commit 1c32a00b8750998562a8d76351ba5c6d2ed788c8
author Glenn Kasten <gkasten@google.com> Fri Mar 07 01:28:22 2014 +0000
committer Android (Google) Code Review <android-gerrit@google.com> Fri Mar 07 01:28:23 2014 +0000
tree 68ac5f1ccd1c53a7ba7da918eec2ea96c5918b94
parent 1bdf803fde294017d8b504891618f688a21a2e75
parent 96f04886d1c1bfbc422e2be033ea66be83e42441

Merge "Simplify AudioTrack stream end and fix race"

camera/CameraMetadata.cpp

diff --git a/camera/CameraMetadata.cpp b/camera/CameraMetadata.cpp
index f447c5b..6b726e0 100644
--- a/camera/CameraMetadata.cpp
+++ b/camera/CameraMetadata.cpp
@@ -25,6 +25,9 @@
 
 namespace android {
 
+#define ALIGN_TO(val, alignment) \
+    (((uintptr_t)(val) + ((alignment) - 1)) & ~((alignment) - 1))
+
 typedef Parcel::WritableBlob WritableBlob;
 typedef Parcel::ReadableBlob ReadableBlob;
 
@@ -133,11 +136,19 @@
 }
 
 status_t CameraMetadata::append(const CameraMetadata &other) {
+    return append(other.mBuffer);
+}
+
+status_t CameraMetadata::append(const camera_metadata_t* other) {
     if (mLocked) {
         ALOGE("%s: CameraMetadata is locked", __FUNCTION__);
         return INVALID_OPERATION;
     }
-    return append_camera_metadata(mBuffer, other.mBuffer);
+    size_t extraEntries = get_camera_metadata_entry_count(other);
+    size_t extraData = get_camera_metadata_data_count(other);
+    resizeIfNeeded(extraEntries, extraData);
+
+    return append_camera_metadata(mBuffer, other);
 }
 
 size_t CameraMetadata::entryCount() const {
@@ -423,40 +434,70 @@
         *out = NULL;
     }
 
-    // arg0 = metadataSize (int32)
-    int32_t metadataSizeTmp = -1;
-    if ((err = data.readInt32(&metadataSizeTmp)) != OK) {
+    // See CameraMetadata::writeToParcel for parcel data layout diagram and explanation.
+    // arg0 = blobSize (int32)
+    int32_t blobSizeTmp = -1;
+    if ((err = data.readInt32(&blobSizeTmp)) != OK) {
         ALOGE("%s: Failed to read metadata size (error %d %s)",
               __FUNCTION__, err, strerror(-err));
         return err;
     }
-    const size_t metadataSize = static_cast<size_t>(metadataSizeTmp);
+    const size_t blobSize = static_cast<size_t>(blobSizeTmp);
+    const size_t alignment = get_camera_metadata_alignment();
 
-    if (metadataSize == 0) {
+    // Special case: zero blob size means zero sized (NULL) metadata.
+    if (blobSize == 0) {
         ALOGV("%s: Read 0-sized metadata", __FUNCTION__);
         return OK;
     }
 
-    // NOTE: this doesn't make sense to me. shouldnt the blob
+    if (blobSize <= alignment) {
+        ALOGE("%s: metadata blob is malformed, blobSize(%zu) should be larger than alignment(%zu)",
+                __FUNCTION__, blobSize, alignment);
+        return BAD_VALUE;
+    }
+
+    const size_t metadataSize = blobSize - alignment;
+
+    // NOTE: this doesn't make sense to me. shouldn't the blob
     // know how big it is? why do we have to specify the size
     // to Parcel::readBlob ?
-
     ReadableBlob blob;
     // arg1 = metadata (blob)
     do {
-        if ((err = data.readBlob(metadataSize, &blob)) != OK) {
-            ALOGE("%s: Failed to read metadata blob (sized %d). Possible "
+        if ((err = data.readBlob(blobSize, &blob)) != OK) {
+            ALOGE("%s: Failed to read metadata blob (sized %zu). Possible "
                   " serialization bug. Error %d %s",
-                  __FUNCTION__, metadataSize, err, strerror(-err));
+                  __FUNCTION__, blobSize, err, strerror(-err));
             break;
         }
-        const camera_metadata_t* tmp =
-                       reinterpret_cast<const camera_metadata_t*>(blob.data());
 
+        // arg2 = offset (blob)
+        // Must be after blob since we don't know offset until after writeBlob.
+        int32_t offsetTmp;
+        if ((err = data.readInt32(&offsetTmp)) != OK) {
+            ALOGE("%s: Failed to read metadata offsetTmp (error %d %s)",
+                  __FUNCTION__, err, strerror(-err));
+            break;
+        }
+        const size_t offset = static_cast<size_t>(offsetTmp);
+        if (offset >= alignment) {
+            ALOGE("%s: metadata offset(%zu) should be less than alignment(%zu)",
+                    __FUNCTION__, blobSize, alignment);
+            err = BAD_VALUE;
+            break;
+        }
+
+        const uintptr_t metadataStart = reinterpret_cast<uintptr_t>(blob.data()) + offset;
+        const camera_metadata_t* tmp =
+                       reinterpret_cast<const camera_metadata_t*>(metadataStart);
+        ALOGV("%s: alignment is: %zu, metadata start: %p, offset: %zu",
+                __FUNCTION__, alignment, tmp, offset);
         metadata = allocate_copy_camera_metadata_checked(tmp, metadataSize);
         if (metadata == NULL) {
             // We consider that allocation only fails if the validation
             // also failed, therefore the readFromParcel was a failure.
+            ALOGE("%s: metadata allocation and copy failed", __FUNCTION__);
             err = BAD_VALUE;
         }
     } while(0);
@@ -477,38 +518,79 @@
                                        const camera_metadata_t* metadata) {
     status_t res = OK;
 
-    // arg0 = metadataSize (int32)
+    /**
+     * Below is the camera metadata parcel layout:
+     *
+     * |--------------------------------------------|
+     * |             arg0: blobSize                 |
+     * |              (length = 4)                  |
+     * |--------------------------------------------|<--Skip the rest if blobSize == 0.
+     * |                                            |
+     * |                                            |
+     * |              arg1: blob                    |
+     * | (length = variable, see arg1 layout below) |
+     * |                                            |
+     * |                                            |
+     * |--------------------------------------------|
+     * |              arg2: offset                  |
+     * |              (length = 4)                  |
+     * |--------------------------------------------|
+     */
 
+    // arg0 = blobSize (int32)
     if (metadata == NULL) {
+        // Write zero blobSize for null metadata.
         return data.writeInt32(0);
     }
 
+    /**
+     * Always make the blob size sufficiently larger, as we need put alignment
+     * padding and metadata into the blob. Since we don't know the alignment
+     * offset before writeBlob. Then write the metadata to aligned offset.
+     */
     const size_t metadataSize = get_camera_metadata_compact_size(metadata);
-    res = data.writeInt32(static_cast<int32_t>(metadataSize));
+    const size_t alignment = get_camera_metadata_alignment();
+    const size_t blobSize = metadataSize + alignment;
+    res = data.writeInt32(static_cast<int32_t>(blobSize));
     if (res != OK) {
         return res;
     }
 
-    // arg1 = metadata (blob)
+    size_t offset = 0;
+    /**
+     * arg1 = metadata (blob).
+     *
+     * The blob size is the sum of front padding size, metadata size and back padding
+     * size, which is equal to metadataSize + alignment.
+     *
+     * The blob layout is:
+     * |------------------------------------|<----Start address of the blob (unaligned).
+     * |           front padding            |
+     * |          (size = offset)           |
+     * |------------------------------------|<----Aligned start address of metadata.
+     * |                                    |
+     * |                                    |
+     * |            metadata                |
+     * |       (size = metadataSize)        |
+     * |                                    |
+     * |                                    |
+     * |------------------------------------|
+     * |           back padding             |
+     * |     (size = alignment - offset)    |
+     * |------------------------------------|<----End address of blob.
+     *                                            (Blob start address + blob size).
+     */
     WritableBlob blob;
     do {
-        res = data.writeBlob(metadataSize, &blob);
+        res = data.writeBlob(blobSize, &blob);
         if (res != OK) {
             break;
         }
-        copy_camera_metadata(blob.data(), metadataSize, metadata);
-
-        IF_ALOGV() {
-            if (validate_camera_metadata_structure(
-                        (const camera_metadata_t*)blob.data(),
-                        &metadataSize) != OK) {
-                ALOGV("%s: Failed to validate metadata %p after writing blob",
-                       __FUNCTION__, blob.data());
-            } else {
-                ALOGV("%s: Metadata written to blob. Validation success",
-                        __FUNCTION__);
-            }
-        }
+        const uintptr_t metadataStart = ALIGN_TO(blob.data(), alignment);
+        offset = metadataStart - reinterpret_cast<uintptr_t>(blob.data());
+        ALOGV("%s: alignment is: %zu, metadata start: %p, offset: %zu",
+                __FUNCTION__, alignment, metadataStart, offset);
+        copy_camera_metadata(reinterpret_cast<void*>(metadataStart), metadataSize, metadata);
 
         // Not too big of a problem since receiving side does hard validation
         // Don't check the size since the compact size could be larger
@@ -520,6 +602,9 @@
     } while(false);
     blob.release();
 
+    // arg2 = offset (int32)
+    res = data.writeInt32(static_cast<int32_t>(offset));
+
     return res;
 }
 

camera/CameraParameters.cpp

diff --git a/camera/CameraParameters.cpp b/camera/CameraParameters.cpp
index c51f265..af091f4 100644
--- a/camera/CameraParameters.cpp
+++ b/camera/CameraParameters.cpp
@@ -470,7 +470,7 @@
     const size_t SIZE = 256;
     char buffer[SIZE];
     String8 result;
-    snprintf(buffer, 255, "CameraParameters::dump: mMap.size = %d\n", mMap.size());
+    snprintf(buffer, 255, "CameraParameters::dump: mMap.size = %zu\n", mMap.size());
     result.append(buffer);
     for (size_t i = 0; i < mMap.size(); i++) {
         String8 k, v;

camera/IProCameraCallbacks.cpp

diff --git a/camera/IProCameraCallbacks.cpp b/camera/IProCameraCallbacks.cpp
index 0fdb85a..bd3d420 100644
--- a/camera/IProCameraCallbacks.cpp
+++ b/camera/IProCameraCallbacks.cpp
@@ -67,11 +67,11 @@
                            IBinder::FLAG_ONEWAY);
     }
 
-    void onResultReceived(int32_t frameId, camera_metadata* result) {
+    void onResultReceived(int32_t requestId, camera_metadata* result) {
         ALOGV("onResultReceived");
         Parcel data, reply;
         data.writeInterfaceToken(IProCameraCallbacks::getInterfaceDescriptor());
-        data.writeInt32(frameId);
+        data.writeInt32(requestId);
         CameraMetadata::writeToParcel(data, result);
         remote()->transact(RESULT_RECEIVED, data, &reply, IBinder::FLAG_ONEWAY);
     }
@@ -107,10 +107,10 @@
         case RESULT_RECEIVED: {
             ALOGV("RESULT_RECEIVED");
             CHECK_INTERFACE(IProCameraCallbacks, data, reply);
-            int32_t frameId = data.readInt32();
+            int32_t requestId = data.readInt32();
             camera_metadata_t *result = NULL;
             CameraMetadata::readFromParcel(data, &result);
-            onResultReceived(frameId, result);
+            onResultReceived(requestId, result);
             return NO_ERROR;
             break;
         }

camera/ProCamera.cpp

diff --git a/camera/ProCamera.cpp b/camera/ProCamera.cpp
index 577c760..ba5a48c 100644
--- a/camera/ProCamera.cpp
+++ b/camera/ProCamera.cpp
@@ -90,8 +90,8 @@
     }
 }
 
-void ProCamera::onResultReceived(int32_t frameId, camera_metadata* result) {
-    ALOGV("%s: frameId = %d, result = %p", __FUNCTION__, frameId, result);
+void ProCamera::onResultReceived(int32_t requestId, camera_metadata* result) {
+    ALOGV("%s: requestId = %d, result = %p", __FUNCTION__, requestId, result);
 
     sp<ProCameraListener> listener;
     {
@@ -112,7 +112,7 @@
     result = tmp.release();
 
     if (listener != NULL) {
-        listener->onResultReceived(frameId, result);
+        listener->onResultReceived(requestId, result);
     } else {
         free_camera_metadata(result);
     }

camera/camera2/ICameraDeviceCallbacks.cpp

diff --git a/camera/camera2/ICameraDeviceCallbacks.cpp b/camera/camera2/ICameraDeviceCallbacks.cpp
index 3cec1f4..613358a 100644
--- a/camera/camera2/ICameraDeviceCallbacks.cpp
+++ b/camera/camera2/ICameraDeviceCallbacks.cpp
@@ -32,7 +32,9 @@
 namespace android {
 
 enum {
-    NOTIFY_CALLBACK = IBinder::FIRST_CALL_TRANSACTION,
+    CAMERA_ERROR = IBinder::FIRST_CALL_TRANSACTION,
+    CAMERA_IDLE,
+    CAPTURE_STARTED,
     RESULT_RECEIVED,
 };
 
@@ -44,19 +46,37 @@
     {
     }
 
-    // generic callback from camera service to app
-    void notifyCallback(int32_t msgType, int32_t ext1, int32_t ext2)
+    void onDeviceError(CameraErrorCode errorCode)
     {
-        ALOGV("notifyCallback");
+        ALOGV("onDeviceError");
         Parcel data, reply;
         data.writeInterfaceToken(ICameraDeviceCallbacks::getInterfaceDescriptor());
-        data.writeInt32(msgType);
-        data.writeInt32(ext1);
-        data.writeInt32(ext2);
-        remote()->transact(NOTIFY_CALLBACK, data, &reply, IBinder::FLAG_ONEWAY);
+        data.writeInt32(static_cast<int32_t>(errorCode));
+        remote()->transact(CAMERA_ERROR, data, &reply, IBinder::FLAG_ONEWAY);
         data.writeNoException();
     }
 
+    void onDeviceIdle()
+    {
+        ALOGV("onDeviceIdle");
+        Parcel data, reply;
+        data.writeInterfaceToken(ICameraDeviceCallbacks::getInterfaceDescriptor());
+        remote()->transact(CAMERA_IDLE, data, &reply, IBinder::FLAG_ONEWAY);
+        data.writeNoException();
+    }
+
+    void onCaptureStarted(int32_t requestId, int64_t timestamp)
+    {
+        ALOGV("onCaptureStarted");
+        Parcel data, reply;
+        data.writeInterfaceToken(ICameraDeviceCallbacks::getInterfaceDescriptor());
+        data.writeInt32(requestId);
+        data.writeInt64(timestamp);
+        remote()->transact(CAPTURE_STARTED, data, &reply, IBinder::FLAG_ONEWAY);
+        data.writeNoException();
+    }
+
+
     void onResultReceived(int32_t requestId, const CameraMetadata& result) {
         ALOGV("onResultReceived");
         Parcel data, reply;
@@ -79,18 +99,33 @@
 {
     ALOGV("onTransact - code = %d", code);
     switch(code) {
-        case NOTIFY_CALLBACK: {
-            ALOGV("NOTIFY_CALLBACK");
+        case CAMERA_ERROR: {
+            ALOGV("onDeviceError");
             CHECK_INTERFACE(ICameraDeviceCallbacks, data, reply);
-            int32_t msgType = data.readInt32();
-            int32_t ext1 = data.readInt32();
-            int32_t ext2 = data.readInt32();
-            notifyCallback(msgType, ext1, ext2);
+            CameraErrorCode errorCode =
+                    static_cast<CameraErrorCode>(data.readInt32());
+            onDeviceError(errorCode);
+            data.readExceptionCode();
+            return NO_ERROR;
+        } break;
+        case CAMERA_IDLE: {
+            ALOGV("onDeviceIdle");
+            CHECK_INTERFACE(ICameraDeviceCallbacks, data, reply);
+            onDeviceIdle();
+            data.readExceptionCode();
+            return NO_ERROR;
+        } break;
+        case CAPTURE_STARTED: {
+            ALOGV("onCaptureStarted");
+            CHECK_INTERFACE(ICameraDeviceCallbacks, data, reply);
+            int32_t requestId = data.readInt32();
+            int64_t timestamp = data.readInt64();
+            onCaptureStarted(requestId, timestamp);
             data.readExceptionCode();
             return NO_ERROR;
         } break;
         case RESULT_RECEIVED: {
-            ALOGV("RESULT_RECEIVED");
+            ALOGV("onResultReceived");
             CHECK_INTERFACE(ICameraDeviceCallbacks, data, reply);
             int32_t requestId = data.readInt32();
             CameraMetadata result;
@@ -102,8 +137,7 @@
             onResultReceived(requestId, result);
             data.readExceptionCode();
             return NO_ERROR;
-            break;
-        }
+        } break;
         default:
             return BBinder::onTransact(code, data, reply, flags);
     }

camera/tests/ProCameraTests.cpp

diff --git a/camera/tests/ProCameraTests.cpp b/camera/tests/ProCameraTests.cpp
index e9aa99d..1f5867a 100644
--- a/camera/tests/ProCameraTests.cpp
+++ b/camera/tests/ProCameraTests.cpp
@@ -284,9 +284,9 @@
         }
     }
 
-    virtual void onResultReceived(int32_t frameId,
+    virtual void onResultReceived(int32_t requestId,
                                   camera_metadata* request) {
-        dout << "Result received frameId = " << frameId
+        dout << "Result received requestId = " << requestId
              << ", requestPtr = " << (void*)request << std::endl;
         QueueEvent(RESULT_RECEIVED);
         free_camera_metadata(request);
@@ -1276,4 +1276,3 @@
 }
 }
 }
-

cmds/screenrecord/Android.mk

diff --git a/cmds/screenrecord/Android.mk b/cmds/screenrecord/Android.mk
index b4a5947..6747e60 100644
--- a/cmds/screenrecord/Android.mk
+++ b/cmds/screenrecord/Android.mk
@@ -18,10 +18,15 @@
 
