY;
+    }
+    res = mDevice->createStream(mAnw, width, height, format, &mStreamId);
+    if (res) {
+        return res;
+    }
+
+    res = mDevice->configureStreams();
+    if (res) {
+        return res;
+    }
+
+    return res;
+}
+
+status_t CameraDeviceClientFlashControl::getSmallestSurfaceSize(
+        const camera_info& info, int32_t *width, int32_t *height) {
+    if (!width || !height) {
+        return BAD_VALUE;
+    }
+
+    int32_t w = INT32_MAX;
+    int32_t h = 1;
+
+    CameraMetadata metadata;
+    metadata = info.static_camera_characteristics;
+    camera_metadata_entry streamConfigs =
+            metadata.find(ANDROID_SCALER_AVAILABLE_STREAM_CONFIGURATIONS);
+    for (size_t i = 0; i < streamConfigs.count; i += 4) {
+        int32_t fmt = streamConfigs.data.i32[i];
+        if (fmt == ANDROID_SCALER_AVAILABLE_FORMATS_IMPLEMENTATION_DEFINED) {
+            int32_t ww = streamConfigs.data.i32[i + 1];
+            int32_t hh = streamConfigs.data.i32[i + 2];
+
+            if (w* h > ww * hh) {
+                w = ww;
+                h = hh;
+            }
+        }
+    }
+
+    if (w == INT32_MAX) {
+        return NAME_NOT_FOUND;
+    }
+
+    *width = w;
+    *height = h;
+
+    return OK;
+}
+
+status_t CameraDeviceClientFlashControl::connectCameraDevice(
+        const String16& cameraId) {
+    String8 id = String8(cameraId);
+    camera_info info;
+    status_t res = mCameraModule->getCameraInfo(atoi(id.string()), &info);
+    if (res != 0) {
+        ALOGE("%s: failed to get camera info for camera %s", __FUNCTION__,
+                mCameraId.string());
+        return res;
+    }
+
+    mDevice = CameraDeviceFactory::createDevice(atoi(id.string()));
+    if (mDevice == NULL) {
+        return NO_MEMORY;
+    }
+
+    res = mDevice->initialize(mCameraModule);
+    if (res) {
+        goto fail;
+    }
+
+    int32_t width, height;
+    res = getSmallestSurfaceSize(info, &width, &height);
+    if (res) {
+        return res;
+    }
+    res = initializeSurface(width, height);
+    if (res) {
+        goto fail;
+    }
+
+    mCameraId = cameraId;
+
+    return OK;
+
+fail:
+    mDevice.clear();
+    return res;
+}
+
+
+status_t CameraDeviceClientFlashControl::hasFlashUnit(const String16& cameraId,
+        bool *hasFlash) {
+    ALOGV("%s: checking if camera %s has a flash unit", __FUNCTION__,
+            cameraId.string());
+
+    Mutex::Autolock l(mLock);
+    return hasFlashUnitLocked(cameraId, hasFlash);
+
+}
+
+status_t CameraDeviceClientFlashControl::hasFlashUnitLocked(
+        const String16& cameraId, bool *hasFlash) {
+    if (!mCameraModule) {
+        ALOGE("%s: camera module is NULL", __FUNCTION__);
+        return NO_INIT;
+    }
+
+    if (!hasFlash) {
+        return BAD_VALUE;
+    }
+
+    camera_info info;
+    status_t res = mCameraModule->getCameraInfo(
+            atoi(String8(cameraId).string()), &info);
+    if (res != 0) {
+        ALOGE("%s: failed to get camera info for camera %s", __FUNCTION__,
+                cameraId.string());
+        return res;
+    }
+
+    CameraMetadata metadata;
+    metadata = info.static_camera_characteristics;
+    camera_metadata_entry flashAvailable =
+            metadata.find(ANDROID_FLASH_INFO_AVAILABLE);
+    if (flashAvailable.count == 1 && flashAvailable.data.u8[0] == 1) {
+        *hasFlash = true;
+    }
+
+    return OK;
+}
+
+status_t CameraDeviceClientFlashControl::submitTorchRequest(bool enabled) {
+    status_t res;
+
+    if (mMetadata == NULL) {
+        mMetadata = new CameraMetadata();
+        if (mMetadata == NULL) {
+            return NO_MEMORY;
+        }
+        res = mDevice->createDefaultRequest(
+                CAMERA3_TEMPLATE_PREVIEW, mMetadata);
+        if (res) {
+            return res;
+        }
+    }
+
+    uint8_t torchOn = enabled ? ANDROID_FLASH_MODE_TORCH :
+                                ANDROID_FLASH_MODE_OFF;
+
+    mMetadata->update(ANDROID_FLASH_MODE, &torchOn, 1);
+    mMetadata->update(ANDROID_REQUEST_OUTPUT_STREAMS, &mStreamId, 1);
+
+    int32_t requestId = 0;
+    mMetadata->update(ANDROID_REQUEST_ID, &requestId, 1);
+
+    List<const CameraMetadata> metadataRequestList;
+    metadataRequestList.push_back(*mMetadata);
+
+    int64_t lastFrameNumber = 0;
+    res = mDevice->captureList(metadataRequestList, &lastFrameNumber);
+
+    return res;
+}
+
+
+status_t CameraDeviceClientFlashControl::setTorchMode(
+        const String16& cameraId, bool enabled) {
+    bool hasFlash = false;
+
+    Mutex::Autolock l(mLock);
+    status_t res = hasFlashUnitLocked(cameraId, &hasFlash);
+
+    // pre-check
+    if (enabled) {
+        // invalid camera?
+        if (res) {
+            return -EINVAL;
+        }
+        // no flash unit?
+        if (!hasFlash) {
+            return -ENOSYS;
+        }
+        // already opened for a different device?
+        if (mDevice != NULL && cameraId != mCameraId) {
+            return BAD_INDEX;
+        }
+    } else if (mDevice == NULL || cameraId != mCameraId) {
+        // disabling the torch mode of an un-opened or different device.
+        return OK;
+    }
+
+    if (mDevice == NULL) {
+        res = connectCameraDevice(cameraId);
+        if (res) {
+            return res;
+        }
+    }
+
+    res = submitTorchRequest(enabled);
+    if (res) {
+        return res;
+    }
+
+    mTorchEnabled = enabled;
+    return OK;
+}
+
+}
diff --git a/services/camera/libcameraservice/CameraFlashlight.h b/services/camera/libcameraservice/CameraFlashlight.h
new file mode 100644
index 0000000..a0de0b0
--- /dev/null
+++ b/services/camera/libcameraservice/CameraFlashlight.h
@@ -0,0 +1,149 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_SERVERS_CAMERA_CAMERAFLASHLIGHT_H
+#define ANDROID_SERVERS_CAMERA_CAMERAFLASHLIGHT_H
+
+#include "hardware/camera_common.h"
+#include "utils/KeyedVector.h"
+#include "gui/GLConsumer.h"
+#include "gui/Surface.h"
+#include "common/CameraDeviceBase.h"
+
+namespace android {
+
+/**
+ * FlashControlBase is a base class for flash control. It defines the functions
+ * that a flash control for each camera module/device version should implement.
+ */
+class FlashControlBase : public virtual VirtualLightRefBase {
+    public:
+        virtual ~FlashControlBase();
+
+        // Whether a camera device has a flash unit. Calling this function may
+        // cause the torch mode to be turned off in HAL v1 devices. If
+        // previously-on torch mode is turned off,
+        // callbacks.torch_mode_status_change() should be invoked.
+        virtual status_t hasFlashUnit(const String16& cameraId,
+                    bool *hasFlash) = 0;
+
+        // set the torch mode to on or off.
+        virtual status_t setTorchMode(const String16& cameraId,
+                    bool enabled) = 0;
+};
+
+/**
+ * CameraFlashlight can be used by camera service to control flashflight.
+ */
+class CameraFlashlight : public virtual VirtualLightRefBase {
+    public:
+        CameraFlashlight(CameraModule& cameraModule,
+                const camera_module_callbacks_t& callbacks);
+        virtual ~CameraFlashlight();
+
+        // set the torch mode to on or off.
+        status_t setTorchMode(const String16& cameraId, bool enabled);
+
+        // Whether a camera device has a flash unit. Calling this function may
+        // cause the torch mode to be turned off in HAL v1 devices.
+        bool hasFlashUnit(const String16& cameraId);
+
+        // Notify CameraFlashlight that camera service is going to open a camera
+        // device. CameraFlashlight will free the resources that may cause the
+        // camera open to fail. Camera service must call this function before
+        // opening a camera device.
+        status_t prepareDeviceOpen();
+
+    private:
+        // create flashlight control based on camera module API and camera
+        // device API versions.
+        status_t createFlashlightControl(const String16& cameraId);
+
+        sp<FlashControlBase> mFlashControl;
+        CameraModule *mCameraModule;
+        const camera_module_callbacks_t *mCallbacks;
+
+        Mutex mLock;
+};
+
+/**
+ * Flash control for camera module v2.4 and above.
+ */
+class FlashControl : public FlashControlBase {
+    public:
+        FlashControl(CameraModule& cameraModule,
+                const camera_module_callbacks_t& callbacks);
+        virtual ~FlashControl();
+
+        // FlashControlBase
+        status_t hasFlashUnit(const String16& cameraId, bool *hasFlash);
+        status_t setTorchMode(const String16& cameraId, bool enabled);
+
+    private:
+        CameraModule *mCameraModule;
+
+        Mutex mLock;
+};
+
+/**
+ * Flash control for camera module <= v2.3 and camera HAL v2-v3
+ */
+class CameraDeviceClientFlashControl : public FlashControlBase {
+    public:
+        CameraDeviceClientFlashControl(CameraModule& cameraModule,
+                const camera_module_callbacks_t& callbacks);
+        virtual ~CameraDeviceClientFlashControl();
+
+        // FlashControlBase
+        status_t setTorchMode(const String16& cameraId, bool enabled);
+        status_t hasFlashUnit(const String16& cameraId, bool *hasFlash);
+
+    private:
+        // connect to a camera device
+        status_t connectCameraDevice(const String16& cameraId);
+
+        // initialize a surface
+        status_t initializeSurface(int32_t width, int32_t height);
+
+        // submit a request with the given torch mode
+        status_t submitTorchRequest(bool enabled);
+
+        // get the smallest surface size of IMPLEMENTATION_DEFINED
+        status_t getSmallestSurfaceSize(const camera_info& info, int32_t *width,
+                    int32_t *height);
+
+        status_t hasFlashUnitLocked(const String16& cameraId, bool *hasFlash);
+
+        CameraModule *mCameraModule;
+        const camera_module_callbacks_t *mCallbacks;
+        String16 mCameraId;
+        bool mTorchEnabled;
+        CameraMetadata *mMetadata;
+
+        sp<CameraDeviceBase> mDevice;
+
+        sp<IGraphicBufferProducer> mProducer;
+        sp<IGraphicBufferConsumer>  mConsumer;
+        sp<GLConsumer> mSurfaceTexture;
+        sp<ANativeWindow> mAnw;
+        int32_t mStreamId;
+
+        Mutex mLock;
+};
+
+} // namespace android
+
+#endif
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index 1232c32..d65ac21 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -29,6 +29,7 @@
 #include <binder/MemoryHeapBase.h>
 #include <cutils/atomic.h>
 #include <cutils/properties.h>
+#include <cutils/multiuser.h>
 #include <gui/Surface.h>
 #include <hardware/hardware.h>
 #include <media/AudioSystem.h>
@@ -86,6 +87,38 @@
         camera_id,
         new_status);
 }
+
+static void torch_mode_status_change(
+        const struct camera_module_callbacks* callbacks,
+        const char* camera_id,
+        int new_status) {
+    if (!callbacks || !camera_id) {
+        ALOGE("%s invalid parameters. callbacks %p, camera_id %p", __FUNCTION__,
+                callbacks, camera_id);
+    }
+    sp<CameraService> cs = const_cast<CameraService*>(
+                                static_cast<const CameraService*>(callbacks));
+
+    ICameraServiceListener::TorchStatus status;
+    switch (new_status) {
+        case TORCH_MODE_STATUS_AVAILABLE:
+            status = ICameraServiceListener::TORCH_STATUS_AVAILABLE;
+            break;
+        case TORCH_MODE_STATUS_RESOURCE_BUSY:
+            status = ICameraServiceListener::TORCH_STATUS_NOT_AVAILABLE;
+            break;
+        case TORCH_MODE_STATUS_OFF:
+            status = ICameraServiceListener::TORCH_STATUS_OFF;
+            break;
+        default:
+            ALOGE("Unknown torch status %d", new_status);
+            return;
+    }
+
+    cs->onTorchStatusChanged(
+        String16(camera_id),
+        status);
+}
 } // extern "C"
 