 LOCAL_SRC_FILES := \
 	screenrecord.cpp \
+	EglWindow.cpp \
+	FrameOutput.cpp \
+	TextRenderer.cpp \
+	Overlay.cpp \
+	Program.cpp
 
 LOCAL_SHARED_LIBRARIES := \
 	libstagefright libmedia libutils libbinder libstagefright_foundation \
-	libjpeg libgui libcutils liblog
+	libjpeg libgui libcutils liblog libEGL libGLESv2
 
 LOCAL_C_INCLUDES := \
 	frameworks/av/media/libstagefright \
@@ -30,6 +35,7 @@
 	external/jpeg
 
 LOCAL_CFLAGS += -Wno-multichar
+#LOCAL_CFLAGS += -UNDEBUG
 
 LOCAL_MODULE_TAGS := optional
 

cmds/screenrecord/EglWindow.cpp

diff --git a/cmds/screenrecord/EglWindow.cpp b/cmds/screenrecord/EglWindow.cpp
new file mode 100644
index 0000000..c16f2ad
--- /dev/null
+++ b/cmds/screenrecord/EglWindow.cpp
@@ -0,0 +1,190 @@
+/*
+ * Copyright 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "ScreenRecord"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#define EGL_EGLEXT_PROTOTYPES
+
+#include <gui/BufferQueue.h>
+#include <gui/GraphicBufferAlloc.h>
+#include <gui/Surface.h>
+
+#include "EglWindow.h"
+
+#include <EGL/egl.h>
+#include <EGL/eglext.h>
+
+#include <assert.h>
+
+using namespace android;
+
+
+status_t EglWindow::createWindow(const sp<IGraphicBufferProducer>& surface) {
+    if (mEglSurface != EGL_NO_SURFACE) {
+        ALOGE("surface already created");
+        return UNKNOWN_ERROR;
+    }
+    status_t err = eglSetupContext(false);
+    if (err != NO_ERROR) {
+        return err;
+    }
+
+    // Cache the current dimensions.  We're not expecting these to change.
+    surface->query(NATIVE_WINDOW_WIDTH, &mWidth);
+    surface->query(NATIVE_WINDOW_HEIGHT, &mHeight);
+
+    // Output side (EGL surface to draw on).
+    sp<ANativeWindow> anw = new Surface(surface);
+    mEglSurface = eglCreateWindowSurface(mEglDisplay, mEglConfig, anw.get(),
+            NULL);
+    if (mEglSurface == EGL_NO_SURFACE) {
+        ALOGE("eglCreateWindowSurface error: %#x", eglGetError());
+        eglRelease();
+        return UNKNOWN_ERROR;
+    }
+
+    return NO_ERROR;
+}
+
+status_t EglWindow::createPbuffer(int width, int height) {
+    if (mEglSurface != EGL_NO_SURFACE) {
+        ALOGE("surface already created");
+        return UNKNOWN_ERROR;
+    }
+    status_t err = eglSetupContext(true);
+    if (err != NO_ERROR) {
+        return err;
+    }
+
+    mWidth = width;
+    mHeight = height;
+
+    EGLint pbufferAttribs[] = {
+            EGL_WIDTH, width,
+            EGL_HEIGHT, height,
+            EGL_NONE
+    };
+    mEglSurface = eglCreatePbufferSurface(mEglDisplay, mEglConfig, pbufferAttribs);
+    if (mEglSurface == EGL_NO_SURFACE) {
+        ALOGE("eglCreatePbufferSurface error: %#x", eglGetError());
+        eglRelease();
+        return UNKNOWN_ERROR;
+    }
+
+    return NO_ERROR;
+}
+
+status_t EglWindow::makeCurrent() const {
+    if (!eglMakeCurrent(mEglDisplay, mEglSurface, mEglSurface, mEglContext)) {
+        ALOGE("eglMakeCurrent failed: %#x", eglGetError());
+        return UNKNOWN_ERROR;
+    }
+    return NO_ERROR;
+}
+
+status_t EglWindow::eglSetupContext(bool forPbuffer) {
+    EGLBoolean result;
+
+    mEglDisplay = eglGetDisplay(EGL_DEFAULT_DISPLAY);
+    if (mEglDisplay == EGL_NO_DISPLAY) {
+        ALOGE("eglGetDisplay failed: %#x", eglGetError());
+        return UNKNOWN_ERROR;
+    }
+
+    EGLint majorVersion, minorVersion;
+    result = eglInitialize(mEglDisplay, &majorVersion, &minorVersion);
+    if (result != EGL_TRUE) {
+        ALOGE("eglInitialize failed: %#x", eglGetError());
+        return UNKNOWN_ERROR;
+    }
+    ALOGV("Initialized EGL v%d.%d", majorVersion, minorVersion);
+
+    EGLint numConfigs = 0;
+    EGLint windowConfigAttribs[] = {
+            EGL_SURFACE_TYPE, EGL_WINDOW_BIT,
+            EGL_RENDERABLE_TYPE, EGL_OPENGL_ES2_BIT,
+            EGL_RECORDABLE_ANDROID, 1,
+            EGL_RED_SIZE, 8,
+            EGL_GREEN_SIZE, 8,
+            EGL_BLUE_SIZE, 8,
+            // no alpha
+            EGL_NONE
+    };
+    EGLint pbufferConfigAttribs[] = {
+            EGL_SURFACE_TYPE, EGL_PBUFFER_BIT,
+            EGL_RENDERABLE_TYPE, EGL_OPENGL_ES2_BIT,
+            EGL_RED_SIZE, 8,
+            EGL_GREEN_SIZE, 8,
+            EGL_BLUE_SIZE, 8,
+            EGL_ALPHA_SIZE, 8,
+            EGL_NONE
+    };
+    result = eglChooseConfig(mEglDisplay,
+            forPbuffer ? pbufferConfigAttribs : windowConfigAttribs,
+            &mEglConfig, 1, &numConfigs);
+    if (result != EGL_TRUE) {
+        ALOGE("eglChooseConfig error: %#x", eglGetError());
+        return UNKNOWN_ERROR;
+    }
+
+    EGLint contextAttribs[] = {
+        EGL_CONTEXT_CLIENT_VERSION, 2,
+        EGL_NONE
+    };
+    mEglContext = eglCreateContext(mEglDisplay, mEglConfig, EGL_NO_CONTEXT,
+            contextAttribs);
+    if (mEglContext == EGL_NO_CONTEXT) {
+        ALOGE("eglCreateContext error: %#x", eglGetError());
+        return UNKNOWN_ERROR;
+    }
+
+    return NO_ERROR;
+}
+
+void EglWindow::eglRelease() {
+    ALOGV("EglWindow::eglRelease");
+    if (mEglDisplay != EGL_NO_DISPLAY) {
+        eglMakeCurrent(mEglDisplay, EGL_NO_SURFACE, EGL_NO_SURFACE,
+                EGL_NO_CONTEXT);
+
+        if (mEglContext != EGL_NO_CONTEXT) {
+            eglDestroyContext(mEglDisplay, mEglContext);
+        }
+
+        if (mEglSurface != EGL_NO_SURFACE) {
+            eglDestroySurface(mEglDisplay, mEglSurface);
+        }
+    }
+
+    mEglDisplay = EGL_NO_DISPLAY;
+    mEglContext = EGL_NO_CONTEXT;
+    mEglSurface = EGL_NO_SURFACE;
+    mEglConfig = NULL;
+
+    eglReleaseThread();
+}
+
+// Sets the presentation time on the current EGL buffer.
+void EglWindow::presentationTime(nsecs_t whenNsec) const {
+    eglPresentationTimeANDROID(mEglDisplay, mEglSurface, whenNsec);
+}
+
+// Swaps the EGL buffer.
+void EglWindow::swapBuffers() const {
+    eglSwapBuffers(mEglDisplay, mEglSurface);
+}

cmds/screenrecord/EglWindow.h

diff --git a/cmds/screenrecord/EglWindow.h b/cmds/screenrecord/EglWindow.h
new file mode 100644
index 0000000..69d0c31
--- /dev/null
+++ b/cmds/screenrecord/EglWindow.h
@@ -0,0 +1,87 @@
+/*
+ * Copyright 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef SCREENRECORD_EGL_WINDOW_H
+#define SCREENRECORD_EGL_WINDOW_H
+
+#include <gui/BufferQueue.h>
+#include <utils/Errors.h>
+
+#include <EGL/egl.h>
+
+namespace android {
+
+/*
+ * Wraps EGL display, context, surface, config for a window surface.
+ *
+ * Not thread safe.
+ */
+class EglWindow {
+public:
+    EglWindow() :
+        mEglDisplay(EGL_NO_DISPLAY),
+        mEglContext(EGL_NO_CONTEXT),
+        mEglSurface(EGL_NO_SURFACE),
+        mEglConfig(NULL),
+        mWidth(0),
+        mHeight(0)
+        {}
+    ~EglWindow() { eglRelease(); }
+
+    // Creates an EGL window for the supplied surface.
+    status_t createWindow(const sp<IGraphicBufferProducer>& surface);
+
+    // Creates an EGL pbuffer surface.
+    status_t createPbuffer(int width, int height);
+
+    // Return width and height values (obtained from IGBP).
+    int getWidth() const { return mWidth; }
+    int getHeight() const { return mHeight; }
+
+    // Release anything we created.
+    void release() { eglRelease(); }
+
+    // Make this context current.
+    status_t makeCurrent() const;
+
+    // Sets the presentation time on the current EGL buffer.
+    void presentationTime(nsecs_t whenNsec) const;
+
+    // Swaps the EGL buffer.
+    void swapBuffers() const;
+
+private:
+    EglWindow(const EglWindow&);
+    EglWindow& operator=(const EglWindow&);
+
+    // Init display, create config and context.
+    status_t eglSetupContext(bool forPbuffer);
+    void eglRelease();
+
+    // Basic EGL goodies.
+    EGLDisplay mEglDisplay;
+    EGLContext mEglContext;
+    EGLSurface mEglSurface;
+    EGLConfig mEglConfig;
+
+    // Surface dimensions.
+    int mWidth;
+    int mHeight;
+};
+
+}; // namespace android
+
+#endif /*SCREENRECORD_EGL_WINDOW_H*/

cmds/screenrecord/FontBitmap.h

diff --git a/cmds/screenrecord/FontBitmap.h b/cmds/screenrecord/FontBitmap.h
new file mode 100644
index 0000000..2b94f35
--- /dev/null
+++ b/cmds/screenrecord/FontBitmap.h
@@ -0,0 +1,6571 @@
+// auto-generated from Android default system font at 24pts
+class FontBitmap {
+public:
+    static const uint32_t width = 256;
+    static const uint32_t height = 204;
+    static const uint32_t numGlyphs = 95;
+    static const uint32_t firstGlyphChar = 32;
+    static const uint32_t maxGlyphHeight = 34;
+    static const uint32_t outlineWidth = 1;
+    static const uint8_t pixels[];
+    static const uint16_t yoffset[];
+    static const uint16_t glyphWidth[];
+};
+const uint8_t FontBitmap::pixels[] = {
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
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+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0xa8, 0x73, 0xb5,
+    0xef, 0xff, 0xff, 0xd5, 0xfe, 0x26, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0xaa, 0x4d, 0xf7,
+    0xff, 0xff, 0x73, 0x50, 0x0, 0x0, 0x0, 0xfe,
+    0xfb, 0xff, 0x2f, 0x22, 0x0, 0x0, 0x68, 0x87,
+    0xff, 0xff, 0xef, 0x3b, 0x90, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0xe6, 0xc3, 0xff, 0xff, 0xcb, 0xea, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x72, 0x5f, 0xff, 0xff, 0xff, 0x49, 0x50,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0xe2, 0xc1, 0xff,
+    0xff, 0xff, 0xfb, 0x55, 0xb6, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x24, 0xe8, 0x41,
+    0xc7, 0xfd, 0x37, 0x50, 0x0, 0x0, 0x0, 0xee,
+    0x47, 0x49, 0xd, 0x12, 0x0, 0x0, 0x68, 0x4f,
+    0xfd, 0xbd, 0x35, 0xda, 0x18, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0xc6, 0xb1, 0xe9, 0xe9, 0xb7, 0xcc, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x52, 0x57, 0xe9, 0xe9, 0xe9, 0x41, 0x38,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0xd8, 0xd7, 0xff,
+    0xff, 0xdf, 0x63, 0xf2, 0x28, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x1c, 0x9e,
+    0xfa, 0x29, 0xf6, 0x16, 0x0, 0x0, 0x0, 0x1e,
+    0x3a, 0x3c, 0x22, 0x2, 0x0, 0x0, 0x24, 0xf4,
+    0x29, 0xfa, 0x92, 0x14, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x7c, 0xe2, 0xf8, 0xf8, 0xe4, 0x82, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x2a, 0xd0, 0xf4, 0xfc, 0xf4, 0xcc, 0x1c,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0xaa, 0x9, 0x2f,
+    0x17, 0xfe, 0xc6, 0x32, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0xe, 0xe, 0xe, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0xe,
+    0xe, 0xe, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x10, 0x16,
+    0x16, 0x6, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+    0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+};
+const uint16_t FontBitmap::glyphWidth[] = {
+    8, 9, 10, 16, 16, 20, 18, 6,
+    10, 10, 13, 15, 8, 11, 9, 11,
+    16, 16, 16, 16, 16, 16, 16, 16,
+    16, 16, 9, 9, 14, 16, 14, 14,
+    23, 17, 17, 17, 18, 16, 16, 18,
+    19, 9, 16, 18, 15, 23, 19, 19,
+    18, 19, 18, 17, 17, 18, 17, 23,
+    17, 17, 16, 9, 12, 9, 13, 13,
+    10, 15, 16, 14, 16, 15, 11, 16,
+    16, 8, 8, 15, 8, 23, 16, 16,
+    16, 16, 11, 14, 10, 16, 14, 20,
+    14, 14, 14, 10, 8, 10, 18,
+};
+const uint16_t FontBitmap::yoffset[] = {
+    0, 0, 0, 0, 0, 0, 0, 0,
+    0, 0, 0, 0, 0, 0, 0, 0,
+    0, 0, 0, 34, 34, 34, 34, 34,
+    34, 34, 34, 34, 34, 34, 34, 34,
+    34, 34, 68, 68, 68, 68, 68, 68,
+    68, 68, 68, 68, 68, 68, 68, 68,
+    102, 102, 102, 102, 102, 102, 102, 102,
+    102, 102, 102, 102, 102, 102, 102, 136,
+    136, 136, 136, 136, 136, 136, 136, 136,
+    136, 136, 136, 136, 136, 136, 136, 136,
+    170, 170, 170, 170, 170, 170, 170, 170,
+    170, 170, 170, 170, 170, 170, 170,
+};