 // ----------------------------------------------------------------------------
@@ -95,7 +128,7 @@
 static CameraService *gCameraService;
 
 CameraService::CameraService()
-    :mSoundRef(0), mModule(0)
+    :mSoundRef(0), mModule(0), mFlashlight(0)
 {
     ALOGI("CameraService started (pid=%d)", getpid());
     gCameraService = this;
@@ -105,6 +138,8 @@
     }
 
     this->camera_device_status_change = android::camera_device_status_change;
+    this->torch_mode_status_change = android::torch_mode_status_change;
+
 }
 
 void CameraService::onFirstRef()
@@ -113,14 +148,19 @@
 
     BnCameraService::onFirstRef();
 
+    camera_module_t *rawModule;
     if (hw_get_module(CAMERA_HARDWARE_MODULE_ID,
-                (const hw_module_t **)&mModule) < 0) {
+                (const hw_module_t **)&rawModule) < 0) {
         ALOGE("Could not load camera HAL module");
         mNumberOfCameras = 0;
     }
     else {
-        ALOGI("Loaded \"%s\" camera module", mModule->common.name);
-        mNumberOfCameras = mModule->get_number_of_cameras();
+        mModule = new CameraModule(rawModule);
+        mFlashlight = new CameraFlashlight(*mModule, *this);
+
+        const hw_module_t *common = mModule->getRawModule();
+        ALOGI("Loaded \"%s\" camera module", common->name);
+        mNumberOfCameras = mModule->getNumberOfCameras();
         if (mNumberOfCameras > MAX_CAMERAS) {
             ALOGE("Number of cameras(%d) > MAX_CAMERAS(%d).",
                     mNumberOfCameras, MAX_CAMERAS);
@@ -128,16 +168,21 @@
         }
         for (int i = 0; i < mNumberOfCameras; i++) {
             setCameraFree(i);
+
+            String16 cameraName = String16(String8::format("%d", i));
+            if (mFlashlight->hasFlashUnit(cameraName)) {
+                mTorchStatusMap.add(cameraName,
+                        ICameraServiceListener::TORCH_STATUS_AVAILABLE);
+            }
         }
 
-        if (mModule->common.module_api_version >=
-                CAMERA_MODULE_API_VERSION_2_1) {
-            mModule->set_callbacks(this);
+        if (common->module_api_version >= CAMERA_MODULE_API_VERSION_2_1) {
+            mModule->setCallbacks(this);
         }
 
         VendorTagDescriptor::clearGlobalVendorTagDescriptor();
 
-        if (mModule->common.module_api_version >= CAMERA_MODULE_API_VERSION_2_2) {
+        if (common->module_api_version >= CAMERA_MODULE_API_VERSION_2_2) {
             setUpVendorTags();
         }
 
@@ -152,6 +197,9 @@
         }
     }
 
+    if (mModule) {
+        delete mModule;
+    }
     VendorTagDescriptor::clearGlobalVendorTagDescriptor();
     gCameraService = NULL;
 }
@@ -220,6 +268,37 @@
 
 }
 
+void CameraService::onTorchStatusChanged(const String16& cameraId,
+        ICameraServiceListener::TorchStatus newStatus) {
+    Mutex::Autolock al(mTorchStatusMutex);
+    onTorchStatusChangedLocked(cameraId, newStatus);
+}
+
+void CameraService::onTorchStatusChangedLocked(const String16& cameraId,
+        ICameraServiceListener::TorchStatus newStatus) {
+    ALOGI("%s: Torch status changed for cameraId=%s, newStatus=%d",
+            __FUNCTION__, cameraId.string(), newStatus);
+
+    if (getTorchStatusLocked(cameraId) == newStatus) {
+        ALOGE("%s: Torch state transition to the same status 0x%x not allowed",
+              __FUNCTION__, (uint32_t)newStatus);
+        return;
+    }
+
+    status_t res = setTorchStatusLocked(cameraId, newStatus);
+    if (res) {
+        ALOGE("%s: Failed to set the torch status", __FUNCTION__,
+                (uint32_t)newStatus);
+        return;
+    }
+
+    Vector<sp<ICameraServiceListener> >::const_iterator it;
+    for (it = mListenerList.begin(); it != mListenerList.end(); ++it) {
+        (*it)->onTorchStatusChanged(newStatus, cameraId);
+    }
+}
+
+
 int32_t CameraService::getNumberOfCameras() {
     return mNumberOfCameras;
 }
@@ -236,7 +315,7 @@
 
     struct camera_info info;
     status_t rc = filterGetInfoErrorCode(
-        mModule->get_camera_info(cameraId, &info));
+        mModule->getCameraInfo(cameraId, &info));
     cameraInfo->facing = info.facing;
     cameraInfo->orientation = info.orientation;
     return rc;
@@ -347,7 +426,7 @@
 
     int facing;
     status_t ret = OK;
-    if (mModule->common.module_api_version < CAMERA_MODULE_API_VERSION_2_0 ||
+    if (mModule->getRawModule()->module_api_version < CAMERA_MODULE_API_VERSION_2_0 ||
             getDeviceVersion(cameraId, &facing) <= CAMERA_DEVICE_API_VERSION_2_1 ) {
         /**
          * Backwards compatibility mode for old HALs:
@@ -368,7 +447,7 @@
          * Normal HAL 2.1+ codepath.
          */
         struct camera_info info;
-        ret = filterGetInfoErrorCode(mModule->get_camera_info(cameraId, &info));
+        ret = filterGetInfoErrorCode(mModule->getCameraInfo(cameraId, &info));
         *cameraInfo = info.static_camera_characteristics;
     }
 
@@ -387,12 +466,12 @@
 
 int CameraService::getDeviceVersion(int cameraId, int* facing) {
     struct camera_info info;
-    if (mModule->get_camera_info(cameraId, &info) != OK) {
+    if (mModule->getCameraInfo(cameraId, &info) != OK) {
         return -1;
     }
 
     int deviceVersion;
-    if (mModule->common.module_api_version >= CAMERA_MODULE_API_VERSION_2_0) {
+    if (mModule->getRawModule()->module_api_version >= CAMERA_MODULE_API_VERSION_2_0) {
         deviceVersion = info.device_version;
     } else {
         deviceVersion = CAMERA_DEVICE_API_VERSION_1_0;
@@ -433,13 +512,13 @@
     vendor_tag_ops_t vOps = vendor_tag_ops_t();
 
     // Check if vendor operations have been implemented
-    if (mModule->get_vendor_tag_ops == NULL) {
+    if (!mModule->isVendorTagDefined()) {
         ALOGI("%s: No vendor tags defined for this device.", __FUNCTION__);
         return false;
     }
 
     ATRACE_BEGIN("camera3->get_metadata_vendor_tag_ops");
-    mModule->get_vendor_tag_ops(&vOps);
+    mModule->getVendorTagOps(&vOps);
     ATRACE_END();
 
     // Ensure all vendor operations are present
@@ -592,7 +671,10 @@
     }
 
     char value[PROPERTY_VALUE_MAX];
-    property_get("sys.secpolicy.camera.disabled", value, "0");
+    char key[PROPERTY_KEY_MAX];
+    int clientUserId = multiuser_get_user_id(clientUid);
+    snprintf(key, PROPERTY_KEY_MAX, "sys.secpolicy.camera.off_%d", clientUserId);
+    property_get(key, value, "0");
     if (strcmp(value, "1") == 0) {
         // Camera is disabled by DevicePolicyManager.
         ALOGI("Camera is disabled. connect X (pid %d) rejected", callingPid);
@@ -671,6 +753,9 @@
         int halVersion,
         bool legacyMode) {
 
+    // give flashlight a chance to close devices if necessary.
+    mFlashlight->prepareDeviceOpen();
+
     int facing = -1;
     int deviceVersion = getDeviceVersion(cameraId, &facing);
 
@@ -789,8 +874,9 @@
         /*out*/
         sp<ICamera>& device) {
 
+    int apiVersion = mModule->getRawModule()->module_api_version;
     if (halVersion != CAMERA_HAL_API_VERSION_UNSPECIFIED &&
-            mModule->common.module_api_version < CAMERA_MODULE_API_VERSION_2_3) {
+            apiVersion < CAMERA_MODULE_API_VERSION_2_3) {
         /*
          * Either the HAL version is unspecified in which case this just creates
          * a camera client selected by the latest device version, or
@@ -798,7 +884,7 @@
          * the open_legacy call
          */
         ALOGE("%s: camera HAL module version %x doesn't support connecting to legacy HAL devices!",
-                __FUNCTION__, mModule->common.module_api_version);
+                __FUNCTION__, apiVersion);
         return INVALID_OPERATION;
     }
 
@@ -846,6 +932,47 @@
     return OK;
 }
 
+status_t CameraService::setTorchMode(const String16& cameraId, bool enabled,
+        const sp<IBinder>& clientBinder) {
+    if (enabled && clientBinder == NULL) {
+        ALOGE("%s: torch client binder is NULL", __FUNCTION__);
+        return -ENOSYS;
+    }
+
+    Mutex::Autolock al(mTorchStatusMutex);
+    status_t res = mFlashlight->setTorchMode(cameraId, enabled);
+    if (res) {
+        ALOGE("%s: setting torch mode of camera %s to %d failed", __FUNCTION__,
+                cameraId.string(), enabled);
+        return res;
+    }
+
+    // update the link to client's death
+    ssize_t index = mTorchClientMap.indexOfKey(cameraId);
+    if (enabled) {
+        if (index == NAME_NOT_FOUND) {
+            mTorchClientMap.add(cameraId, clientBinder);
+        } else {
+            const sp<IBinder> oldBinder = mTorchClientMap.valueAt(index);
+            oldBinder->unlinkToDeath(this);
+
+            mTorchClientMap.replaceValueAt(index, clientBinder);
+        }
+        clientBinder->linkToDeath(this);
+    } else if (index != NAME_NOT_FOUND) {
+        sp<IBinder> oldBinder = mTorchClientMap.valueAt(index);
+        oldBinder->unlinkToDeath(this);
+    }
+
+    // notify the listeners the change.
+    ICameraServiceListener::TorchStatus status = enabled ?
+            ICameraServiceListener::TORCH_STATUS_ON :
+            ICameraServiceListener::TORCH_STATUS_OFF;
+    onTorchStatusChangedLocked(cameraId, status);
+
+    return OK;
+}
+
 status_t CameraService::connectFinishUnsafe(const sp<BasicClient>& client,
                                             const sp<IBinder>& remoteCallback) {
     status_t status = client->initialize(mModule);
@@ -971,6 +1098,9 @@
         int facing = -1;
         int deviceVersion = getDeviceVersion(cameraId, &facing);
 
+        // give flashlight a chance to close devices if necessary.
+        mFlashlight->prepareDeviceOpen();
+
         switch(deviceVersion) {
           case CAMERA_DEVICE_API_VERSION_1_0:
             ALOGW("Camera using old HAL version: %d", deviceVersion);
@@ -1042,6 +1172,16 @@
         }
     }
 
+    /* Immediately signal current torch status to this listener only */
+    {
+        Mutex::Autolock al(mTorchStatusMutex);
+        for (size_t i = 0; i < mTorchStatusMap.size(); i++ ) {
+            listener->onTorchStatusChanged(mTorchStatusMap.valueAt(i),
+                    mTorchStatusMap.keyAt(i));
+        }
+
+    }
+
     return OK;
 }
 status_t CameraService::removeListener(
@@ -1633,14 +1773,11 @@
             return NO_ERROR;
         }
 
-        result = String8::format("Camera module HAL API version: 0x%x\n",
-                mModule->common.hal_api_version);
-        result.appendFormat("Camera module API version: 0x%x\n",
-                mModule->common.module_api_version);
-        result.appendFormat("Camera module name: %s\n",
-                mModule->common.name);
-        result.appendFormat("Camera module author: %s\n",
-                mModule->common.author);
+        const hw_module_t* common = mModule->getRawModule();
+        result = String8::format("Camera module HAL API version: 0x%x\n", common->hal_api_version);
+        result.appendFormat("Camera module API version: 0x%x\n", common->module_api_version);
+        result.appendFormat("Camera module name: %s\n", common->name);
+        result.appendFormat("Camera module author: %s\n", common->author);
         result.appendFormat("Number of camera devices: %d\n\n", mNumberOfCameras);
 
         sp<VendorTagDescriptor> desc = VendorTagDescriptor::getGlobalVendorTagDescriptor();
@@ -1660,7 +1797,7 @@
             result = String8::format("Camera %d static information:\n", i);
             camera_info info;
 
-            status_t rc = mModule->get_camera_info(i, &info);
+            status_t rc = mModule->getCameraInfo(i, &info);
             if (rc != OK) {
                 result.appendFormat("  Error reading static information!\n");
                 write(fd, result.string(), result.size());
@@ -1669,8 +1806,7 @@
                         info.facing == CAMERA_FACING_BACK ? "BACK" : "FRONT");
                 result.appendFormat("  Orientation: %d\n", info.orientation);
                 int deviceVersion;
-                if (mModule->common.module_api_version <
-                        CAMERA_MODULE_API_VERSION_2_0) {
+                if (common->module_api_version < CAMERA_MODULE_API_VERSION_2_0) {
                     deviceVersion = CAMERA_DEVICE_API_VERSION_1_0;
                 } else {
                     deviceVersion = info.device_version;
@@ -1725,6 +1861,23 @@
     return NO_ERROR;
 }
 
+void CameraService::handleTorchClientBinderDied(const wp<IBinder> &who) {
+    Mutex::Autolock al(mTorchStatusMutex);
+    for (size_t i = 0; i < mTorchClientMap.size(); i++) {
+        if (mTorchClientMap[i] == who) {
+            // turn off the torch mode that was turned on by dead client
+            String16 cameraId = mTorchClientMap.keyAt(i);
+            mFlashlight->setTorchMode(cameraId, false);
+            mTorchClientMap.removeItemsAt(i);
+
+            // notify torch mode was turned off
+            onTorchStatusChangedLocked(cameraId,
+                    ICameraServiceListener::TORCH_STATUS_OFF);
+            break;
+        }
+    }
+}
+
 /*virtual*/void CameraService::binderDied(
     const wp<IBinder> &who) {
 