cmds/screenrecord/FrameOutput.cpp

diff --git a/cmds/screenrecord/FrameOutput.cpp b/cmds/screenrecord/FrameOutput.cpp
new file mode 100644
index 0000000..b5cf2f9
--- /dev/null
+++ b/cmds/screenrecord/FrameOutput.cpp
@@ -0,0 +1,208 @@
+/*
+ * Copyright 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 "ScreenRecord"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include <GLES2/gl2.h>
+#include <GLES2/gl2ext.h>
+
+#include "FrameOutput.h"
+
+using namespace android;
+
+static const bool kShowTiming = false;      // set to "true" for debugging
+static const int kGlBytesPerPixel = 4;      // GL_RGBA
+static const int kOutBytesPerPixel = 3;     // RGB only
+
+inline void FrameOutput::setValueLE(uint8_t* buf, uint32_t value) {
+    // Since we're running on an Android device, we're (almost) guaranteed
+    // to be little-endian, and (almost) guaranteed that unaligned 32-bit
+    // writes will work without any performance penalty... but do it
+    // byte-by-byte anyway.
+    buf[0] = (uint8_t) value;
+    buf[1] = (uint8_t) (value >> 8);
+    buf[2] = (uint8_t) (value >> 16);
+    buf[3] = (uint8_t) (value >> 24);
+}
+
+status_t FrameOutput::createInputSurface(int width, int height,
+        sp<IGraphicBufferProducer>* pBufferProducer) {
+    status_t err;
+
+    err = mEglWindow.createPbuffer(width, height);
+    if (err != NO_ERROR) {
+        return err;
+    }
+    mEglWindow.makeCurrent();
+
+    glViewport(0, 0, width, height);
+    glDisable(GL_DEPTH_TEST);
+    glDisable(GL_CULL_FACE);
+
+    // Shader for rendering the external texture.
+    err = mExtTexProgram.setup(Program::PROGRAM_EXTERNAL_TEXTURE);
+    if (err != NO_ERROR) {
+        return err;
+    }
+
+    // Input side (buffers from virtual display).
+    glGenTextures(1, &mExtTextureName);
+    if (mExtTextureName == 0) {
+        ALOGE("glGenTextures failed: %#x", glGetError());
+        return UNKNOWN_ERROR;
+    }
+
+    mBufferQueue = new BufferQueue(/*new GraphicBufferAlloc()*/);
+    mGlConsumer = new GLConsumer(mBufferQueue, mExtTextureName,
+                GL_TEXTURE_EXTERNAL_OES);
+    mGlConsumer->setName(String8("virtual display"));
+    mGlConsumer->setDefaultBufferSize(width, height);
+    mGlConsumer->setDefaultMaxBufferCount(5);
+    mGlConsumer->setConsumerUsageBits(GRALLOC_USAGE_HW_TEXTURE);
+
+    mGlConsumer->setFrameAvailableListener(this);
+
+    mPixelBuf = new uint8_t[width * height * kGlBytesPerPixel];
+
+    *pBufferProducer = mBufferQueue;
+
+    ALOGD("FrameOutput::createInputSurface OK");
+    return NO_ERROR;
+}
+
+status_t FrameOutput::copyFrame(FILE* fp, long timeoutUsec) {
+    Mutex::Autolock _l(mMutex);
+    ALOGV("copyFrame %ld\n", timeoutUsec);
+
+    if (!mFrameAvailable) {
+        nsecs_t timeoutNsec = (nsecs_t)timeoutUsec * 1000;
+        int cc = mEventCond.waitRelative(mMutex, timeoutNsec);
+        if (cc == -ETIMEDOUT) {
+            ALOGV("cond wait timed out");
+            return ETIMEDOUT;
+        } else if (cc != 0) {
+            ALOGW("cond wait returned error %d", cc);
+            return cc;
+        }
+    }
+    if (!mFrameAvailable) {
+        // This happens when Ctrl-C is hit.  Apparently POSIX says that the
+        // pthread wait call doesn't return EINTR, treating this instead as
+        // an instance of a "spurious wakeup".  We didn't get a frame, so
+        // we just treat it as a timeout.
+        return ETIMEDOUT;
+    }
+
+    // A frame is available.  Clear the flag for the next round.
+    mFrameAvailable = false;
+
+    float texMatrix[16];
+    mGlConsumer->updateTexImage();
+    mGlConsumer->getTransformMatrix(texMatrix);
+
+    // The data is in an external texture, so we need to render it to the
+    // pbuffer to get access to RGB pixel data.  We also want to flip it
+    // upside-down for easy conversion to a bitmap.
+    int width = mEglWindow.getWidth();
+    int height = mEglWindow.getHeight();
+    status_t err = mExtTexProgram.blit(mExtTextureName, texMatrix, 0, 0,
+            width, height, true);
+    if (err != NO_ERROR) {
+        return err;
+    }
+
+    // GLES only guarantees that glReadPixels() will work with GL_RGBA, so we
+    // need to get 4 bytes/pixel and reduce it.  Depending on the size of the
+    // screen and the device capabilities, this can take a while.
+    int64_t startWhenNsec, pixWhenNsec, endWhenNsec;
+    if (kShowTiming) {
+        startWhenNsec = systemTime(CLOCK_MONOTONIC);
+    }
+    GLenum glErr;
+    glReadPixels(0, 0, width, height, GL_RGBA, GL_UNSIGNED_BYTE, mPixelBuf);
+    if ((glErr = glGetError()) != GL_NO_ERROR) {
+        ALOGE("glReadPixels failed: %#x", glErr);
+        return UNKNOWN_ERROR;
+    }
+    if (kShowTiming) {
+        pixWhenNsec = systemTime(CLOCK_MONOTONIC);
+    }
+    reduceRgbaToRgb(mPixelBuf, width * height);
+    if (kShowTiming) {
+        endWhenNsec = systemTime(CLOCK_MONOTONIC);
+        ALOGD("got pixels (get=%.3f ms, reduce=%.3fms)",
+                (pixWhenNsec - startWhenNsec) / 1000000.0,
+                (endWhenNsec - pixWhenNsec) / 1000000.0);
+    }
+
+    // Fill out the header.
+    size_t headerLen = sizeof(uint32_t) * 5;
+    size_t rgbDataLen = width * height * kOutBytesPerPixel;
+    size_t packetLen = headerLen - sizeof(uint32_t) + rgbDataLen;
+    uint8_t header[headerLen];
+    setValueLE(&header[0], packetLen);
+    setValueLE(&header[4], width);
+    setValueLE(&header[8], height);
+    setValueLE(&header[12], width * kOutBytesPerPixel);
+    setValueLE(&header[16], HAL_PIXEL_FORMAT_RGB_888);
+
+    // Currently using buffered I/O rather than writev().  Not expecting it
+    // to make much of a difference, but it might be worth a test for larger
+    // frame sizes.
+    if (kShowTiming) {
+        startWhenNsec = systemTime(CLOCK_MONOTONIC);
+    }
+    fwrite(header, 1, headerLen, fp);
+    fwrite(mPixelBuf, 1, rgbDataLen, fp);
+    fflush(fp);
+    if (kShowTiming) {
+        endWhenNsec = systemTime(CLOCK_MONOTONIC);
+        ALOGD("wrote pixels (%.3f ms)",
+                (endWhenNsec - startWhenNsec) / 1000000.0);
+    }
+
+    if (ferror(fp)) {
+        // errno may not be useful; log it anyway
+        ALOGE("write failed (errno=%d)", errno);
+        return UNKNOWN_ERROR;
+    }
+
+    return NO_ERROR;
+}
+
+void FrameOutput::reduceRgbaToRgb(uint8_t* buf, unsigned int pixelCount) {
+    // Convert RGBA to RGB.
+    //
+    // Unaligned 32-bit accesses are allowed on ARM, so we could do this
+    // with 32-bit copies advancing at different rates (taking care at the
+    // end to not go one byte over).
+    const uint8_t* readPtr = buf;
+    for (unsigned int i = 0; i < pixelCount; i++) {
+        *buf++ = *readPtr++;
+        *buf++ = *readPtr++;
+        *buf++ = *readPtr++;
+        readPtr++;
+    }
+}
+
+// Callback; executes on arbitrary thread.
+void FrameOutput::onFrameAvailable() {
+    Mutex::Autolock _l(mMutex);
+    mFrameAvailable = true;
+    mEventCond.signal();
+}

cmds/screenrecord/FrameOutput.h

diff --git a/cmds/screenrecord/FrameOutput.h b/cmds/screenrecord/FrameOutput.h
new file mode 100644
index 0000000..bb66e05
--- /dev/null
+++ b/cmds/screenrecord/FrameOutput.h
@@ -0,0 +1,103 @@
+/*
+ * Copyright 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 SCREENRECORD_FRAMEOUTPUT_H
+#define SCREENRECORD_FRAMEOUTPUT_H
+
+#include "Program.h"
+#include "EglWindow.h"
+
+#include <gui/BufferQueue.h>
+#include <gui/GLConsumer.h>
+
+namespace android {
+
+/*
+ * Support for "frames" output format.
+ */
+class FrameOutput : public GLConsumer::FrameAvailableListener {
+public:
+    FrameOutput() : mFrameAvailable(false),
+        mExtTextureName(0),
+        mPixelBuf(NULL)
+        {}
+
+    // Create an "input surface", similar in purpose to a MediaCodec input
+    // surface, that the virtual display can send buffers to.  Also configures
+    // EGL with a pbuffer surface on the current thread.
+    status_t createInputSurface(int width, int height,
+            sp<IGraphicBufferProducer>* pBufferProducer);
+
+    // Copy one from input to output.  If no frame is available, this will wait up to the
+    // specified number of microseconds.
+    //
+    // Returns ETIMEDOUT if the timeout expired before we found a frame.
+    status_t copyFrame(FILE* fp, long timeoutUsec);
+
+    // Prepare to copy frames.  Makes the EGL context used by this object current.
+    void prepareToCopy() {
+        mEglWindow.makeCurrent();
+    }
+
+private:
+    FrameOutput(const FrameOutput&);
+    FrameOutput& operator=(const FrameOutput&);
+
+    // Destruction via RefBase.
+    virtual ~FrameOutput() {
+        delete[] mPixelBuf;
+    }
+
+    // (overrides GLConsumer::FrameAvailableListener method)
+    virtual void onFrameAvailable();
+
+    // Reduces RGBA to RGB, in place.
+    static void reduceRgbaToRgb(uint8_t* buf, unsigned int pixelCount);
+
+    // Put a 32-bit value into a buffer, in little-endian byte order.
+    static void setValueLE(uint8_t* buf, uint32_t value);
+
+    // Used to wait for the FrameAvailableListener callback.
+    Mutex mMutex;
+    Condition mEventCond;
+
+    // Set by the FrameAvailableListener callback.
+    bool mFrameAvailable;
+
+    // Our queue.  The producer side is passed to the virtual display, the
+    // consumer side feeds into our GLConsumer.
+    sp<BufferQueue> mBufferQueue;
+
+    // This receives frames from the virtual display and makes them available
+    // as an external texture.
+    sp<GLConsumer> mGlConsumer;
+
+    // EGL display / context / surface.
+    EglWindow mEglWindow;
+
+    // GL rendering support.
+    Program mExtTexProgram;
+
+    // External texture, updated by GLConsumer.
+    GLuint mExtTextureName;
+
+    // Pixel data buffer.
+    uint8_t* mPixelBuf;
+};
+
+}; // namespace android
+
+#endif /*SCREENRECORD_FRAMEOUTPUT_H*/

cmds/screenrecord/Overlay.cpp

diff --git a/cmds/screenrecord/Overlay.cpp b/cmds/screenrecord/Overlay.cpp
new file mode 100644
index 0000000..2e98874
--- /dev/null
+++ b/cmds/screenrecord/Overlay.cpp
@@ -0,0 +1,401 @@
+/*
+ * Copyright 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "ScreenRecord"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include <gui/BufferQueue.h>
+#include <gui/GraphicBufferAlloc.h>
+#include <gui/Surface.h>
+#include <cutils/properties.h>
+#include <utils/misc.h>
+
+#include <GLES2/gl2.h>
+#include <GLES2/gl2ext.h>
+
+#include <stdlib.h>
+#include <assert.h>
+
+#include "screenrecord.h"
+#include "Overlay.h"
+#include "TextRenderer.h"
+
+using namespace android;
+
+// System properties to look up and display on the info screen.
+const char* Overlay::kPropertyNames[] = {
+        "ro.build.description",
+        // includes ro.build.id, ro.build.product, ro.build.tags, ro.build.type,
+        // and ro.build.version.release
+        "ro.product.manufacturer",
+        "ro.product.model",
+        "ro.board.platform",
+        "ro.revision",
+        "dalvik.vm.heapgrowthlimit",
+        "dalvik.vm.heapsize",
+        "persist.sys.dalvik.vm.lib.1",
+        //"ro.product.cpu.abi",
+        //"ro.bootloader",
+        //"this-never-appears!",
+};
+
+
+status_t Overlay::start(const sp<IGraphicBufferProducer>& outputSurface,
+        sp<IGraphicBufferProducer>* pBufferProducer) {
+    ALOGV("Overlay::start");
+    mOutputSurface = outputSurface;
+
+    // Grab the current monotonic time and the current wall-clock time so we
+    // can map one to the other.  This allows the overlay counter to advance
+    // by the exact delay between frames, but if the wall clock gets adjusted
+    // we won't track it, which means we'll gradually go out of sync with the
+    // times in logcat.
+    mStartMonotonicNsecs = systemTime(CLOCK_MONOTONIC);
+    mStartRealtimeNsecs = systemTime(CLOCK_REALTIME);
+
+    Mutex::Autolock _l(mMutex);
+
+    // Start the thread.  Traffic begins immediately.
+    run("overlay");
+
+    mState = INIT;
+    while (mState == INIT) {
+        mStartCond.wait(mMutex);
+    }
+
+    if (mThreadResult != NO_ERROR) {
+        ALOGE("Failed to start overlay thread: err=%d", mThreadResult);
+        return mThreadResult;
+    }
+    assert(mState == RUNNING);
+
+    ALOGV("Overlay::start successful");
+    *pBufferProducer = mBufferQueue;
+    return NO_ERROR;
+}
+
+status_t Overlay::stop() {
+    ALOGV("Overlay::stop");
+    Mutex::Autolock _l(mMutex);
+    mState = STOPPING;
+    mEventCond.signal();
+    return NO_ERROR;
+}
+
+bool Overlay::threadLoop() {
+    Mutex::Autolock _l(mMutex);
+
+    mThreadResult = setup_l();
+
+    if (mThreadResult != NO_ERROR) {
+        ALOGW("Aborting overlay thread");
+        mState = STOPPED;
+        release_l();
+        mStartCond.broadcast();
+        return false;
+    }
+
+    ALOGV("Overlay thread running");
+    mState = RUNNING;
+    mStartCond.broadcast();
+
+    while (mState == RUNNING) {
+        mEventCond.wait(mMutex);
+        if (mFrameAvailable) {
+            ALOGV("Awake, frame available");
+            processFrame_l();
+            mFrameAvailable = false;
+        } else {
+            ALOGV("Awake, frame not available");
+        }
+    }
+
+    ALOGV("Overlay thread stopping");
+    release_l();
+    mState = STOPPED;
+    return false;       // stop
+}
+
+status_t Overlay::setup_l() {
+    status_t err;
+
+    err = mEglWindow.createWindow(mOutputSurface);
+    if (err != NO_ERROR) {
+        return err;
+    }
+    mEglWindow.makeCurrent();
+
+    int width = mEglWindow.getWidth();
+    int height = mEglWindow.getHeight();
+
+    glViewport(0, 0, width, height);
+    glDisable(GL_DEPTH_TEST);
+    glDisable(GL_CULL_FACE);
+
+    // Shaders for rendering from different types of textures.
+    err = mTexProgram.setup(Program::PROGRAM_TEXTURE_2D);
+    if (err != NO_ERROR) {
+        return err;
+    }
+    err = mExtTexProgram.setup(Program::PROGRAM_EXTERNAL_TEXTURE);
+    if (err != NO_ERROR) {
+        return err;
+    }
+
+    err = mTextRenderer.loadIntoTexture();
+    if (err != NO_ERROR) {
+        return err;
+    }
+    mTextRenderer.setScreenSize(width, height);
+
+    // Input side (buffers from virtual display).
+    glGenTextures(1, &mExtTextureName);
+    if (mExtTextureName == 0) {
+        ALOGE("glGenTextures failed: %#x", glGetError());
+        return UNKNOWN_ERROR;
+    }
+
+    mBufferQueue = new BufferQueue(/*new GraphicBufferAlloc()*/);
+    mGlConsumer = new GLConsumer(mBufferQueue, mExtTextureName,
+                GL_TEXTURE_EXTERNAL_OES);
+    mGlConsumer->setName(String8("virtual display"));
+    mGlConsumer->setDefaultBufferSize(width, height);
+    mGlConsumer->setDefaultMaxBufferCount(5);
+    mGlConsumer->setConsumerUsageBits(GRALLOC_USAGE_HW_TEXTURE);
+
+    mGlConsumer->setFrameAvailableListener(this);
+
+    return NO_ERROR;
+}
+
+
+void Overlay::release_l() {
+    ALOGV("Overlay::release_l");
+    mOutputSurface.clear();
+    mGlConsumer.clear();
+    mBufferQueue.clear();
+
+    mTexProgram.release();
+    mExtTexProgram.release();
+    mEglWindow.release();
+}
+
+void Overlay::processFrame_l() {
+    float texMatrix[16];
+
+    mGlConsumer->updateTexImage();
+    mGlConsumer->getTransformMatrix(texMatrix);
+    nsecs_t monotonicNsec = mGlConsumer->getTimestamp();
+    nsecs_t frameNumber = mGlConsumer->getFrameNumber();
+    int64_t droppedFrames = 0;
+
+    if (mLastFrameNumber > 0) {
+        mTotalDroppedFrames += size_t(frameNumber - mLastFrameNumber) - 1;
+    }
+    mLastFrameNumber = frameNumber;
+
+    mTextRenderer.setProportionalScale(35);
+
+    if (false) {  // DEBUG - full blue background
+        glClearColor(0.0f, 0.0f, 1.0f, 1.0f);
+        glClear(GL_DEPTH_BUFFER_BIT | GL_COLOR_BUFFER_BIT);
+    }
+
+    int width = mEglWindow.getWidth();
+    int height = mEglWindow.getHeight();
+    if (false) {  // DEBUG - draw inset
+        mExtTexProgram.blit(mExtTextureName, texMatrix,
+                100, 100, width-200, height-200);
+    } else {
+        mExtTexProgram.blit(mExtTextureName, texMatrix,
+                0, 0, width, height);
+    }
+
+    glEnable(GL_BLEND);
+    glBlendFunc(GL_ONE, GL_ONE_MINUS_SRC_ALPHA);
+    if (false) {  // DEBUG - show entire font bitmap
+        mTexProgram.blit(mTextRenderer.getTextureName(), Program::kIdentity,
+                100, 100, width-200, height-200);
+    }
+
+    char textBuf[64];
+    getTimeString_l(monotonicNsec, textBuf, sizeof(textBuf));
+    String8 timeStr(String8::format("%s f=%lld (%zd)",
+            textBuf, frameNumber, mTotalDroppedFrames));
+    mTextRenderer.drawString(mTexProgram, Program::kIdentity, 0, 0, timeStr);
+
+    glDisable(GL_BLEND);
+
+    if (false) {  // DEBUG - add red rectangle in lower-left corner
+        glEnable(GL_SCISSOR_TEST);
+        glScissor(0, 0, 200, 200);
+        glClearColor(1.0f, 0.0f, 0.0f, 1.0f);
+        glClear(GL_COLOR_BUFFER_BIT);
+        glDisable(GL_SCISSOR_TEST);
+    }
+
+    mEglWindow.presentationTime(monotonicNsec);
+    mEglWindow.swapBuffers();
+}
+
+void Overlay::getTimeString_l(nsecs_t monotonicNsec, char* buf, size_t bufLen) {
+    //const char* format = "%m-%d %T";    // matches log output
+    const char* format = "%T";
+    struct tm tm;
+
+    // localtime/strftime is not the fastest way to do this, but a trivial
+    // benchmark suggests that the cost is negligible.
+    int64_t realTime = mStartRealtimeNsecs +
+            (monotonicNsec - mStartMonotonicNsecs);
+    time_t secs = (time_t) (realTime / 1000000000);
+    localtime_r(&secs, &tm);
+    strftime(buf, bufLen, format, &tm);
+
+    int32_t msec = (int32_t) ((realTime % 1000000000) / 1000000);
+    char tmpBuf[5];
+    snprintf(tmpBuf, sizeof(tmpBuf), ".%03d", msec);
+    strlcat(buf, tmpBuf, bufLen);
+}
+
+// Callback; executes on arbitrary thread.
+void Overlay::onFrameAvailable() {
+    ALOGV("Overlay::onFrameAvailable");
+    Mutex::Autolock _l(mMutex);
+    mFrameAvailable = true;
+    mEventCond.signal();
+}
+
+
+/*static*/ status_t Overlay::drawInfoPage(
+        const sp<IGraphicBufferProducer>& outputSurface) {
+    status_t err;
+
+    EglWindow window;
+    err = window.createWindow(outputSurface);
+    if (err != NO_ERROR) {
+        return err;
+    }
+    window.makeCurrent();
+
+    int width = window.getWidth();
+    int height = window.getHeight();
+    glViewport(0, 0, width, height);
+    glDisable(GL_DEPTH_TEST);
+    glDisable(GL_CULL_FACE);
+
+    // Shaders for rendering.
+    Program texProgram;
+    err = texProgram.setup(Program::PROGRAM_TEXTURE_2D);
+    if (err != NO_ERROR) {
+        return err;
+    }
+    TextRenderer textRenderer;
+    err = textRenderer.loadIntoTexture();
+    if (err != NO_ERROR) {
+        return err;
+    }
+    textRenderer.setScreenSize(width, height);
+
+    doDrawInfoPage(window, texProgram, textRenderer);
+
+    // Destroy the surface.  This causes a disconnect.
+    texProgram.release();
+    window.release();
+
+    return NO_ERROR;
+}
+
+/*static*/ void Overlay::doDrawInfoPage(const EglWindow& window,
+        const Program& texProgram, TextRenderer& textRenderer) {
+    const nsecs_t holdTime = 250000000LL;
+
+    glClearColor(0.0f, 0.0f, 0.0f, 1.0f);
+    glClear(GL_COLOR_BUFFER_BIT);
+
+    int width = window.getWidth();
+    int height = window.getHeight();
+
+    // Draw a thin border around the screen.  Some players, e.g. browser
+    // plugins, make it hard to see where the edges are when the device
+    // is using a black background, so this gives the viewer a frame of
+    // reference.
+    //
+    // This is a clumsy way to do it, but we're only doing it for one frame,
+    // and it's easier than actually drawing lines.
+    const int lineWidth = 4;
+    glEnable(GL_SCISSOR_TEST);
+    glClearColor(0.5f, 0.5f, 0.5f, 1.0f);
+    glScissor(0, 0, width, lineWidth);
+    glClear(GL_COLOR_BUFFER_BIT);
+    glScissor(0, height - lineWidth, width, lineWidth);
+    glClear(GL_COLOR_BUFFER_BIT);
+    glScissor(0, 0, lineWidth, height);
+    glClear(GL_COLOR_BUFFER_BIT);
+    glScissor(width - lineWidth, 0, lineWidth, height);
+    glClear(GL_COLOR_BUFFER_BIT);
+    glDisable(GL_SCISSOR_TEST);
+
+    //glEnable(GL_BLEND);
+    //glBlendFunc(GL_ONE, GL_ONE_MINUS_SRC_ALPHA);
+    textRenderer.setProportionalScale(30);
+
+    float xpos = 0;
+    float ypos = 0;
+    ypos = textRenderer.drawWrappedString(texProgram, xpos, ypos,
+            String8::format("Android screenrecord v%d.%d",
+                    kVersionMajor, kVersionMinor));
+
+    // Show date/time
+    time_t now = time(0);
+    struct tm tm;
+    localtime_r(&now, &tm);
+    char timeBuf[64];
+    strftime(timeBuf, sizeof(timeBuf), "%a, %d %b %Y %T %z", &tm);
+    String8 header("Started ");
+    header += timeBuf;
+    ypos = textRenderer.drawWrappedString(texProgram, xpos, ypos, header);
+    ypos += 8 * textRenderer.getScale();    // slight padding
+
+    // Show selected system property values
+    for (int i = 0; i < NELEM(kPropertyNames); i++) {
+        char valueBuf[PROPERTY_VALUE_MAX];
+
+        property_get(kPropertyNames[i], valueBuf, "");
+        if (valueBuf[0] == '\0') {
+            continue;
+        }
+        String8 str(String8::format("%s: [%s]", kPropertyNames[i], valueBuf));
+        ypos = textRenderer.drawWrappedString(texProgram, xpos, ypos, str);
+    }
+    ypos += 8 * textRenderer.getScale();    // slight padding
+
+    // Show GL info
+    String8 glStr("OpenGL: ");
+    glStr += (char*) glGetString(GL_VENDOR);
+    glStr += " / ";
+    glStr += (char*) glGetString(GL_RENDERER);
+    glStr += ", ";
+    glStr += (char*) glGetString(GL_VERSION);
+    ypos = textRenderer.drawWrappedString(texProgram, xpos, ypos, glStr);
+
+    //glDisable(GL_BLEND);
+
+    // Set a presentation time slightly in the past.  This will cause the
+    // player to hold the frame on screen.
+    window.presentationTime(systemTime(CLOCK_MONOTONIC) - holdTime);
+    window.swapBuffers();
+}