@@ -1735,6 +1888,10 @@
 
     ALOGV("java clients' binder died");
 
+    // check torch client
+    handleTorchClientBinderDied(who);
+
+    // check camera device client
     sp<BasicClient> cameraClient = getClientByRemote(who);
 
     if (cameraClient == 0) {
@@ -1828,4 +1985,27 @@
     return mStatusList[cameraId];
 }
 
+ICameraServiceListener::TorchStatus CameraService::getTorchStatusLocked(
+        const String16& cameraId) const {
+    ssize_t index = mTorchStatusMap.indexOfKey(cameraId);
+    if (index == NAME_NOT_FOUND) {
+        return ICameraServiceListener::TORCH_STATUS_NOT_AVAILABLE;
+    }
+
+    return mTorchStatusMap.valueAt(index);
+}
+
+status_t CameraService::setTorchStatusLocked(const String16& cameraId,
+        ICameraServiceListener::TorchStatus status) {
+    ssize_t index = mTorchStatusMap.indexOfKey(cameraId);
+    if (index == NAME_NOT_FOUND) {
+        return BAD_VALUE;
+    }
+    ICameraServiceListener::TorchStatus& item =
+            mTorchStatusMap.editValueAt(index);
+    item = status;
+
+    return OK;
+}
+
 }; // namespace android
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index 126d8d9..84bcdb8 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -36,6 +36,10 @@
 #include <camera/CameraParameters.h>
 
 #include <camera/ICameraServiceListener.h>
+#include "CameraFlashlight.h"
+
+
+#include "common/CameraModule.h"
 
 /* This needs to be increased if we can have more cameras */
 #define MAX_CAMERAS 2
@@ -68,6 +72,9 @@
     // HAL Callbacks
     virtual void        onDeviceStatusChanged(int cameraId,
                                               int newStatus);
+    virtual void        onTorchStatusChanged(const String16& cameraId,
+                                             ICameraServiceListener::TorchStatus
+                                                   newStatus);
 
     /////////////////////////////////////////////////////////////////////
     // ICameraService
@@ -110,6 +117,9 @@
             /*out*/
             String16* parameters);
 
+    virtual status_t    setTorchMode(const String16& cameraId, bool enabled,
+            const sp<IBinder>& clientBinder);
+
     // OK = supports api of that version, -EOPNOTSUPP = does not support
     virtual status_t    supportsCameraApi(
             int cameraId, int apiVersion);
@@ -153,7 +163,7 @@
 
     class BasicClient : public virtual RefBase {
     public:
-        virtual status_t    initialize(camera_module_t *module) = 0;
+        virtual status_t    initialize(CameraModule *module) = 0;
         virtual void        disconnect();
 
         // because we can't virtually inherit IInterface, which breaks
@@ -385,7 +395,7 @@
     sp<MediaPlayer>     mSoundPlayer[NUM_SOUNDS];
     int                 mSoundRef;  // reference count (release all MediaPlayer when 0)
 
-    camera_module_t *mModule;
+    CameraModule*     mModule;
 
     Vector<sp<ICameraServiceListener> >
                         mListenerList;
@@ -406,6 +416,32 @@
                             int32_t cameraId,
                             const StatusVector *rejectSourceStates = NULL);
 
+    // flashlight control
+    sp<CameraFlashlight> mFlashlight;
+    // guard mTorchStatusMap and mTorchClientMap
+    Mutex                mTorchStatusMutex;
+    // camera id -> torch status
+    KeyedVector<String16, ICameraServiceListener::TorchStatus> mTorchStatusMap;
+    // camera id -> torch client binder
+    // only store the last client that turns on each camera's torch mode
+    KeyedVector<String16, sp<IBinder> > mTorchClientMap;
+
+    // check and handle if torch client's process has died
+    void handleTorchClientBinderDied(const wp<IBinder> &who);
+
+    // handle torch mode status change and invoke callbacks. mTorchStatusMutex
+    // should be locked.
+    void onTorchStatusChangedLocked(const String16& cameraId,
+            ICameraServiceListener::TorchStatus newStatus);
+
+    // get a camera's torch status. mTorchStatusMutex should be locked.
+    ICameraServiceListener::TorchStatus getTorchStatusLocked(
+            const String16 &cameraId) const;
+
+    // set a camera's torch status. mTorchStatusMutex should be locked.
+    status_t setTorchStatusLocked(const String16 &cameraId,
+            ICameraServiceListener::TorchStatus status);
+
     // IBinder::DeathRecipient implementation
     virtual void        binderDied(const wp<IBinder> &who);
 
diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp
index 0ed5586..5dbdeb2 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.cpp
+++ b/services/camera/libcameraservice/api1/Camera2Client.cpp
@@ -67,7 +67,7 @@
     mLegacyMode = legacyMode;
 }
 
-status_t Camera2Client::initialize(camera_module_t *module)
+status_t Camera2Client::initialize(CameraModule *module)
 {
     ATRACE_CALL();
     ALOGV("%s: Initializing client for camera %d", __FUNCTION__, mCameraId);
@@ -1959,7 +1959,7 @@
             return width * height * 2;
         case HAL_PIXEL_FORMAT_RGBA_8888:
             return width * height * 4;
-        case HAL_PIXEL_FORMAT_RAW_SENSOR:
+        case HAL_PIXEL_FORMAT_RAW16:
             return width * height * 2;
         default:
             ALOGE("%s: Unknown preview format: %x",
diff --git a/services/camera/libcameraservice/api1/Camera2Client.h b/services/camera/libcameraservice/api1/Camera2Client.h
index d68bb29..5a8241f 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.h
+++ b/services/camera/libcameraservice/api1/Camera2Client.h
@@ -94,7 +94,7 @@
 
     virtual ~Camera2Client();
 
-    status_t initialize(camera_module_t *module);
+    status_t initialize(CameraModule *module);
 
     virtual status_t dump(int fd, const Vector<String16>& args);
 
diff --git a/services/camera/libcameraservice/api1/CameraClient.cpp b/services/camera/libcameraservice/api1/CameraClient.cpp
index bbb2fe0..6bea3b6 100644
--- a/services/camera/libcameraservice/api1/CameraClient.cpp
+++ b/services/camera/libcameraservice/api1/CameraClient.cpp
@@ -59,7 +59,7 @@
     LOG1("CameraClient::CameraClient X (pid %d, id %d)", callingPid, cameraId);
 }
 
-status_t CameraClient::initialize(camera_module_t *module) {
+status_t CameraClient::initialize(CameraModule *module) {
     int callingPid = getCallingPid();
     status_t res;
 