cmds/screenrecord/Overlay.h

diff --git a/cmds/screenrecord/Overlay.h b/cmds/screenrecord/Overlay.h
new file mode 100644
index 0000000..48e48e0
--- /dev/null
+++ b/cmds/screenrecord/Overlay.h
@@ -0,0 +1,159 @@
+/*
+ * Copyright 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef SCREENRECORD_OVERLAY_H
+#define SCREENRECORD_OVERLAY_H
+
+#include "Program.h"
+#include "TextRenderer.h"
+#include "EglWindow.h"
+
+#include <gui/BufferQueue.h>
+#include <gui/GLConsumer.h>
+#include <utils/Thread.h>
+
+#include <EGL/egl.h>
+
+namespace android {
+
+/*
+ * Overlay "filter".  This sits between the virtual display and the video
+ * encoder.
+ *
+ * Most functions run on a thread created by start().
+ */
+class Overlay : public GLConsumer::FrameAvailableListener, Thread {
+public:
+    Overlay() : Thread(false),
+        mThreadResult(UNKNOWN_ERROR),
+        mState(UNINITIALIZED),
+        mFrameAvailable(false),
+        mExtTextureName(0),
+        mStartMonotonicNsecs(0),
+        mStartRealtimeNsecs(0),
+        mLastFrameNumber(-1),
+        mTotalDroppedFrames(0)
+        {}
+
+    // Creates a thread that performs the overlay.  Pass in the surface that
+    // output will be sent to.
+    //
+    // This creates a dedicated thread for processing frames.
+    //
+    // Returns a reference to the producer side of a new BufferQueue that will
+    // be used by the virtual display.
+    status_t start(const sp<IGraphicBufferProducer>& outputSurface,
+            sp<IGraphicBufferProducer>* pBufferProducer);
+
+    // Stops the thread and releases resources.  It's okay to call this even
+    // if start() was never called.
+    status_t stop();
+
+    // This creates an EGL context and window surface, draws some informative
+    // text on it, swaps the buffer, and then tears the whole thing down.
+    static status_t drawInfoPage(const sp<IGraphicBufferProducer>& outputSurface);
+
+private:
+    Overlay(const Overlay&);
+    Overlay& operator=(const Overlay&);
+
+    // Destruction via RefBase.
+    virtual ~Overlay() { assert(mState == UNINITIALIZED || mState == STOPPED); }
+
+    // Draw the initial info screen.
+    static void doDrawInfoPage(const EglWindow& window,
+            const Program& texRender, TextRenderer& textRenderer);
+
+    // (overrides GLConsumer::FrameAvailableListener method)
+    virtual void onFrameAvailable();
+
+    // (overrides Thread method)
+    virtual bool threadLoop();
+
+    // One-time setup (essentially object construction on the overlay thread).
+    status_t setup_l();
+
+    // Release all resources held.
+    void release_l();
+
+    // Release EGL display, context, surface.
+    void eglRelease_l();
+
+    // Process a frame received from the virtual display.
+    void processFrame_l();
+
+    // Convert a monotonic time stamp into a string with the current time.
+    void getTimeString_l(nsecs_t monotonicNsec, char* buf, size_t bufLen);
+
+    // Guards all fields below.
+    Mutex mMutex;
+
+    // Initialization gate.
+    Condition mStartCond;
+
+    // Thread status, mostly useful during startup.
+    status_t mThreadResult;
+
+    // Overlay thread state.  States advance from left to right; object may
+    // not be restarted.
+    enum { UNINITIALIZED, INIT, RUNNING, STOPPING, STOPPED } mState;
+
+    // Event notification.  Overlay thread sleeps on this until a frame
+    // arrives or it's time to shut down.
+    Condition mEventCond;
+
+    // Set by the FrameAvailableListener callback.
+    bool mFrameAvailable;
+
+    // The surface we send our output to, i.e. the video encoder's input
+    // surface.
+    sp<IGraphicBufferProducer> mOutputSurface;
+
+    // Our queue.  The producer side is passed to the virtual display, the
+    // consumer side feeds into our GLConsumer.
+    sp<BufferQueue> mBufferQueue;
+
+    // This receives frames from the virtual display and makes them available
+    // as an external texture.
+    sp<GLConsumer> mGlConsumer;
+
+    // EGL display / context / surface.
+    EglWindow mEglWindow;
+
+    // GL rendering support.
+    Program mExtTexProgram;
+    Program mTexProgram;
+
+    // Text rendering.
+    TextRenderer mTextRenderer;
+
+    // External texture, updated by GLConsumer.
+    GLuint mExtTextureName;
+
+    // Start time, used to map monotonic to wall-clock time.
+    nsecs_t mStartMonotonicNsecs;
+    nsecs_t mStartRealtimeNsecs;
+
+    // Used for tracking dropped frames.
+    nsecs_t mLastFrameNumber;
+    size_t mTotalDroppedFrames;
+
+    static const char* kPropertyNames[];
+};
+
+}; // namespace android
+
+#endif /*SCREENRECORD_OVERLAY_H*/

cmds/screenrecord/Program.cpp

diff --git a/cmds/screenrecord/Program.cpp b/cmds/screenrecord/Program.cpp
new file mode 100644
index 0000000..73cae6e
--- /dev/null
+++ b/cmds/screenrecord/Program.cpp
@@ -0,0 +1,307 @@
+/*
+ * Copyright 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "ScreenRecord"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include "Program.h"
+
+#include <GLES2/gl2.h>
+#include <GLES2/gl2ext.h>
+
+#include <assert.h>
+
+using namespace android;
+
+// 4x4 identity matrix
+const float Program::kIdentity[] = {
+        1.0f, 0.0f, 0.0f, 0.0f,
+        0.0f, 1.0f, 0.0f, 0.0f,
+        0.0f, 0.0f, 1.0f, 0.0f,
+        0.0f, 0.0f, 0.0f, 1.0f
+};
+
+// Simple vertex shader.  Texture coord calc includes matrix for GLConsumer
+// transform.
+static const char* kVertexShader =
+        "uniform mat4 uMVPMatrix;\n"
+        "uniform mat4 uGLCMatrix;\n"
+        "attribute vec4 aPosition;\n"
+        "attribute vec4 aTextureCoord;\n"
+        "varying vec2 vTextureCoord;\n"
+        "void main() {\n"
+        "    gl_Position = uMVPMatrix * aPosition;\n"
+        "    vTextureCoord = (uGLCMatrix * aTextureCoord).xy;\n"
+        "}\n";
+
+// Trivial fragment shader for external texture.
+static const char* kExtFragmentShader =
+        "#extension GL_OES_EGL_image_external : require\n"
+        "precision mediump float;\n"
+        "varying vec2 vTextureCoord;\n"
+        "uniform samplerExternalOES uTexture;\n"
+        "void main() {\n"
+        "    gl_FragColor = texture2D(uTexture, vTextureCoord);\n"
+        "}\n";
+
+// Trivial fragment shader for mundane texture.
+static const char* kFragmentShader =
+        "precision mediump float;\n"
+        "varying vec2 vTextureCoord;\n"
+        "uniform sampler2D uTexture;\n"
+        "void main() {\n"
+        "    gl_FragColor = texture2D(uTexture, vTextureCoord);\n"
+        //"    gl_FragColor = vec4(0.2, 1.0, 0.2, 1.0);\n"
+        "}\n";
+
+status_t Program::setup(ProgramType type) {
+    ALOGV("Program::setup type=%d", type);
+    status_t err;
+
+    mProgramType = type;
+
+    GLuint program;
+    if (type == PROGRAM_TEXTURE_2D) {
+        err = createProgram(&program, kVertexShader, kFragmentShader);
+    } else {
+        err = createProgram(&program, kVertexShader, kExtFragmentShader);
+    }
+    if (err != NO_ERROR) {
+        return err;
+    }
+    assert(program != 0);
+
+    maPositionLoc = glGetAttribLocation(program, "aPosition");
+    maTextureCoordLoc = glGetAttribLocation(program, "aTextureCoord");
+    muMVPMatrixLoc = glGetUniformLocation(program, "uMVPMatrix");
+    muGLCMatrixLoc = glGetUniformLocation(program, "uGLCMatrix");
+    muTextureLoc = glGetUniformLocation(program, "uTexture");
+    if ((maPositionLoc | maTextureCoordLoc | muMVPMatrixLoc |
+            muGLCMatrixLoc | muTextureLoc) == -1) {
+        ALOGE("Attrib/uniform lookup failed: %#x", glGetError());
+        glDeleteProgram(program);
+        return UNKNOWN_ERROR;
+    }
+
+    mProgram = program;
+    return NO_ERROR;
+}
+
+void Program::release() {
+    ALOGV("Program::release");
+    if (mProgram != 0) {
+        glDeleteProgram(mProgram);
+        mProgram = 0;
+    }
+}
+
+status_t Program::createProgram(GLuint* outPgm, const char* vertexShader,
+        const char* fragmentShader) {
+    GLuint vs, fs;
+    status_t err;
+
+    err = compileShader(GL_VERTEX_SHADER, vertexShader, &vs);
+    if (err != NO_ERROR) {
+        return err;
+    }
+    err = compileShader(GL_FRAGMENT_SHADER, fragmentShader, &fs);
+    if (err != NO_ERROR) {
+        glDeleteShader(vs);
+        return err;
+    }
+
+    GLuint program;
+    err = linkShaderProgram(vs, fs, &program);
+    glDeleteShader(vs);
+    glDeleteShader(fs);
+    if (err == NO_ERROR) {
+        *outPgm = program;
+    }
+    return err;
+}
+
+status_t Program::compileShader(GLenum shaderType, const char* src,
+        GLuint* outShader) {
+    GLuint shader = glCreateShader(shaderType);
+    if (shader == 0) {
+        ALOGE("glCreateShader error: %#x", glGetError());
+        return UNKNOWN_ERROR;
+    }
+
+    glShaderSource(shader, 1, &src, NULL);
+    glCompileShader(shader);
+
+    GLint compiled = 0;
+    glGetShaderiv(shader, GL_COMPILE_STATUS, &compiled);
+    if (!compiled) {
+        ALOGE("Compile of shader type %d failed", shaderType);
+        GLint infoLen = 0;
+        glGetShaderiv(shader, GL_INFO_LOG_LENGTH, &infoLen);
+        if (infoLen) {
+            char* buf = new char[infoLen];
+            if (buf) {
+                glGetShaderInfoLog(shader, infoLen, NULL, buf);
+                ALOGE("Compile log: %s", buf);
+                delete[] buf;
+            }
+        }
+        glDeleteShader(shader);
+        return UNKNOWN_ERROR;
+    }
+    *outShader = shader;
+    return NO_ERROR;
+}
+
+status_t Program::linkShaderProgram(GLuint vs, GLuint fs, GLuint* outPgm) {
+    GLuint program = glCreateProgram();
+    if (program == 0) {
+        ALOGE("glCreateProgram error: %#x", glGetError());
+        return UNKNOWN_ERROR;
+    }
+
+    glAttachShader(program, vs);
+    glAttachShader(program, fs);
+    glLinkProgram(program);
+    GLint linkStatus = GL_FALSE;
+    glGetProgramiv(program, GL_LINK_STATUS, &linkStatus);
+    if (linkStatus != GL_TRUE) {
+        ALOGE("glLinkProgram failed");
+        GLint bufLength = 0;
+        glGetProgramiv(program, GL_INFO_LOG_LENGTH, &bufLength);
+        if (bufLength) {
+            char* buf = new char[bufLength];
+            if (buf) {
+                glGetProgramInfoLog(program, bufLength, NULL, buf);
+                ALOGE("Link log: %s", buf);
+                delete[] buf;
+            }
+        }
+        glDeleteProgram(program);
+        return UNKNOWN_ERROR;
+    }
+
+    *outPgm = program;
+    return NO_ERROR;
+}
+
+
+
+status_t Program::blit(GLuint texName, const float* texMatrix,
+        int32_t x, int32_t y, int32_t w, int32_t h, bool invert) const {
+    ALOGV("Program::blit %d xy=%d,%d wh=%d,%d", texName, x, y, w, h);
+
+    const float pos[] = {
+        float(x),   float(y+h),
+        float(x+w), float(y+h),
+        float(x),   float(y),
+        float(x+w), float(y),
+    };
+    const float uv[] = {
+        0.0f, 0.0f,
+        1.0f, 0.0f,
+        0.0f, 1.0f,
+        1.0f, 1.0f,
+    };
+    status_t err;
+
+    err = beforeDraw(texName, texMatrix, pos, uv, invert);
+    if (err == NO_ERROR) {
+        glDrawArrays(GL_TRIANGLE_STRIP, 0, 4);
+        err = afterDraw();
+    }
+    return err;
+}
+
+status_t Program::drawTriangles(GLuint texName, const float* texMatrix,
+        const float* vertices, const float* texes, size_t count) const {
+    ALOGV("Program::drawTriangles texName=%d", texName);
+
+    status_t err;
+
+    err = beforeDraw(texName, texMatrix, vertices, texes, false);
+    if (err == NO_ERROR) {
+        glDrawArrays(GL_TRIANGLES, 0, count);
+        err = afterDraw();
+    }
+    return err;
+}
+
+status_t Program::beforeDraw(GLuint texName, const float* texMatrix,
+        const float* vertices, const float* texes, bool invert) const {
+    // Create an orthographic projection matrix based on viewport size.
+    GLint vp[4];
+    glGetIntegerv(GL_VIEWPORT, vp);
+    float screenToNdc[16] = {
+        2.0f/float(vp[2]),  0.0f,               0.0f,   0.0f,
+        0.0f,               -2.0f/float(vp[3]), 0.0f,   0.0f,
+        0.0f,               0.0f,               1.0f,   0.0f,
+        -1.0f,              1.0f,               0.0f,   1.0f,
+    };
+    if (invert) {
+        screenToNdc[5] = -screenToNdc[5];
+        screenToNdc[13] = -screenToNdc[13];
+    }
+
+    glUseProgram(mProgram);
+
+    glVertexAttribPointer(maPositionLoc, 2, GL_FLOAT, GL_FALSE, 0, vertices);
+    glVertexAttribPointer(maTextureCoordLoc, 2, GL_FLOAT, GL_FALSE, 0, texes);
+    glEnableVertexAttribArray(maPositionLoc);
+    glEnableVertexAttribArray(maTextureCoordLoc);
+
+    glUniformMatrix4fv(muMVPMatrixLoc, 1, GL_FALSE, screenToNdc);
+    glUniformMatrix4fv(muGLCMatrixLoc, 1, GL_FALSE, texMatrix);
+
+    glActiveTexture(GL_TEXTURE0);
+
+    switch (mProgramType) {
+    case PROGRAM_EXTERNAL_TEXTURE:
+        glBindTexture(GL_TEXTURE_EXTERNAL_OES, texName);
+        break;
+    case PROGRAM_TEXTURE_2D:
+        glBindTexture(GL_TEXTURE_2D, texName);
+        break;
+    default:
+        ALOGE("unexpected program type %d", mProgramType);
+        return UNKNOWN_ERROR;
+    }
+
+    glUniform1i(muTextureLoc, 0);
+
+    GLenum glErr;
+    if ((glErr = glGetError()) != GL_NO_ERROR) {
+        ALOGE("GL error before draw: %#x", glErr);
+        glDisableVertexAttribArray(maPositionLoc);
+        glDisableVertexAttribArray(maTextureCoordLoc);
+        return UNKNOWN_ERROR;
+    }
+
+    return NO_ERROR;
+}
+
+status_t Program::afterDraw() const {
+    glDisableVertexAttribArray(maPositionLoc);
+    glDisableVertexAttribArray(maTextureCoordLoc);
+
+    GLenum glErr;
+    if ((glErr = glGetError()) != GL_NO_ERROR) {
+        ALOGE("GL error after draw: %#x", glErr);
+        return UNKNOWN_ERROR;
+    }
+
+    return NO_ERROR;
+}

cmds/screenrecord/Program.h

diff --git a/cmds/screenrecord/Program.h b/cmds/screenrecord/Program.h
new file mode 100644
index 0000000..558be8d
--- /dev/null
+++ b/cmds/screenrecord/Program.h
@@ -0,0 +1,94 @@
+/*
+ * Copyright 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef SCREENRECORD_PROGRAM_H
+#define SCREENRECORD_PROGRAM_H
+
+#include <utils/Errors.h>
+
+#include <EGL/egl.h>
+#include <GLES2/gl2.h>
+
+namespace android {
+
+/*
+ * Utility class for GLES rendering.
+ *
+ * Not thread-safe.
+ */
+class Program {
+public:
+    enum ProgramType { PROGRAM_UNKNOWN=0, PROGRAM_EXTERNAL_TEXTURE,
+            PROGRAM_TEXTURE_2D };
+
+    Program() :
+        mProgramType(PROGRAM_UNKNOWN),
+        mProgram(0),
+        maPositionLoc(0),
+        maTextureCoordLoc(0),
+        muMVPMatrixLoc(0),
+        muGLCMatrixLoc(0),
+        muTextureLoc(0)
+        {}
+    ~Program() { release(); }
+
+    // Initialize the program for use with the specified texture type.
+    status_t setup(ProgramType type);
+
+    // Release the program and associated resources.
+    void release();
+
+    // Blit the specified texture to { x, y, x+w, y+h }.  Inverts the
+    // content if "invert" is set.
+    status_t blit(GLuint texName, const float* texMatrix,
+            int32_t x, int32_t y, int32_t w, int32_t h,
+            bool invert = false) const;
+
+    // Draw a number of triangles.
+    status_t drawTriangles(GLuint texName, const float* texMatrix,
+            const float* vertices, const float* texes, size_t count) const;
+
+    static const float kIdentity[];
+
+private:
+    Program(const Program&);
+    Program& operator=(const Program&);
+
+    // Common code for draw functions.
+    status_t beforeDraw(GLuint texName, const float* texMatrix,
+            const float* vertices, const float* texes, bool invert) const;
+    status_t afterDraw() const;
+
+    // GLES 2 shader utilities.
+    status_t createProgram(GLuint* outPgm, const char* vertexShader,
+            const char* fragmentShader);
+    static status_t compileShader(GLenum shaderType, const char* src,
+            GLuint* outShader);
+    static status_t linkShaderProgram(GLuint vs, GLuint fs, GLuint* outPgm);
+
+    ProgramType mProgramType;
+    GLuint mProgram;
+
+    GLint maPositionLoc;
+    GLint maTextureCoordLoc;
+    GLint muMVPMatrixLoc;
+    GLint muGLCMatrixLoc;
+    GLint muTextureLoc;
+};
+
+}; // namespace android
+
+#endif /*SCREENRECORD_PROGRAM_H*/

cmds/screenrecord/TextRenderer.cpp

diff --git a/cmds/screenrecord/TextRenderer.cpp b/cmds/screenrecord/TextRenderer.cpp
new file mode 100644
index 0000000..784055c
--- /dev/null
+++ b/cmds/screenrecord/TextRenderer.cpp
@@ -0,0 +1,358 @@
+/*
+ * Copyright 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "ScreenRecord"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include "TextRenderer.h"
+
+#include <assert.h>
+
+namespace android {
+#include "FontBitmap.h"
+};
+
+using namespace android;
+
+const char TextRenderer::kWhitespace[] = " \t\n\r";
+
+bool TextRenderer::mInitialized = false;
+uint32_t TextRenderer::mXOffset[FontBitmap::numGlyphs];
+
+void TextRenderer::initOnce() {
+    if (!mInitialized) {
+        initXOffset();
+        mInitialized = true;
+    }
+}
+
+void TextRenderer::initXOffset() {
+    // Generate a table of X offsets.  They start at zero and reset whenever
+    // we move down a line (i.e. the Y offset changes).  The offset increases
+    // by one pixel more than the width because the generator left a gap to
+    // avoid reading pixels from adjacent glyphs in the texture filter.
+    uint16_t offset = 0;
+    uint16_t prevYOffset = (int16_t) -1;
+    for (unsigned int i = 0; i < FontBitmap::numGlyphs; i++) {
+        if (prevYOffset != FontBitmap::yoffset[i]) {
+            prevYOffset = FontBitmap::yoffset[i];
+            offset = 0;
+        }
+        mXOffset[i] = offset;
+        offset += FontBitmap::glyphWidth[i] + 1;
+    }
+}
+
+static bool isPowerOfTwo(uint32_t val) {
+    // a/k/a "is exactly one bit set"; note returns true for 0
+    return (val & (val -1)) == 0;
+}
+
+static uint32_t powerOfTwoCeil(uint32_t val) {
+    // drop it, smear the bits across, pop it
+    val--;
+    val |= val >> 1;
+    val |= val >> 2;
+    val |= val >> 4;
+    val |= val >> 8;
+    val |= val >> 16;
+    val++;
+
+    return val;
+}
+
+float TextRenderer::getGlyphHeight() const {
+    return FontBitmap::maxGlyphHeight;
+}
+
+status_t TextRenderer::loadIntoTexture() {
+    ALOGV("Font::loadIntoTexture");
+
+    glGenTextures(1, &mTextureName);
+    if (mTextureName == 0) {
+        ALOGE("glGenTextures failed: %#x", glGetError());
+        return UNKNOWN_ERROR;
+    }
+    glBindTexture(GL_TEXTURE_2D, mTextureName);
+    glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_LINEAR);
+    glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
+
+    // The pixel data is stored as combined color+alpha, 8 bits per pixel.
+    // It's guaranteed to be a power-of-two wide, but we cut off the height
+    // where the data ends.  We want to expand it to a power-of-two bitmap
+    // with ARGB data and hand that to glTexImage2D.
+
+    if (!isPowerOfTwo(FontBitmap::width)) {
+        ALOGE("npot glyph bitmap width %u", FontBitmap::width);
+        return UNKNOWN_ERROR;
+    }
+
+    uint32_t potHeight = powerOfTwoCeil(FontBitmap::height);
+    uint8_t* rgbaPixels = new uint8_t[FontBitmap::width * potHeight * 4];
+    memset(rgbaPixels, 0, FontBitmap::width * potHeight * 4);
+    uint8_t* pix = rgbaPixels;
+
+    for (unsigned int i = 0; i < FontBitmap::width * FontBitmap::height; i++) {
+        uint8_t alpha, color;
+        if ((FontBitmap::pixels[i] & 1) == 0) {
+            // black pixel with varying alpha
+            color = 0x00;
+            alpha = FontBitmap::pixels[i] & ~1;
+        } else {
+            // opaque grey pixel
+            color = FontBitmap::pixels[i] & ~1;
+            alpha = 0xff;
+        }
+        *pix++ = color;
+        *pix++ = color;
+        *pix++ = color;
+        *pix++ = alpha;
+    }
+
+    glTexImage2D(GL_TEXTURE_2D, 0, GL_RGBA, FontBitmap::width, potHeight, 0,
+            GL_RGBA, GL_UNSIGNED_BYTE, rgbaPixels);
+    delete[] rgbaPixels;
+    GLint glErr = glGetError();
+    if (glErr != 0) {
+        ALOGE("glTexImage2D failed: %#x", glErr);
+        return UNKNOWN_ERROR;
+    }
+    return NO_ERROR;
+}
+
+void TextRenderer::setProportionalScale(float linesPerScreen) {
+    if (mScreenWidth == 0 || mScreenHeight == 0) {
+        ALOGW("setFontScale: can't set scale for width=%d height=%d",
+                mScreenWidth, mScreenHeight);
+        return;
+    }
+    float tallest = mScreenWidth > mScreenHeight ? mScreenWidth : mScreenHeight;
+    setScale(tallest / (linesPerScreen * getGlyphHeight()));
+}
+
+float TextRenderer::computeScaledStringWidth(const String8& str8) const {
+    // String8.length() isn't documented, but I'm assuming it will return
+    // the number of characters rather than the number of bytes.  Since
+    // we can only display ASCII we want to ignore anything else, so we
+    // just convert to char* -- but String8 doesn't document what it does
+    // with values outside 0-255.  So just convert to char* and use strlen()
+    // to see what we get.
+    const char* str = str8.string();
+    return computeScaledStringWidth(str, strlen(str));
+}
+
+size_t TextRenderer::glyphIndex(char ch) const {
+    size_t chi = ch - FontBitmap::firstGlyphChar;
+    if (chi >= FontBitmap::numGlyphs) {
+        chi = '?' - FontBitmap::firstGlyphChar;
+    }
+    assert(chi < FontBitmap::numGlyphs);
+    return chi;
+}
+
+float TextRenderer::computeScaledStringWidth(const char* str,
+        size_t len) const {
+    float width = 0.0f;
+    for (size_t i = 0; i < len; i++) {
+        size_t chi = glyphIndex(str[i]);
+        float glyphWidth = FontBitmap::glyphWidth[chi];
+        width += (glyphWidth - 1 - FontBitmap::outlineWidth) * mScale;
+    }
+
+    return width;
+}
+
+void TextRenderer::drawString(const Program& program, const float* texMatrix,
+        float x, float y, const String8& str8) const {
+    ALOGV("drawString %.3f,%.3f '%s' (scale=%.3f)", x, y, str8.string(),mScale);
+    initOnce();
+
+    // We want to draw the entire string with a single GLES call.  We
+    // generate two arrays, one with screen coordinates, one with texture
+    // coordinates.  Need two triangles per character.
+    const char* str = str8.string();
+    size_t len = strlen(str);       // again, unsure about String8 handling
+
+    const size_t quadCoords =
+            2 /*triangles*/ * 3 /*vertex/tri*/ * 2 /*coord/vertex*/;
+    float vertices[len * quadCoords];
+    float texes[len * quadCoords];
+
+    float fullTexWidth = FontBitmap::width;
+    float fullTexHeight = powerOfTwoCeil(FontBitmap::height);
+    for (size_t i = 0; i < len; i++) {
+        size_t chi = glyphIndex(str[i]);
+        float glyphWidth = FontBitmap::glyphWidth[chi];
+        float glyphHeight = FontBitmap::maxGlyphHeight;
+
+        float vertLeft = x;
+        float vertRight = x + glyphWidth * mScale;
+        float vertTop = y;
+        float vertBottom = y + glyphHeight * mScale;
+
+        // Lowest-numbered glyph is in top-left of bitmap, which puts it at
+        // the bottom-left in texture coordinates.
+        float texLeft = mXOffset[chi] / fullTexWidth;
+        float texRight = (mXOffset[chi] + glyphWidth) / fullTexWidth;
+        float texTop = FontBitmap::yoffset[chi] / fullTexHeight;
+        float texBottom = (FontBitmap::yoffset[chi] + glyphHeight) /
+                fullTexHeight;
+
+        size_t off = i * quadCoords;
+        vertices[off +  0] = vertLeft;
+        vertices[off +  1] = vertBottom;
+        vertices[off +  2] = vertRight;
+        vertices[off +  3] = vertBottom;
+        vertices[off +  4] = vertLeft;
+        vertices[off +  5] = vertTop;
+        vertices[off +  6] = vertLeft;
+        vertices[off +  7] = vertTop;
+        vertices[off +  8] = vertRight;
+        vertices[off +  9] = vertBottom;
+        vertices[off + 10] = vertRight;
+        vertices[off + 11] = vertTop;
+        texes[off +  0] = texLeft;
+        texes[off +  1] = texBottom;
+        texes[off +  2] = texRight;
+        texes[off +  3] = texBottom;
+        texes[off +  4] = texLeft;
+        texes[off +  5] = texTop;
+        texes[off +  6] = texLeft;
+        texes[off +  7] = texTop;
+        texes[off +  8] = texRight;
+        texes[off +  9] = texBottom;
+        texes[off + 10] = texRight;
+        texes[off + 11] = texTop;
+
+        // We added 1-pixel padding in the texture, so we want to advance by
+        // one less.  Also, each glyph is surrounded by a black outline, which
+        // we want to merge.
+        x += (glyphWidth - 1 - FontBitmap::outlineWidth) * mScale;
+    }
+
+    program.drawTriangles(mTextureName, texMatrix, vertices, texes,
+            len * quadCoords / 2);
+}
+
+float TextRenderer::drawWrappedString(const Program& texRender,
+        float xpos, float ypos, const String8& str) {
+    ALOGV("drawWrappedString %.3f,%.3f '%s'", xpos, ypos, str.string());
+    initOnce();
+
+    if (mScreenWidth == 0 || mScreenHeight == 0) {
+        ALOGW("drawWrappedString: can't wrap with width=%d height=%d",
+                mScreenWidth, mScreenHeight);
+        return ypos;
+    }
+
+    const float indentWidth = mIndentMult * getScale();
+    if (xpos < mBorderWidth) {
+        xpos = mBorderWidth;
+    }
+    if (ypos < mBorderWidth) {
+        ypos = mBorderWidth;
+    }
+
+    const size_t maxWidth = (mScreenWidth - mBorderWidth) - xpos;
+    if (maxWidth < 1) {
+        ALOGE("Unable to render text: xpos=%.3f border=%.3f width=%u",
+                xpos, mBorderWidth, mScreenWidth);
+        return ypos;
+    }
+    float stringWidth = computeScaledStringWidth(str);
+    if (stringWidth <= maxWidth) {
+        // Trivial case.
+        drawString(texRender, Program::kIdentity, xpos, ypos, str);
+        ypos += getScaledGlyphHeight();
+    } else {
+        // We need to break the string into pieces, ideally at whitespace
+        // boundaries.
+        char* mangle = strdup(str.string());
+        char* start = mangle;
+        while (start != NULL) {
+            float xposAdj = (start == mangle) ? xpos : xpos + indentWidth;
+            char* brk = breakString(start,
+                    (float) (mScreenWidth - mBorderWidth - xposAdj));
+            if (brk == NULL) {
+                // draw full string
+                drawString(texRender, Program::kIdentity, xposAdj, ypos,
+                        String8(start));
+                start = NULL;
+            } else {
+                // draw partial string
+                char ch = *brk;
+                *brk = '\0';
+                drawString(texRender, Program::kIdentity, xposAdj, ypos,
+                        String8(start));
+                *brk = ch;
+                start = brk;
+                if (strchr(kWhitespace, ch) != NULL) {
+                    // if we broke on whitespace, skip past it
+                    start++;
+                }
+            }
+            ypos += getScaledGlyphHeight();
+        }
+        free(mangle);
+    }
+
+    return ypos;
+}
+
+char* TextRenderer::breakString(const char* str, float maxWidth) const {
+    // Ideally we'd do clever things like binary search.  Not bothering.
+    ALOGV("breakString '%s' %.3f", str, maxWidth);
+
+    size_t len = strlen(str);
+    if (len == 0) {
+        // Caller should detect this and not advance ypos.
+        return NULL;
+    }
+
+    float stringWidth = computeScaledStringWidth(str, len);
+    if (stringWidth <= maxWidth) {
+        return NULL;        // trivial -- use full string
+    }
+
+    // Find the longest string that will fit.
+    size_t goodPos = 0;
+    for (size_t i = 0; i < len; i++) {
+        stringWidth = computeScaledStringWidth(str, i);
+        if (stringWidth < maxWidth) {
+            goodPos = i;
+        } else {
+            break;  // too big
+        }
+    }
+    if (goodPos == 0) {
+        // space is too small to hold any glyph; output a single char
+        ALOGW("Couldn't find a nonzero prefix that fit from '%s'", str);
+        goodPos = 1;
+    }
+
+    // Scan back for whitespace.  If we can't find any we'll just have
+    // an ugly mid-word break.
+    for (size_t i = goodPos; i > 0; i--) {
+        if (strchr(kWhitespace, str[i]) != NULL) {
+            goodPos = i;
+            break;
+        }
+    }
+
+    ALOGV("goodPos=%d for str='%s'", goodPos, str);
+    return const_cast<char*>(str + goodPos);
+}