@@ -75,7 +75,7 @@
     snprintf(camera_device_name, sizeof(camera_device_name), "%d", mCameraId);
 
     mHardware = new CameraHardwareInterface(camera_device_name);
-    res = mHardware->initialize(&module->common);
+    res = mHardware->initialize(module);
     if (res != OK) {
         ALOGE("%s: Camera %d: unable to initialize device: %s (%d)",
                 __FUNCTION__, mCameraId, strerror(-res), res);
diff --git a/services/camera/libcameraservice/api1/CameraClient.h b/services/camera/libcameraservice/api1/CameraClient.h
index 63a9d0f..95616b2 100644
--- a/services/camera/libcameraservice/api1/CameraClient.h
+++ b/services/camera/libcameraservice/api1/CameraClient.h
@@ -68,7 +68,7 @@
             bool legacyMode = false);
     ~CameraClient();
 
-    status_t initialize(camera_module_t *module);
+    status_t initialize(CameraModule *module);
 
     status_t dump(int fd, const Vector<String16>& args);
 
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.cpp b/services/camera/libcameraservice/api1/client2/Parameters.cpp
index 4f4cfb0..87e0132 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.cpp
+++ b/services/camera/libcameraservice/api1/client2/Parameters.cpp
@@ -182,9 +182,9 @@
                 supportedPreviewFormats +=
                     CameraParameters::PIXEL_FORMAT_YUV420SP;
                 break;
-            // Not advertizing JPEG, RAW_SENSOR, etc, for preview formats
+            // Not advertizing JPEG, RAW16, etc, for preview formats
             case HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED:
-            case HAL_PIXEL_FORMAT_RAW_SENSOR:
+            case HAL_PIXEL_FORMAT_RAW16:
             case HAL_PIXEL_FORMAT_BLOB:
                 addComma = false;
                 break;
@@ -2253,7 +2253,7 @@
         case HAL_PIXEL_FORMAT_RGBA_8888:   // RGBA8888
             fmt = CameraParameters::PIXEL_FORMAT_RGBA8888;
             break;
-        case HAL_PIXEL_FORMAT_RAW_SENSOR:
+        case HAL_PIXEL_FORMAT_RAW16:
             ALOGW("Raw sensor preview format requested.");
             fmt = CameraParameters::PIXEL_FORMAT_BAYER_RGGB;
             break;
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
index 6a1ee44..acc092c 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
@@ -71,7 +71,7 @@
     ALOGI("CameraDeviceClient %d: Opened", cameraId);
 }
 
-status_t CameraDeviceClient::initialize(camera_module_t *module)
+status_t CameraDeviceClient::initialize(CameraModule *module)
 {
     ATRACE_CALL();
     status_t res;
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.h b/services/camera/libcameraservice/api2/CameraDeviceClient.h
index 84e46b7..e687175 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.h
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.h
@@ -119,7 +119,7 @@
             int servicePid);
     virtual ~CameraDeviceClient();
 
-    virtual status_t      initialize(camera_module_t *module);
+    virtual status_t      initialize(CameraModule *module);
 
     virtual status_t      dump(int fd, const Vector<String16>& args);
 
diff --git a/services/camera/libcameraservice/api_pro/ProCamera2Client.cpp b/services/camera/libcameraservice/api_pro/ProCamera2Client.cpp
index 59e5083..30a89c2 100644
--- a/services/camera/libcameraservice/api_pro/ProCamera2Client.cpp
+++ b/services/camera/libcameraservice/api_pro/ProCamera2Client.cpp
@@ -50,7 +50,7 @@
     mExclusiveLock = false;
 }
 
-status_t ProCamera2Client::initialize(camera_module_t *module)
+status_t ProCamera2Client::initialize(CameraModule *module)
 {
     ATRACE_CALL();
     status_t res;
diff --git a/services/camera/libcameraservice/api_pro/ProCamera2Client.h b/services/camera/libcameraservice/api_pro/ProCamera2Client.h
index 9d83122..7f5f6ac 100644
--- a/services/camera/libcameraservice/api_pro/ProCamera2Client.h
+++ b/services/camera/libcameraservice/api_pro/ProCamera2Client.h
@@ -85,7 +85,7 @@
             int servicePid);
     virtual ~ProCamera2Client();
 
-    virtual status_t      initialize(camera_module_t *module);
+    virtual status_t      initialize(CameraModule *module);
 
     virtual status_t      dump(int fd, const Vector<String16>& args);
 
diff --git a/services/camera/libcameraservice/common/Camera2ClientBase.cpp b/services/camera/libcameraservice/common/Camera2ClientBase.cpp
index 453c8bd..0415d67 100644
--- a/services/camera/libcameraservice/common/Camera2ClientBase.cpp
+++ b/services/camera/libcameraservice/common/Camera2ClientBase.cpp
@@ -78,7 +78,7 @@
 }
 
 template <typename TClientBase>
-status_t Camera2ClientBase<TClientBase>::initialize(camera_module_t *module) {
+status_t Camera2ClientBase<TClientBase>::initialize(CameraModule *module) {
     ATRACE_CALL();
     ALOGV("%s: Initializing client for camera %d", __FUNCTION__,
           TClientBase::mCameraId);
diff --git a/services/camera/libcameraservice/common/Camera2ClientBase.h b/services/camera/libcameraservice/common/Camera2ClientBase.h
index e09c1b5..eb21d55 100644
--- a/services/camera/libcameraservice/common/Camera2ClientBase.h
+++ b/services/camera/libcameraservice/common/Camera2ClientBase.h
@@ -18,6 +18,7 @@
 #define ANDROID_SERVERS_CAMERA_CAMERA2CLIENT_BASE_H
 
 #include "common/CameraDeviceBase.h"
+#include "common/CameraModule.h"
 #include "camera/CaptureResult.h"
 
 namespace android {
@@ -55,7 +56,7 @@
                       int servicePid);
     virtual ~Camera2ClientBase();
 
-    virtual status_t      initialize(camera_module_t *module);
+    virtual status_t      initialize(CameraModule *module);
     virtual status_t      dump(int fd, const Vector<String16>& args);
 
     /**
diff --git a/services/camera/libcameraservice/common/CameraDeviceBase.h b/services/camera/libcameraservice/common/CameraDeviceBase.h
index d26e20c..06615f6 100644
--- a/services/camera/libcameraservice/common/CameraDeviceBase.h
+++ b/services/camera/libcameraservice/common/CameraDeviceBase.h
@@ -29,6 +29,7 @@
 #include "hardware/camera3.h"
 #include "camera/CameraMetadata.h"
 #include "camera/CaptureResult.h"
+#include "common/CameraModule.h"
 
 namespace android {
 
@@ -45,7 +46,7 @@
      */
     virtual int      getId() const = 0;
 