cmds/screenrecord/TextRenderer.h

diff --git a/cmds/screenrecord/TextRenderer.h b/cmds/screenrecord/TextRenderer.h
new file mode 100644
index 0000000..03dd2fb
--- /dev/null
+++ b/cmds/screenrecord/TextRenderer.h
@@ -0,0 +1,140 @@
+/*
+ * Copyright 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef SCREENRECORD_TEXT_RENDER_H
+#define SCREENRECORD_TEXT_RENDER_H
+
+#include "Program.h"
+
+#include <utils/String8.h>
+#include <utils/Errors.h>
+
+#include <GLES2/gl2.h>
+
+
+namespace android {
+
+/*
+ * Simple font representation.
+ *
+ * Not thread-safe.
+ */
+class TextRenderer {
+public:
+    TextRenderer() :
+        mTextureName(0),
+        mScale(1.0f),
+        mBorderWidth(10.0f),
+        mIndentMult(30.0f),
+        mScreenWidth(0),
+        mScreenHeight(0)
+        {}
+    ~TextRenderer() {}
+
+    // Load the glyph bitmap into a 2D texture in the current context.
+    status_t loadIntoTexture();
+
+    // Set the screen dimensions, used for scaling and line wrap.
+    void setScreenSize(uint32_t width, uint32_t height) {
+        mScreenWidth = width;
+        mScreenHeight = height;
+    }
+
+    // Get/set the font scaling.
+    float getScale() const { return mScale; }
+    void setScale(float scale) { mScale = scale; }
+
+    // Set the font scaling based on the desired number of lines per screen.
+    // The display's tallest axis is used, so if the device is in landscape
+    // the screen will fit fewer lines.
+    void setProportionalScale(float linesPerScreen);
+
+    // Render the text string at the specified coordinates.  Pass in the
+    // upper-left corner in non-GL-flipped coordinates, i.e. to print text
+    // at the top left of the screen use (0,0).
+    //
+    // Set blend func (1, 1-srcAlpha) before calling if drawing onto
+    // something other than black.
+    void drawString(const Program& program, const float* texMatrix,
+            float x, float y, const String8& str) const;
+
+    // Draw a string, possibly wrapping it at the screen boundary.  Top-left
+    // is at (0,0).
+    //
+    // Returns the updated Y position.
+    float drawWrappedString(const Program& texRender, float xpos, float ypos,
+            const String8& str);
+
+    // Returns the name of the texture the font was loaded into.
+    GLuint getTextureName() const { return mTextureName; }
+
+private:
+    TextRenderer(const TextRenderer&);
+    TextRenderer& operator=(const TextRenderer&);
+
+    // Perform one-time initialization.
+    static void initOnce();
+
+    // Populate the mXOffset array.
+    static void initXOffset();
+
+    // Find a good place to break the string.  Returns NULL if the entire
+    // string will fit.
+    char* breakString(const char* str, float maxWidth) const;
+
+    // Computes the width of the string, in pixels.
+    float computeScaledStringWidth(const String8& str8) const;
+
+    // Computes the width of first N characters in the string.
+    float computeScaledStringWidth(const char* str, size_t len) const;
+
+    // Returns the font's glyph height.  This is the full pixel height of the
+    // tallest glyph, both above and below the baseline, NOT adjusted by the
+    // current scale factor.
+    float getGlyphHeight() const;
+
+    // Like getGlyphHeight(), but result is scaled.
+    float getScaledGlyphHeight() const { return getGlyphHeight() * mScale; }
+
+    // Convert an ASCII character to a glyph index.  Returns the glyph for
+    // '?' if we have no glyph for the specified character.
+    size_t glyphIndex(char ch) const;
+
+    GLuint mTextureName;
+    float mScale;
+
+    // Number of pixels preserved at the left/right edges of the screen by
+    // drawWrappedString().  Not scaled.
+    float mBorderWidth;
+
+    // Distance to indent a broken line.  Used by drawWrappedString().
+    // Value will be adjusted by the current scale factor.
+    float mIndentMult;
+
+    // Screen dimensions.
+    uint32_t mScreenWidth;
+    uint32_t mScreenHeight;
+
+    // Static font info.
+    static bool mInitialized;
+    static uint32_t mXOffset[];
+
+    static const char kWhitespace[];
+};
+
+}; // namespace android
+
+#endif /*SCREENRECORD_TEXT_RENDER_H*/

cmds/screenrecord/screenrecord.cpp

diff --git a/cmds/screenrecord/screenrecord.cpp b/cmds/screenrecord/screenrecord.cpp
index 49999b5..a17fc51 100644
--- a/cmds/screenrecord/screenrecord.cpp
+++ b/cmds/screenrecord/screenrecord.cpp
@@ -15,13 +15,14 @@
  */
 
 #define LOG_TAG "ScreenRecord"
+#define ATRACE_TAG ATRACE_TAG_GRAPHICS
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
 #include <binder/IPCThreadState.h>
 #include <utils/Errors.h>
-#include <utils/Thread.h>
 #include <utils/Timers.h>
+#include <utils/Trace.h>
 
 #include <gui/Surface.h>
 #include <gui/SurfaceComposerClient.h>
@@ -29,7 +30,6 @@
 #include <ui/DisplayInfo.h>
 #include <media/openmax/OMX_IVCommon.h>
 #include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AMessage.h>
 #include <media/stagefright/MediaCodec.h>
 #include <media/stagefright/MediaErrors.h>
@@ -40,30 +40,43 @@
 #include <unistd.h>
 #include <string.h>
 #include <stdio.h>
+#include <ctype.h>
 #include <fcntl.h>
 #include <signal.h>
 #include <getopt.h>
 #include <sys/wait.h>
+#include <termios.h>
+#include <assert.h>
+
+#include "screenrecord.h"
+#include "Overlay.h"
+#include "FrameOutput.h"
 
 using namespace android;
 
 static const uint32_t kMinBitRate = 100000;         // 0.1Mbps
-static const uint32_t kMaxBitRate = 100 * 1000000;  // 100Mbps
+static const uint32_t kMaxBitRate = 200 * 1000000;  // 200Mbps
 static const uint32_t kMaxTimeLimitSec = 180;       // 3 minutes
 static const uint32_t kFallbackWidth = 1280;        // 720p
 static const uint32_t kFallbackHeight = 720;
+static const char* kMimeTypeAvc = "video/avc";
 
 // Command-line parameters.
-static bool gVerbose = false;               // chatty on stdout
-static bool gRotate = false;                // rotate 90 degrees
-static bool gSizeSpecified = false;         // was size explicitly requested?
-static uint32_t gVideoWidth = 0;            // default width+height
+static bool gVerbose = false;           // chatty on stdout
+static bool gRotate = false;            // rotate 90 degrees
+static enum {
+    FORMAT_MP4, FORMAT_H264, FORMAT_FRAMES
+} gOutputFormat = FORMAT_MP4;           // data format for output
+static bool gSizeSpecified = false;     // was size explicitly requested?
+static bool gWantInfoScreen = false;    // do we want initial info screen?
+static bool gWantFrameTime = false;     // do we want times on each frame?
+static uint32_t gVideoWidth = 0;        // default width+height
 static uint32_t gVideoHeight = 0;
-static uint32_t gBitRate = 4000000;         // 4Mbps
+static uint32_t gBitRate = 4000000;     // 4Mbps
 static uint32_t gTimeLimitSec = kMaxTimeLimitSec;
 
 // Set by signal handler to stop recording.
-static bool gStopRequested;
+static volatile bool gStopRequested;
 
 // Previous signal handler state, restored after first hit.
 static struct sigaction gOrigSigactionINT;
@@ -97,8 +110,7 @@
  * when Ctrl-C is hit.  If we're run from the host, the local adb process
  * gets the signal, and we get a SIGHUP when the terminal disconnects.
  */
-static status_t configureSignals()
-{
+static status_t configureSignals() {
     struct sigaction act;
     memset(&act, 0, sizeof(act));
     act.sa_handler = signalCatcher;
@@ -134,14 +146,14 @@
     status_t err;
 
     if (gVerbose) {
-        printf("Configuring recorder for %dx%d video at %.2fMbps\n",
-                gVideoWidth, gVideoHeight, gBitRate / 1000000.0);
+        printf("Configuring recorder for %dx%d %s at %.2fMbps\n",
+                gVideoWidth, gVideoHeight, kMimeTypeAvc, gBitRate / 1000000.0);
     }
 
     sp<AMessage> format = new AMessage;
     format->setInt32("width", gVideoWidth);
     format->setInt32("height", gVideoHeight);
-    format->setString("mime", "video/avc");
+    format->setString("mime", kMimeTypeAvc);
     format->setInt32("color-format", OMX_COLOR_FormatAndroidOpaque);
     format->setInt32("bitrate", gBitRate);
     format->setFloat("frame-rate", displayFps);
@@ -151,40 +163,37 @@
     looper->setName("screenrecord_looper");
     looper->start();
     ALOGV("Creating codec");
-    sp<MediaCodec> codec = MediaCodec::CreateByType(looper, "video/avc", true);
+    sp<MediaCodec> codec = MediaCodec::CreateByType(looper, kMimeTypeAvc, true);
     if (codec == NULL) {
-        fprintf(stderr, "ERROR: unable to create video/avc codec instance\n");
+        fprintf(stderr, "ERROR: unable to create %s codec instance\n",
+                kMimeTypeAvc);
         return UNKNOWN_ERROR;
     }
+
     err = codec->configure(format, NULL, NULL,
             MediaCodec::CONFIGURE_FLAG_ENCODE);
     if (err != NO_ERROR) {
+        fprintf(stderr, "ERROR: unable to configure %s codec at %dx%d (err=%d)\n",
+                kMimeTypeAvc, gVideoWidth, gVideoHeight, err);
         codec->release();
-        codec.clear();
-
-        fprintf(stderr, "ERROR: unable to configure codec (err=%d)\n", err);
         return err;
     }
 
-    ALOGV("Creating buffer producer");
+    ALOGV("Creating encoder input surface");
     sp<IGraphicBufferProducer> bufferProducer;
     err = codec->createInputSurface(&bufferProducer);
     if (err != NO_ERROR) {
-        codec->release();
-        codec.clear();
-
         fprintf(stderr,
             "ERROR: unable to create encoder input surface (err=%d)\n", err);
+        codec->release();
         return err;
     }
 
     ALOGV("Starting codec");
     err = codec->start();
     if (err != NO_ERROR) {
-        codec->release();
-        codec.clear();
-
         fprintf(stderr, "ERROR: unable to start codec (err=%d)\n", err);
+        codec->release();
         return err;
     }
 
@@ -195,12 +204,11 @@
 }
 
 /*
- * Configures the virtual display.  When this completes, virtual display
- * frames will start being sent to the encoder's surface.
+ * Sets the display projection, based on the display dimensions, video size,
+ * and device orientation.
  */
-static status_t prepareVirtualDisplay(const DisplayInfo& mainDpyInfo,
-        const sp<IGraphicBufferProducer>& bufferProducer,
-        sp<IBinder>* pDisplayHandle) {
+static status_t setDisplayProjection(const sp<IBinder>& dpy,
+        const DisplayInfo& mainDpyInfo) {
     status_t err;
 
     // Set the region of the layer stack we're interested in, which in our
@@ -266,15 +274,25 @@
         }
     }
 
-
-    sp<IBinder> dpy = SurfaceComposerClient::createDisplay(
-            String8("ScreenRecorder"), false /* secure */);
-
-    SurfaceComposerClient::openGlobalTransaction();
-    SurfaceComposerClient::setDisplaySurface(dpy, bufferProducer);
     SurfaceComposerClient::setDisplayProjection(dpy,
             gRotate ? DISPLAY_ORIENTATION_90 : DISPLAY_ORIENTATION_0,
             layerStackRect, displayRect);
+    return NO_ERROR;
+}
+
+/*
+ * Configures the virtual display.  When this completes, virtual display
+ * frames will start arriving from the buffer producer.
+ */
+static status_t prepareVirtualDisplay(const DisplayInfo& mainDpyInfo,
+        const sp<IGraphicBufferProducer>& bufferProducer,
+        sp<IBinder>* pDisplayHandle) {
+    sp<IBinder> dpy = SurfaceComposerClient::createDisplay(
+            String8("ScreenRecorder"), false /*secure*/);
+
+    SurfaceComposerClient::openGlobalTransaction();
+    SurfaceComposerClient::setDisplaySurface(dpy, bufferProducer);
+    setDisplayProjection(dpy, mainDpyInfo);
     SurfaceComposerClient::setDisplayLayerStack(dpy, 0);    // default stack
     SurfaceComposerClient::closeGlobalTransaction();
 
@@ -288,16 +306,22 @@
  * input frames are coming from the virtual display as fast as SurfaceFlinger
  * wants to send them.
  *
+ * Exactly one of muxer or rawFp must be non-null.
+ *
  * The muxer must *not* have been started before calling.
  */
 static status_t runEncoder(const sp<MediaCodec>& encoder,
-        const sp<MediaMuxer>& muxer) {
+        const sp<MediaMuxer>& muxer, FILE* rawFp, const sp<IBinder>& mainDpy,
+        const sp<IBinder>& virtualDpy, uint8_t orientation) {
     static int kTimeout = 250000;   // be responsive on signal
     status_t err;
     ssize_t trackIdx = -1;
     uint32_t debugNumFrames = 0;
     int64_t startWhenNsec = systemTime(CLOCK_MONOTONIC);
     int64_t endWhenNsec = startWhenNsec + seconds_to_nanoseconds(gTimeLimitSec);
+    DisplayInfo mainDpyInfo;
+
+    assert((rawFp == NULL && muxer != NULL) || (rawFp != NULL && muxer == NULL));
 
     Vector<sp<ABuffer> > buffers;
     err = encoder->getOutputBuffers(&buffers);
@@ -330,31 +354,68 @@
         case NO_ERROR:
             // got a buffer
             if ((flags & MediaCodec::BUFFER_FLAG_CODECCONFIG) != 0) {
-                // ignore this -- we passed the CSD into MediaMuxer when
-                // we got the format change notification
-                ALOGV("Got codec config buffer (%u bytes); ignoring", size);
-                size = 0;
+                ALOGV("Got codec config buffer (%u bytes)", size);
+                if (muxer != NULL) {
+                    // ignore this -- we passed the CSD into MediaMuxer when
+                    // we got the format change notification
+                    size = 0;
+                }
             }
             if (size != 0) {
                 ALOGV("Got data in buffer %d, size=%d, pts=%lld",
                         bufIndex, size, ptsUsec);
-                CHECK(trackIdx != -1);
+
+                { // scope
+                    ATRACE_NAME("orientation");
+                    // Check orientation, update if it has changed.
+                    //
+                    // Polling for changes is inefficient and wrong, but the
+                    // useful stuff is hard to get at without a Dalvik VM.
+                    err = SurfaceComposerClient::getDisplayInfo(mainDpy,
+                            &mainDpyInfo);
+                    if (err != NO_ERROR) {
+                        ALOGW("getDisplayInfo(main) failed: %d", err);
+                    } else if (orientation != mainDpyInfo.orientation) {
+                        ALOGD("orientation changed, now %d", mainDpyInfo.orientation);
+                        SurfaceComposerClient::openGlobalTransaction();
+                        setDisplayProjection(virtualDpy, mainDpyInfo);
+                        SurfaceComposerClient::closeGlobalTransaction();
+                        orientation = mainDpyInfo.orientation;
+                    }
+                }
 
                 // If the virtual display isn't providing us with timestamps,
-                // use the current time.
+                // use the current time.  This isn't great -- we could get
+                // decoded data in clusters -- but we're not expecting
+                // to hit this anyway.
                 if (ptsUsec == 0) {
                     ptsUsec = systemTime(SYSTEM_TIME_MONOTONIC) / 1000;
                 }
 
-                // The MediaMuxer docs are unclear, but it appears that we
-                // need to pass either the full set of BufferInfo flags, or
-                // (flags & BUFFER_FLAG_SYNCFRAME).
-                err = muxer->writeSampleData(buffers[bufIndex], trackIdx,
-                        ptsUsec, flags);
-                if (err != NO_ERROR) {
-                    fprintf(stderr, "Failed writing data to muxer (err=%d)\n",
-                            err);
-                    return err;
+                if (muxer == NULL) {
+                    fwrite(buffers[bufIndex]->data(), 1, size, rawFp);
+                    // Flush the data immediately in case we're streaming.
+                    // We don't want to do this if all we've written is
+                    // the SPS/PPS data because mplayer gets confused.
+                    if ((flags & MediaCodec::BUFFER_FLAG_CODECCONFIG) == 0) {
+                        fflush(rawFp);
+                    }
+                } else {
+                    // The MediaMuxer docs are unclear, but it appears that we
+                    // need to pass either the full set of BufferInfo flags, or
+                    // (flags & BUFFER_FLAG_SYNCFRAME).
+                    //
+                    // If this blocks for too long we could drop frames.  We may
+                    // want to queue these up and do them on a different thread.
+                    ATRACE_NAME("write sample");
+                    assert(trackIdx != -1);
+                    err = muxer->writeSampleData(buffers[bufIndex], trackIdx,
+                            ptsUsec, flags);
+                    if (err != NO_ERROR) {
+                        fprintf(stderr,
+                            "Failed writing data to muxer (err=%d)\n", err);
+                        return err;
+                    }
                 }
                 debugNumFrames++;
             }
@@ -366,8 +427,8 @@
             }
             if ((flags & MediaCodec::BUFFER_FLAG_EOS) != 0) {
                 // Not expecting EOS from SurfaceFlinger.  Go with it.
-                ALOGD("Received end-of-stream");
-                gStopRequested = false;
+                ALOGI("Received end-of-stream");
+                gStopRequested = true;
             }
             break;
         case -EAGAIN:                       // INFO_TRY_AGAIN_LATER
@@ -375,21 +436,23 @@
             break;
         case INFO_FORMAT_CHANGED:           // INFO_OUTPUT_FORMAT_CHANGED
             {
-                // format includes CSD, which we must provide to muxer
+                // Format includes CSD, which we must provide to muxer.
                 ALOGV("Encoder format changed");
                 sp<AMessage> newFormat;
                 encoder->getOutputFormat(&newFormat);
-                trackIdx = muxer->addTrack(newFormat);
-                ALOGV("Starting muxer");
-                err = muxer->start();
-                if (err != NO_ERROR) {
-                    fprintf(stderr, "Unable to start muxer (err=%d)\n", err);
-                    return err;
+                if (muxer != NULL) {
+                    trackIdx = muxer->addTrack(newFormat);
+                    ALOGV("Starting muxer");
+                    err = muxer->start();
+                    if (err != NO_ERROR) {
+                        fprintf(stderr, "Unable to start muxer (err=%d)\n", err);
+                        return err;
+                    }
                 }
             }
             break;
         case INFO_OUTPUT_BUFFERS_CHANGED:   // INFO_OUTPUT_BUFFERS_CHANGED
-            // not expected for an encoder; handle it anyway
+            // Not expected for an encoder; handle it anyway.
             ALOGV("Encoder buffers changed");
             err = encoder->getOutputBuffers(&buffers);
             if (err != NO_ERROR) {
@@ -399,7 +462,7 @@
             }
             break;
         case INVALID_OPERATION:
-            fprintf(stderr, "Request for encoder buffer failed\n");
+            ALOGW("dequeueOutputBuffer returned INVALID_OPERATION");
             return err;
         default:
             fprintf(stderr,
@@ -411,14 +474,52 @@
     ALOGV("Encoder stopping (req=%d)", gStopRequested);
     if (gVerbose) {
         printf("Encoder stopping; recorded %u frames in %lld seconds\n",
-                debugNumFrames,
-                nanoseconds_to_seconds(systemTime(CLOCK_MONOTONIC) - startWhenNsec));
+                debugNumFrames, nanoseconds_to_seconds(
+                        systemTime(CLOCK_MONOTONIC) - startWhenNsec));
     }
     return NO_ERROR;
 }
 
 /*
- * Main "do work" method.
+ * Raw H.264 byte stream output requested.  Send the output to stdout
+ * if desired.  If the output is a tty, reconfigure it to avoid the
+ * CRLF line termination that we see with "adb shell" commands.
+ */
+static FILE* prepareRawOutput(const char* fileName) {
+    FILE* rawFp = NULL;
+
+    if (strcmp(fileName, "-") == 0) {
+        if (gVerbose) {
+            fprintf(stderr, "ERROR: verbose output and '-' not compatible");
+            return NULL;
+        }
+        rawFp = stdout;
+    } else {
+        rawFp = fopen(fileName, "w");
+        if (rawFp == NULL) {
+            fprintf(stderr, "fopen raw failed: %s\n", strerror(errno));
+            return NULL;
+        }
+    }
+
+    int fd = fileno(rawFp);
+    if (isatty(fd)) {
+        // best effort -- reconfigure tty for "raw"
+        ALOGD("raw video output to tty (fd=%d)", fd);
+        struct termios term;
+        if (tcgetattr(fd, &term) == 0) {
+            cfmakeraw(&term);
+            if (tcsetattr(fd, TCSANOW, &term) == 0) {
+                ALOGD("tty successfully configured for raw");
+            }
+        }
+    }
+
+    return rawFp;
+}
+
+/*
+ * Main "do work" start point.
  *
  * Configures codec, muxer, and virtual display, then starts moving bits
  * around.
@@ -460,66 +561,158 @@
 
     // Configure and start the encoder.
     sp<MediaCodec> encoder;
-    sp<IGraphicBufferProducer> bufferProducer;
-    err = prepareEncoder(mainDpyInfo.fps, &encoder, &bufferProducer);
+    sp<FrameOutput> frameOutput;
+    sp<IGraphicBufferProducer> encoderInputSurface;
+    if (gOutputFormat != FORMAT_FRAMES) {
+        err = prepareEncoder(mainDpyInfo.fps, &encoder, &encoderInputSurface);
 
-    if (err != NO_ERROR && !gSizeSpecified) {
-        // fallback is defined for landscape; swap if we're in portrait
-        bool needSwap = gVideoWidth < gVideoHeight;
-        uint32_t newWidth = needSwap ? kFallbackHeight : kFallbackWidth;
-        uint32_t newHeight = needSwap ? kFallbackWidth : kFallbackHeight;
-        if (gVideoWidth != newWidth && gVideoHeight != newHeight) {
-            ALOGV("Retrying with 720p");
-            fprintf(stderr, "WARNING: failed at %dx%d, retrying at %dx%d\n",
-                    gVideoWidth, gVideoHeight, newWidth, newHeight);
-            gVideoWidth = newWidth;
-            gVideoHeight = newHeight;
-            err = prepareEncoder(mainDpyInfo.fps, &encoder, &bufferProducer);
+        if (err != NO_ERROR && !gSizeSpecified) {
+            // fallback is defined for landscape; swap if we're in portrait
+            bool needSwap = gVideoWidth < gVideoHeight;
+            uint32_t newWidth = needSwap ? kFallbackHeight : kFallbackWidth;
+            uint32_t newHeight = needSwap ? kFallbackWidth : kFallbackHeight;
+            if (gVideoWidth != newWidth && gVideoHeight != newHeight) {
+                ALOGV("Retrying with 720p");
+                fprintf(stderr, "WARNING: failed at %dx%d, retrying at %dx%d\n",
+                        gVideoWidth, gVideoHeight, newWidth, newHeight);
+                gVideoWidth = newWidth;
+                gVideoHeight = newHeight;
+                err = prepareEncoder(mainDpyInfo.fps, &encoder,
+                        &encoderInputSurface);
+            }
+        }
+        if (err != NO_ERROR) return err;
+
+        // From here on, we must explicitly release() the encoder before it goes
+        // out of scope, or we will get an assertion failure from stagefright
+        // later on in a different thread.
+    } else {
+        // We're not using an encoder at all.  The "encoder input surface" we hand to
+        // SurfaceFlinger will just feed directly to us.
+        frameOutput = new FrameOutput();
+        err = frameOutput->createInputSurface(gVideoWidth, gVideoHeight, &encoderInputSurface);
+        if (err != NO_ERROR) {
+            return err;
         }
     }
-    if (err != NO_ERROR) {
-        return err;
+
+    // Draw the "info" page by rendering a frame with GLES and sending
+    // it directly to the encoder.
+    // TODO: consider displaying this as a regular layer to avoid b/11697754
+    if (gWantInfoScreen) {
+        Overlay::drawInfoPage(encoderInputSurface);
+    }
+
+    // Configure optional overlay.
+    sp<IGraphicBufferProducer> bufferProducer;
+    sp<Overlay> overlay;
+    if (gWantFrameTime) {
+        // Send virtual display frames to an external texture.
+        overlay = new Overlay();
+        err = overlay->start(encoderInputSurface, &bufferProducer);
+        if (err != NO_ERROR) {
+            if (encoder != NULL) encoder->release();
+            return err;
+        }
+        if (gVerbose) {
+            printf("Bugreport overlay created\n");
+        }
+    } else {
+        // Use the encoder's input surface as the virtual display surface.
+        bufferProducer = encoderInputSurface;
     }
 
     // Configure virtual display.
     sp<IBinder> dpy;
     err = prepareVirtualDisplay(mainDpyInfo, bufferProducer, &dpy);
     if (err != NO_ERROR) {
-        encoder->release();
-        encoder.clear();
-
+        if (encoder != NULL) encoder->release();
         return err;
     }
 
-    // Configure, but do not start, muxer.
-    sp<MediaMuxer> muxer = new MediaMuxer(fileName,
-            MediaMuxer::OUTPUT_FORMAT_MPEG_4);
-    if (gRotate) {
-        muxer->setOrientationHint(90);
+    sp<MediaMuxer> muxer = NULL;
+    FILE* rawFp = NULL;
+    switch (gOutputFormat) {
+        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);
+            if (gRotate) {
+                muxer->setOrientationHint(90);  // TODO: does this do anything?
+            }
+            break;
+        }
+        case FORMAT_H264:
+        case FORMAT_FRAMES: {
+            rawFp = prepareRawOutput(fileName);
+            if (rawFp == NULL) {
+                if (encoder != NULL) encoder->release();
+                return -1;
+            }
+            break;
+        }
+        default:
+            fprintf(stderr, "ERROR: unknown format %d\n", gOutputFormat);
+            abort();
     }
 
-    // Main encoder loop.
-    err = runEncoder(encoder, muxer);
-    if (err != NO_ERROR) {
-        encoder->release();
-        encoder.clear();
+    if (gOutputFormat == FORMAT_FRAMES) {
+        // TODO: if we want to make this a proper feature, we should output
+        //       an outer header with version info.  Right now we never change
+        //       the frame size or format, so we could conceivably just send
+        //       the current frame header once and then follow it with an
+        //       unbroken stream of data.
 
-        return err;
-    }
+        // Make the EGL context current again.  This gets unhooked if we're
+        // using "--bugreport" mode.
+        // TODO: figure out if we can eliminate this
+        frameOutput->prepareToCopy();
 
-    if (gVerbose) {
-        printf("Stopping encoder and muxer\n");
+        while (!gStopRequested) {
+            // Poll for frames, the same way we do for MediaCodec.  We do
+            // all of the work on the main thread.
+            //
+            // Ideally we'd sleep indefinitely and wake when the
+            // stop was requested, but this will do for now.  (It almost
+            // works because wait() wakes when a signal hits, but we
+            // need to handle the edge cases.)
+            err = frameOutput->copyFrame(rawFp, 250000);
+            if (err == ETIMEDOUT) {
+                err = NO_ERROR;
+            } else if (err != NO_ERROR) {
+                ALOGE("Got error %d from copyFrame()", err);
+                break;
+            }
+        }
+    } else {
+        // Main encoder loop.
+        err = runEncoder(encoder, muxer, rawFp, mainDpy, dpy,
+                mainDpyInfo.orientation);
+        if (err != NO_ERROR) {
+            fprintf(stderr, "Encoder failed (err=%d)\n", err);
+            // fall through to cleanup
+        }
+
+        if (gVerbose) {
+            printf("Stopping encoder and muxer\n");
+        }
     }
 
     // Shut everything down, starting with the producer side.
-    bufferProducer = NULL;
+    encoderInputSurface = NULL;
     SurfaceComposerClient::destroyDisplay(dpy);
+    if (overlay != NULL) overlay->stop();
+    if (encoder != NULL) encoder->stop();
+    if (muxer != NULL) {
+        // If we don't stop muxer explicitly, i.e. let the destructor run,
+        // it may hang (b/11050628).
+        muxer->stop();
+    } else if (rawFp != stdout) {
+        fclose(rawFp);
+    }
+    if (encoder != NULL) encoder->release();
 
-    encoder->stop();
-    muxer->stop();
-    encoder->release();
-
-    return 0;
+    return err;
 }
 
 /*
@@ -528,6 +721,28 @@
  * This is optional, but nice to have.
  */
 static status_t notifyMediaScanner(const char* fileName) {
+    // need to do allocations before the fork()
+    String8 fileUrl("file://");
+    fileUrl.append(fileName);
+
+    const char* kCommand = "/system/bin/am";
+    const char* const argv[] = {
+            kCommand,
+            "broadcast",
+            "-a",
+            "android.intent.action.MEDIA_SCANNER_SCAN_FILE",
+            "-d",
+            fileUrl.string(),
+            NULL
+    };
+    if (gVerbose) {
+        printf("Executing:");
+        for (int i = 0; argv[i] != NULL; i++) {
+            printf(" %s", argv[i]);
+        }
+        putchar('\n');
+    }
+
     pid_t pid = fork();
     if (pid < 0) {
         int err = errno;
@@ -539,34 +754,14 @@
         int status;
         pid_t actualPid = TEMP_FAILURE_RETRY(waitpid(pid, &status, 0));
         if (actualPid != pid) {
-            ALOGW("waitpid() returned %d (errno=%d)", actualPid, errno);
+            ALOGW("waitpid(%d) returned %d (errno=%d)", pid, actualPid, errno);
         } else if (status != 0) {
             ALOGW("'am broadcast' exited with status=%d", status);
         } else {
             ALOGV("'am broadcast' exited successfully");
         }
     } else {
-        const char* kCommand = "/system/bin/am";
-
-        // child; we're single-threaded, so okay to alloc
-        String8 fileUrl("file://");
-        fileUrl.append(fileName);
-        const char* const argv[] = {
-                kCommand,
-                "broadcast",
-                "-a",
-                "android.intent.action.MEDIA_SCANNER_SCAN_FILE",
-                "-d",
-                fileUrl.string(),
-                NULL
-        };
-        if (gVerbose) {
-            printf("Executing:");
-            for (int i = 0; argv[i] != NULL; i++) {
-                printf(" %s", argv[i]);
-            }
-            putchar('\n');
-        } else {
+        if (!gVerbose) {
             // non-verbose, suppress 'am' output
             ALOGV("closing stdout/stderr in child");
             int fd = open("/dev/null", O_WRONLY);
@@ -611,13 +806,37 @@
 }
 
 /*
+ * Accepts a string with a bare number ("4000000") or with a single-character
+ * unit ("4m").
+ *
+ * Returns an error if parsing fails.
+ */
+static status_t parseValueWithUnit(const char* str, uint32_t* pValue) {
+    long value;
+    char* endptr;
+
+    value = strtol(str, &endptr, 10);
+    if (*endptr == '\0') {
+        // bare number
+        *pValue = value;
+        return NO_ERROR;
+    } else if (toupper(*endptr) == 'M' && *(endptr+1) == '\0') {
+        *pValue = value * 1000000;  // check for overflow?
+        return NO_ERROR;
+    } else {
+        fprintf(stderr, "Unrecognized value: %s\n", str);
+        return UNKNOWN_ERROR;
+    }
+}
+
+/*
  * Dumps usage on stderr.
  */
 static void usage() {
     fprintf(stderr,
         "Usage: screenrecord [options] <filename>\n"
         "\n"
-        "Records the device's display to a .mp4 file.\n"
+        "Android screenrecord v%d.%d.  Records the device's display to a .mp4 file.\n"
         "\n"
         "Options:\n"
         "--size WIDTHxHEIGHT\n"
@@ -625,11 +844,13 @@
         "    display resolution (if supported), 1280x720 if not.  For best results,\n"
         "    use a size supported by the AVC encoder.\n"
         "--bit-rate RATE\n"
-        "    Set the video bit rate, in megabits per second.  Default %dMbps.\n"
+        "    Set the video bit rate, in bits per second.  Value may be specified as\n"
+        "    bits or megabits, e.g. '4000000' is equivalent to '4M'.  Default %dMbps.\n"
+        "--bugreport\n"
+        "    Add additional information, such as a timestamp overlay, that is helpful\n"
+        "    in videos captured to illustrate bugs.\n"
         "--time-limit TIME\n"
         "    Set the maximum recording time, in seconds.  Default / maximum is %d.\n"
-        "--rotate\n"
-        "    Rotate the output 90 degrees.\n"
         "--verbose\n"
         "    Display interesting information on stdout.\n"
         "--help\n"
@@ -637,7 +858,7 @@
         "\n"
         "Recording continues until Ctrl-C is hit or the time limit is reached.\n"
         "\n",
-        gBitRate / 1000000, gTimeLimitSec
+        kVersionMajor, kVersionMinor, gBitRate / 1000000, gTimeLimitSec
         );
 }
 