-    virtual status_t initialize(camera_module_t *module) = 0;
+    virtual status_t initialize(CameraModule *module) = 0;
     virtual status_t disconnect() = 0;
 
     virtual status_t dump(int fd, const Vector<String16> &args) = 0;
diff --git a/services/camera/libcameraservice/common/CameraModule.cpp b/services/camera/libcameraservice/common/CameraModule.cpp
new file mode 100644
index 0000000..bbf47e8
--- /dev/null
+++ b/services/camera/libcameraservice/common/CameraModule.cpp
@@ -0,0 +1,130 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "CameraModule"
+//#define LOG_NDEBUG 0
+
+#include "CameraModule.h"
+
+namespace android {
+
+void CameraModule::deriveCameraCharacteristicsKeys(
+        uint32_t deviceVersion, CameraMetadata &chars) {
+    // HAL1 devices should not reach here
+    if (deviceVersion < CAMERA_DEVICE_API_VERSION_2_0) {
+        ALOGV("%s: Cannot derive keys for HAL version < 2.0");
+        return;
+    }
+
+    // Keys added in HAL3.3
+    if (deviceVersion < CAMERA_DEVICE_API_VERSION_3_3) {
+        Vector<uint8_t> controlModes;
+        uint8_t data = ANDROID_CONTROL_AE_LOCK_AVAILABLE_TRUE;
+        chars.update(ANDROID_CONTROL_AE_LOCK_AVAILABLE, &data, /*count*/1);
+        data = ANDROID_CONTROL_AWB_LOCK_AVAILABLE_TRUE;
+        chars.update(ANDROID_CONTROL_AWB_LOCK_AVAILABLE, &data, /*count*/1);
+        controlModes.push(ANDROID_CONTROL_MODE_OFF);
+        controlModes.push(ANDROID_CONTROL_MODE_AUTO);
+        camera_metadata_entry entry = chars.find(ANDROID_CONTROL_AVAILABLE_SCENE_MODES);
+        if (entry.count > 1 || entry.data.u8[0] != ANDROID_CONTROL_SCENE_MODE_DISABLED) {
+            controlModes.push(ANDROID_CONTROL_MODE_USE_SCENE_MODE);
+        }
+        chars.update(ANDROID_CONTROL_AVAILABLE_MODES, controlModes);
+    }
+    return;
+}
+
+CameraModule::CameraModule(camera_module_t *module) {
+    if (module == NULL) {
+        ALOGE("%s: camera hardware module must not be null", __FUNCTION__);
+        assert(0);
+    }
+
+    mModule = module;
+    for (int i = 0; i < MAX_CAMERAS_PER_MODULE; i++) {
+        mCameraInfoCached[i] = false;
+    }
+}
+
+int CameraModule::getCameraInfo(int cameraId, struct camera_info *info) {
+    Mutex::Autolock lock(mCameraInfoLock);
+    if (cameraId < 0 || cameraId >= MAX_CAMERAS_PER_MODULE) {
+        ALOGE("%s: Invalid camera ID %d", __FUNCTION__, cameraId);
+        return -EINVAL;
+    }
+
+    camera_info &wrappedInfo = mCameraInfo[cameraId];
+    if (!mCameraInfoCached[cameraId]) {
+        camera_info rawInfo;
+        int ret = mModule->get_camera_info(cameraId, &rawInfo);
+        if (ret != 0) {
+            return ret;
+        }
+        CameraMetadata &m = mCameraCharacteristics[cameraId];
+        m = rawInfo.static_camera_characteristics;
+        int deviceVersion;
+        int apiVersion = mModule->common.module_api_version;
+        if (apiVersion >= CAMERA_MODULE_API_VERSION_2_0) {
+            deviceVersion = rawInfo.device_version;
+        } else {
+            deviceVersion = CAMERA_DEVICE_API_VERSION_1_0;
+        }
+        deriveCameraCharacteristicsKeys(deviceVersion, m);
+        wrappedInfo = rawInfo;
+        wrappedInfo.static_camera_characteristics = m.getAndLock();
+        mCameraInfoCached[cameraId] = true;
+    }
+    *info = wrappedInfo;
+    return 0;
+}
+
+int CameraModule::open(const char* id, struct hw_device_t** device) {
+    return mModule->common.methods->open(&mModule->common, id, device);
+}
+
+int CameraModule::openLegacy(
+        const char* id, uint32_t halVersion, struct hw_device_t** device) {
+    return mModule->open_legacy(&mModule->common, id, halVersion, device);
+}
+
+const hw_module_t* CameraModule::getRawModule() {
+    return &mModule->common;
+}
+
+int CameraModule::getNumberOfCameras() {
+    return mModule->get_number_of_cameras();
+}
+
+int CameraModule::setCallbacks(const camera_module_callbacks_t *callbacks) {
+    return mModule->set_callbacks(callbacks);
+}
+
+bool CameraModule::isVendorTagDefined() {
+    return mModule->get_vendor_tag_ops != NULL;
+}
+
+void CameraModule::getVendorTagOps(vendor_tag_ops_t* ops) {
+    if (mModule->get_vendor_tag_ops) {
+        mModule->get_vendor_tag_ops(ops);
+    }
+}
+
+int CameraModule::setTorchMode(const char* camera_id, bool enable) {
+    return mModule->set_torch_mode(camera_id, enable);
+}
+
+}; // namespace android
+
diff --git a/services/camera/libcameraservice/common/CameraModule.h b/services/camera/libcameraservice/common/CameraModule.h
new file mode 100644
index 0000000..31b9ae2
--- /dev/null
+++ b/services/camera/libcameraservice/common/CameraModule.h
@@ -0,0 +1,63 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_SERVERS_CAMERA_CAMERAMODULE_H
+#define ANDROID_SERVERS_CAMERA_CAMERAMODULE_H
+
+#include <hardware/camera.h>
+#include <camera/CameraMetadata.h>
+#include <utils/Mutex.h>
+
+/* This needs to be increased if we can have more cameras */
+#define MAX_CAMERAS_PER_MODULE 2
+
+
+namespace android {
+/**
+ * A wrapper class for HAL camera module.
+ *
+ * This class wraps camera_module_t returned from HAL to provide a wrapped
+ * get_camera_info implementation which CameraService generates some
+ * camera characteristics keys defined in newer HAL version on an older HAL.
+ */
+class CameraModule {
+public:
+    CameraModule(camera_module_t *module);
+
+    const hw_module_t* getRawModule();
+    int getCameraInfo(int cameraId, struct camera_info *info);
+    int getNumberOfCameras(void);
+    int open(const char* id, struct hw_device_t** device);
+    int openLegacy(const char* id, uint32_t halVersion, struct hw_device_t** device);
+    int setCallbacks(const camera_module_callbacks_t *callbacks);
+    bool isVendorTagDefined();
+    void getVendorTagOps(vendor_tag_ops_t* ops);
+    int setTorchMode(const char* camera_id, bool enable);
+
+private:
+    // Derive camera characteristics keys defined after HAL device version
+    static void deriveCameraCharacteristicsKeys(uint32_t deviceVersion, CameraMetadata &chars);
+    camera_module_t *mModule;
+    CameraMetadata mCameraCharacteristics[MAX_CAMERAS_PER_MODULE];
+    camera_info mCameraInfo[MAX_CAMERAS_PER_MODULE];
+    bool mCameraInfoCached[MAX_CAMERAS_PER_MODULE];
+    Mutex mCameraInfoLock;
+};
+
+} // namespace android
+
+#endif
+
diff --git a/services/camera/libcameraservice/device1/CameraHardwareInterface.h b/services/camera/libcameraservice/device1/CameraHardwareInterface.h
index 1935c2b..9e1cdc9 100644
--- a/services/camera/libcameraservice/device1/CameraHardwareInterface.h
+++ b/services/camera/libcameraservice/device1/CameraHardwareInterface.h
@@ -89,24 +89,23 @@
         }
     }
 
-    status_t initialize(hw_module_t *module)
+    status_t initialize(CameraModule *module)
     {
         ALOGI("Opening camera %s", mName.string());
-        camera_module_t *cameraModule = reinterpret_cast<camera_module_t *>(module);
         camera_info info;
-        status_t res = cameraModule->get_camera_info(atoi(mName.string()), &info);
+        status_t res = module->getCameraInfo(atoi(mName.string()), &info);
         if (res != OK) return res;
 
         int rc = OK;
-        if (module->module_api_version >= CAMERA_MODULE_API_VERSION_2_3 &&
+        if (module->getRawModule()->module_api_version >= CAMERA_MODULE_API_VERSION_2_3 &&
             info.device_version > CAMERA_DEVICE_API_VERSION_1_0) {
             // Open higher version camera device as HAL1.0 device.
-            rc = cameraModule->open_legacy(module, mName.string(),
-                                               CAMERA_DEVICE_API_VERSION_1_0,
-                                               (hw_device_t **)&mDevice);
+            rc = module->openLegacy(mName.string(),
+                                     CAMERA_DEVICE_API_VERSION_1_0,
+                                     (hw_device_t **)&mDevice);
         } else {
-            rc = CameraService::filterOpenErrorCode(module->methods->open(
-                module, mName.string(), (hw_device_t **)&mDevice));
+            rc = CameraService::filterOpenErrorCode(module->open(
+                    mName.string(), (hw_device_t **)&mDevice));
         }
         if (rc != OK) {
             ALOGE("Could not open camera %s: %d", mName.string(), rc);
diff --git a/services/camera/libcameraservice/device2/Camera2Device.cpp b/services/camera/libcameraservice/device2/Camera2Device.cpp
index d1158d6..be66c4d 100644
--- a/services/camera/libcameraservice/device2/Camera2Device.cpp
+++ b/services/camera/libcameraservice/device2/Camera2Device.cpp
@@ -53,7 +53,7 @@
     return mId;
 }
 
-status_t Camera2Device::initialize(camera_module_t *module)
+status_t Camera2Device::initialize(CameraModule *module)
 {
     ATRACE_CALL();
     ALOGV("%s: Initializing device for camera %d", __FUNCTION__, mId);
@@ -68,8 +68,8 @@
 
     camera2_device_t *device;
 
-    res = CameraService::filterOpenErrorCode(module->common.methods->open(
-        &module->common, name, reinterpret_cast<hw_device_t**>(&device)));
+    res = CameraService::filterOpenErrorCode(module->open(
+            name, reinterpret_cast<hw_device_t**>(&device)));
 
     if (res != OK) {
         ALOGE("%s: Could not open camera %d: %s (%d)", __FUNCTION__,
@@ -87,7 +87,7 @@
     }
 
     camera_info info;
-    res = module->get_camera_info(mId, &info);
+    res = module->getCameraInfo(mId, &info);
     if (res != OK ) return res;
 
     if (info.device_version != device->common.version) {
diff --git a/services/camera/libcameraservice/device2/Camera2Device.h b/services/camera/libcameraservice/device2/Camera2Device.h
index 4def8ae..1cc5482 100644
--- a/services/camera/libcameraservice/device2/Camera2Device.h
+++ b/services/camera/libcameraservice/device2/Camera2Device.h
@@ -43,7 +43,7 @@
      * CameraDevice interface
      */
     virtual int      getId() const;
-    virtual status_t initialize(camera_module_t *module);
+    virtual status_t initialize(CameraModule *module);
     virtual status_t disconnect();
     virtual status_t dump(int fd, const Vector<String16>& args);
     virtual const CameraMetadata& info() const;
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index 53e6fa9..9a4e5ac 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -86,7 +86,7 @@
  * CameraDeviceBase interface
  */
 
-status_t Camera3Device::initialize(camera_module_t *module)
+status_t Camera3Device::initialize(CameraModule *module)
 {
     ATRACE_CALL();
     Mutex::Autolock il(mInterfaceLock);
@@ -106,9 +106,8 @@
     camera3_device_t *device;
 
     ATRACE_BEGIN("camera3->open");
-    res = CameraService::filterOpenErrorCode(module->common.methods->open(
-        &module->common, deviceName.string(),
-        reinterpret_cast<hw_device_t**>(&device)));
+    res = CameraService::filterOpenErrorCode(module->open(
+            deviceName.string(), reinterpret_cast<hw_device_t**>(&device)));
     ATRACE_END();
 
     if (res != OK) {
@@ -127,7 +126,7 @@
     }
 
     camera_info info;
-    res = CameraService::filterGetInfoErrorCode(module->get_camera_info(
+    res = CameraService::filterGetInfoErrorCode(module->getCameraInfo(
         mId, &info));
     if (res != OK) return res;
 
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index ec8dc10..de10cfe 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -73,7 +73,7 @@
     virtual int      getId() const;
 
     // Transitions to idle state on success.
-    virtual status_t initialize(camera_module_t *module);
+    virtual status_t initialize(CameraModule *module);
     virtual status_t disconnect();
     virtual status_t dump(int fd, const Vector<String16> &args);
     virtual const CameraMetadata& info() const;
diff --git a/services/medialog/Android.mk b/services/medialog/Android.mk
index 95f2fef..03438bf 100644
--- a/services/medialog/Android.mk
+++ b/services/medialog/Android.mk
@@ -10,4 +10,6 @@
 
 LOCAL_32_BIT_ONLY := true
 
+LOCAL_C_INCLUDES := $(call include-path-for, audio-utils)
+
 include $(BUILD_SHARED_LIBRARY)
diff --git a/tools/resampler_tools/Android.mk b/tools/resampler_tools/Android.mk
index e8cbe39..b58e4cd 100644
--- a/tools/resampler_tools/Android.mk
+++ b/tools/resampler_tools/Android.mk
@@ -1,6 +1,6 @@
 # Copyright 2005 The Android Open Source Project
 #
-# Android.mk for resampler_tools 
+# Android.mk for resampler_tools
 #