@@ -646,13 +867,18 @@
  */
 int main(int argc, char* const argv[]) {
     static const struct option longOptions[] = {
-        { "help",       no_argument,        NULL, 'h' },
-        { "verbose",    no_argument,        NULL, 'v' },
-        { "size",       required_argument,  NULL, 's' },
-        { "bit-rate",   required_argument,  NULL, 'b' },
-        { "time-limit", required_argument,  NULL, 't' },
-        { "rotate",     no_argument,        NULL, 'r' },
-        { NULL,         0,                  NULL, 0 }
+        { "help",               no_argument,        NULL, 'h' },
+        { "verbose",            no_argument,        NULL, 'v' },
+        { "size",               required_argument,  NULL, 's' },
+        { "bit-rate",           required_argument,  NULL, 'b' },
+        { "time-limit",         required_argument,  NULL, 't' },
+        { "bugreport",          no_argument,        NULL, 'u' },
+        // "unofficial" options
+        { "show-device-info",   no_argument,        NULL, 'i' },
+        { "show-frame-time",    no_argument,        NULL, 'f' },
+        { "rotate",             no_argument,        NULL, 'r' },
+        { "output-format",      required_argument,  NULL, 'o' },
+        { NULL,                 0,                  NULL, 0 }
     };
 
     while (true) {
@@ -684,7 +910,9 @@
             gSizeSpecified = true;
             break;
         case 'b':
-            gBitRate = atoi(optarg);
+            if (parseValueWithUnit(optarg, &gBitRate) != NO_ERROR) {
+                return 2;
+            }
             if (gBitRate < kMinBitRate || gBitRate > kMaxBitRate) {
                 fprintf(stderr,
                         "Bit rate %dbps outside acceptable range [%d,%d]\n",
@@ -701,9 +929,32 @@
                 return 2;
             }
             break;
+        case 'u':
+            gWantInfoScreen = true;
+            gWantFrameTime = true;
+            break;
+        case 'i':
+            gWantInfoScreen = true;
+            break;
+        case 'f':
+            gWantFrameTime = true;
+            break;
         case 'r':
+            // experimental feature
             gRotate = true;
             break;
+        case 'o':
+            if (strcmp(optarg, "mp4") == 0) {
+                gOutputFormat = FORMAT_MP4;
+            } else if (strcmp(optarg, "h264") == 0) {
+                gOutputFormat = FORMAT_H264;
+            } else if (strcmp(optarg, "frames") == 0) {
+                gOutputFormat = FORMAT_FRAMES;
+            } else {
+                fprintf(stderr, "Unknown format '%s'\n", optarg);
+                return 2;
+            }
+            break;
         default:
             if (ic != '?') {
                 fprintf(stderr, "getopt_long returned unexpected value 0x%x\n", ic);
@@ -717,17 +968,19 @@
         return 2;
     }
 
-    // MediaMuxer tries to create the file in the constructor, but we don't
-    // learn about the failure until muxer.start(), which returns a generic
-    // error code without logging anything.  We attempt to create the file
-    // now for better diagnostics.
     const char* fileName = argv[optind];
-    int fd = open(fileName, O_CREAT | O_RDWR, 0644);
-    if (fd < 0) {
-        fprintf(stderr, "Unable to open '%s': %s\n", fileName, strerror(errno));
-        return 1;
+    if (gOutputFormat == FORMAT_MP4) {
+        // MediaMuxer tries to create the file in the constructor, but we don't
+        // learn about the failure until muxer.start(), which returns a generic
+        // error code without logging anything.  We attempt to create the file
+        // now for better diagnostics.
+        int fd = open(fileName, O_CREAT | O_RDWR, 0644);
+        if (fd < 0) {
+            fprintf(stderr, "Unable to open '%s': %s\n", fileName, strerror(errno));
+            return 1;
+        }
+        close(fd);
     }
-    close(fd);
 
     status_t err = recordScreen(fileName);
     if (err == NO_ERROR) {

cmds/screenrecord/screenrecord.h

diff --git a/cmds/screenrecord/screenrecord.h b/cmds/screenrecord/screenrecord.h
new file mode 100644
index 0000000..9b058c2
--- /dev/null
+++ b/cmds/screenrecord/screenrecord.h
@@ -0,0 +1,23 @@
+/*
+ * Copyright 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef SCREENRECORD_SCREENRECORD_H
+#define SCREENRECORD_SCREENRECORD_H
+
+#define kVersionMajor 1
+#define kVersionMinor 2
+
+#endif /*SCREENRECORD_SCREENRECORD_H*/

cmds/stagefright/SimplePlayer.cpp

diff --git a/cmds/stagefright/SimplePlayer.cpp b/cmds/stagefright/SimplePlayer.cpp
index 5d2d721..1b2f792 100644
--- a/cmds/stagefright/SimplePlayer.cpp
+++ b/cmds/stagefright/SimplePlayer.cpp
@@ -23,6 +23,7 @@
 #include <gui/Surface.h>
 #include <media/AudioTrack.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>
@@ -275,7 +276,8 @@
 
     mExtractor = new NuMediaExtractor;
 
-    status_t err = mExtractor->setDataSource(mPath.c_str());
+    status_t err = mExtractor->setDataSource(
+            NULL /* httpService */, mPath.c_str());
 
     if (err != OK) {
         mExtractor.clear();

cmds/stagefright/SineSource.cpp

diff --git a/cmds/stagefright/SineSource.cpp b/cmds/stagefright/SineSource.cpp
index 14b4306..587077a 100644
--- a/cmds/stagefright/SineSource.cpp
+++ b/cmds/stagefright/SineSource.cpp
@@ -24,7 +24,7 @@
     }
 }
 
-status_t SineSource::start(MetaData *params) {
+status_t SineSource::start(MetaData * /* params */) {
     CHECK(!mStarted);
 
     mGroup = new MediaBufferGroup;
@@ -58,7 +58,7 @@
 }
 
 status_t SineSource::read(
-        MediaBuffer **out, const ReadOptions *options) {
+        MediaBuffer **out, const ReadOptions * /* options */) {
     *out = NULL;
 
     MediaBuffer *buffer;

cmds/stagefright/codec.cpp

diff --git a/cmds/stagefright/codec.cpp b/cmds/stagefright/codec.cpp
index fdfefdf..fd02bcc 100644
--- a/cmds/stagefright/codec.cpp
+++ b/cmds/stagefright/codec.cpp
@@ -16,6 +16,7 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "codec"
+#include <inttypes.h>
 #include <utils/Log.h>
 
 #include "SimplePlayer.h"
@@ -23,6 +24,7 @@
 #include <binder/IServiceManager.h>
 #include <binder/ProcessState.h>
 #include <media/ICrypto.h>
+#include <media/IMediaHTTPService.h>
 #include <media/IMediaPlayerService.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -75,7 +77,7 @@
     static int64_t kTimeout = 500ll;
 
     sp<NuMediaExtractor> extractor = new NuMediaExtractor;
-    if (extractor->setDataSource(path) != OK) {
+    if (extractor->setDataSource(NULL /* httpService */, path) != OK) {
         fprintf(stderr, "unable to instantiate extractor.\n");
         return 1;
     }
@@ -291,13 +293,13 @@
         CHECK_EQ((status_t)OK, state->mCodec->release());
 
         if (state->mIsAudio) {
-            printf("track %d: %lld bytes received. %.2f KB/sec\n",
+            printf("track %zu: %" PRId64 " bytes received. %.2f KB/sec\n",
                    i,
                    state->mNumBytesDecoded,
                    state->mNumBytesDecoded * 1E6 / 1024 / elapsedTimeUs);
         } else {
-            printf("track %d: %lld frames decoded, %.2f fps. %lld bytes "
-                   "received. %.2f KB/sec\n",
+            printf("track %zu: %" PRId64 " frames decoded, %.2f fps. %" PRId64
+                    " bytes received. %.2f KB/sec\n",
                    i,
                    state->mNumBuffersDecoded,
                    state->mNumBuffersDecoded * 1E6 / elapsedTimeUs,

cmds/stagefright/muxer.cpp

diff --git a/cmds/stagefright/muxer.cpp b/cmds/stagefright/muxer.cpp
index cca33e0..f4a33e8 100644
--- a/cmds/stagefright/muxer.cpp
+++ b/cmds/stagefright/muxer.cpp
@@ -16,9 +16,11 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "muxer"
+#include <inttypes.h>
 #include <utils/Log.h>
 
 #include <binder/ProcessState.h>
+#include <media/IMediaHTTPService.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/ALooper.h>
@@ -58,7 +60,7 @@
         int trimEndTimeMs,
         int rotationDegrees) {
     sp<NuMediaExtractor> extractor = new NuMediaExtractor;
-    if (extractor->setDataSource(path) != OK) {
+    if (extractor->setDataSource(NULL /* httpService */, path) != OK) {
         fprintf(stderr, "unable to instantiate extractor. %s\n", path);
         return 1;
     }
@@ -198,7 +200,7 @@
     trackIndexMap.clear();
 
     int64_t elapsedTimeUs = ALooper::GetNowUs() - muxerStartTimeUs;
-    fprintf(stderr, "SUCCESS: muxer generate the video in %lld ms\n",
+    fprintf(stderr, "SUCCESS: muxer generate the video in %" PRId64 " ms\n",
             elapsedTimeUs / 1000);
 
     return 0;

cmds/stagefright/record.cpp

diff --git a/cmds/stagefright/record.cpp b/cmds/stagefright/record.cpp
index b7a40c2..fdc352e 100644
--- a/cmds/stagefright/record.cpp
+++ b/cmds/stagefright/record.cpp
@@ -296,7 +296,7 @@
 }
 #else
 
-int main(int argc, char **argv) {
+int main(int /* argc */, char ** /* argv */) {
     android::ProcessState::self()->startThreadPool();
 
     OMXClient client;

cmds/stagefright/recordvideo.cpp

diff --git a/cmds/stagefright/recordvideo.cpp b/cmds/stagefright/recordvideo.cpp
index c30c122..1d267f9 100644
--- a/cmds/stagefright/recordvideo.cpp
+++ b/cmds/stagefright/recordvideo.cpp
@@ -16,6 +16,7 @@
 
 #include "SineSource.h"
 
+#include <inttypes.h>
 #include <binder/ProcessState.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/AudioPlayer.h>
@@ -312,7 +313,7 @@
         fprintf(stderr, "record failed: %d\n", err);
         return 1;
     }
-    fprintf(stderr, "encoding %d frames in %lld us\n", nFrames, (end-start)/1000);
+    fprintf(stderr, "encoding %d frames in %" PRId64 " us\n", nFrames, (end-start)/1000);
     fprintf(stderr, "encoding speed is: %.2f fps\n", (nFrames * 1E9) / (end-start));
     return 0;
 }

cmds/stagefright/sf2.cpp

diff --git a/cmds/stagefright/sf2.cpp b/cmds/stagefright/sf2.cpp
index c817443..3c0c7ec 100644
--- a/cmds/stagefright/sf2.cpp
+++ b/cmds/stagefright/sf2.cpp
@@ -16,10 +16,15 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "sf2"
+#include <inttypes.h>
 #include <utils/Log.h>
 
+#include <signal.h>
+
 #include <binder/ProcessState.h>
 
+#include <media/IMediaHTTPService.h>
+
 #include <media/stagefright/foundation/hexdump.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -42,6 +47,18 @@
 
 using namespace android;
 
+volatile static bool ctrlc = false;
+
+static sighandler_t oldhandler = NULL;
+
+static void mysighandler(int signum) {
+    if (signum == SIGINT) {
+        ctrlc = true;
+        return;
+    }
+    oldhandler(signum);
+}
+
 struct Controller : public AHandler {
     Controller(const char *uri, bool decodeAudio,
                const sp<Surface> &surface, bool renderToSurface)
@@ -62,7 +79,30 @@
     virtual ~Controller() {
     }
 
+    virtual void printStatistics() {
+        int64_t delayUs = ALooper::GetNowUs() - mStartTimeUs;
+
+        if (mDecodeAudio) {
+            printf("%" PRId64 " bytes received. %.2f KB/sec\n",
+            mTotalBytesReceived,
+            mTotalBytesReceived * 1E6 / 1024 / delayUs);
+        } else {
+            printf("%d frames decoded, %.2f fps. %" PRId64 " bytes "
+                    "received. %.2f KB/sec\n",
+            mNumOutputBuffersReceived,
+            mNumOutputBuffersReceived * 1E6 / delayUs,
+            mTotalBytesReceived,
+            mTotalBytesReceived * 1E6 / 1024 / delayUs);
+        }
+    }
+
     virtual void onMessageReceived(const sp<AMessage> &msg) {
+        if (ctrlc) {
+            printf("\n");
+            printStatistics();
+            (new AMessage(kWhatStop, id()))->post();
+            ctrlc = false;
+        }
         switch (msg->what()) {
             case kWhatStart:
             {
@@ -75,7 +115,8 @@
 #endif
 
                 sp<DataSource> dataSource =
-                    DataSource::CreateFromURI(mURI.c_str());
+                    DataSource::CreateFromURI(
+                            NULL /* httpService */, mURI.c_str());
 
                 sp<MediaExtractor> extractor =
                     MediaExtractor::Create(dataSource);
@@ -98,7 +139,10 @@
                         break;
                     }
                 }
-                CHECK(mSource != NULL);
+                if (mSource == NULL) {
+                    printf("no %s track found\n", mDecodeAudio ? "audio" : "video");
+                    exit (1);
+                }
 
                 CHECK_EQ(mSource->start(), (status_t)OK);
 
@@ -180,21 +224,7 @@
                         || what == ACodec::kWhatError) {
                     printf((what == ACodec::kWhatEOS) ? "$\n" : "E\n");
 
-                    int64_t delayUs = ALooper::GetNowUs() - mStartTimeUs;
-
-                    if (mDecodeAudio) {
-                        printf("%lld bytes received. %.2f KB/sec\n",
-                               mTotalBytesReceived,
-                               mTotalBytesReceived * 1E6 / 1024 / delayUs);
-                    } else {
-                        printf("%d frames decoded, %.2f fps. %lld bytes "
-                               "received. %.2f KB/sec\n",
-                               mNumOutputBuffersReceived,
-                               mNumOutputBuffersReceived * 1E6 / delayUs,
-                               mTotalBytesReceived,
-                               mTotalBytesReceived * 1E6 / 1024 / delayUs);
-                    }
-
+                    printStatistics();
                     (new AMessage(kWhatStop, id()))->post();
                 } else if (what == ACodec::kWhatFlushCompleted) {
                     mSeekState = SEEK_FLUSH_COMPLETED;
@@ -638,6 +668,8 @@
 
     looper->registerHandler(controller);
 
+    signal(SIGINT, mysighandler);
+
     controller->startAsync();
 
     CHECK_EQ(looper->start(true /* runOnCallingThread */), (status_t)OK);

cmds/stagefright/stagefright.cpp

diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index 030bf1b..daaea27 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -22,12 +22,14 @@
 
 #include <stdlib.h>
 #include <string.h>
+#include <inttypes.h>
 
 #include "jpeg.h"
 #include "SineSource.h"
 
 #include <binder/IServiceManager.h>
 #include <binder/ProcessState.h>
+#include <media/IMediaHTTPService.h>
 #include <media/IMediaPlayerService.h>
 #include <media/stagefright/foundation/ALooper.h>
 #include "include/NuCachedSource2.h"
@@ -89,8 +91,8 @@
     int64_t minUs = decodeTimesUs->itemAt(0);
     int64_t maxUs = decodeTimesUs->itemAt(n - 1);
 
-    printf("min decode time %lld us (%.2f secs)\n", minUs, minUs / 1E6);
-    printf("max decode time %lld us (%.2f secs)\n", maxUs, maxUs / 1E6);
+    printf("min decode time %" PRId64 " us (%.2f secs)\n", minUs, minUs / 1E6);
+    printf("max decode time %" PRId64 " us (%.2f secs)\n", maxUs, maxUs / 1E6);
 
     size_t counts[100];
     for (size_t i = 0; i < 100; ++i) {
@@ -110,7 +112,7 @@
         int64_t slotUs = minUs + (i * (maxUs - minUs) / 100);
 
         double fps = 1E6 / slotUs;
-        printf("[%.2f fps]: %d\n", fps, counts[i]);
+        printf("[%.2f fps]: %zu\n", fps, counts[i]);
     }
 }
 
@@ -262,7 +264,7 @@
                     }
                 }
 
-                printf("buffer has timestamp %lld us (%.2f secs)\n",
+                printf("buffer has timestamp %" PRId64 " us (%.2f secs)\n",
                        timestampUs, timestampUs / 1E6);
 
                 buffer->release();
@@ -285,7 +287,7 @@
                 seekTimeUs = (rand() * (float)durationUs) / RAND_MAX;
                 options.setSeekTo(seekTimeUs);
 
-                printf("seeking to %lld us (%.2f secs)\n",
+                printf("seeking to %" PRId64 " us (%.2f secs)\n",
                        seekTimeUs, seekTimeUs / 1E6);
             }
         }
@@ -388,7 +390,7 @@
         // sizes may be different across decoders.
         printf("avg. %.2f KB/sec\n", totalBytes / 1024 * 1E6 / delay);
 
-        printf("decoded a total of %lld bytes\n", totalBytes);
+        printf("decoded a total of %" PRId64 " bytes\n", totalBytes);
     }
 }
 
@@ -574,7 +576,8 @@
             int64_t timeUs;
             CHECK(buffer->meta_data()->findInt64(kKeyTime, &timeUs));
 
-            printf("%lld\t%lld\t%lld\n", seekTimeUs, timeUs, seekTimeUs - timeUs);
+            printf("%" PRId64 "\t%" PRId64 "\t%" PRId64 "\n",
+                   seekTimeUs, timeUs, seekTimeUs - timeUs);
 
             buffer->release();
             buffer = NULL;
@@ -958,7 +961,8 @@
 
         const char *filename = argv[k];
 
-        sp<DataSource> dataSource = DataSource::CreateFromURI(filename);
+        sp<DataSource> dataSource =
+            DataSource::CreateFromURI(NULL /* httpService */, filename);
 
         if (strncasecmp(filename, "sine:", 5) && dataSource == NULL) {
             fprintf(stderr, "Unable to create data source.\n");
@@ -1071,7 +1075,7 @@
 
                 int64_t thumbTimeUs;
                 if (meta->findInt64(kKeyThumbnailTime, &thumbTimeUs)) {
-                    printf("thumbnailTime: %lld us (%.2f secs)\n",
+                    printf("thumbnailTime: %" PRId64 " us (%.2f secs)\n",
                            thumbTimeUs, thumbTimeUs / 1E6);
                 }
 

cmds/stagefright/stream.cpp

diff --git a/cmds/stagefright/stream.cpp b/cmds/stagefright/stream.cpp
index dba67a9..b2abc0f 100644
--- a/cmds/stagefright/stream.cpp
+++ b/cmds/stagefright/stream.cpp
@@ -21,6 +21,7 @@
 #include <binder/ProcessState.h>
 #include <cutils/properties.h> // for property_get
 
+#include <media/IMediaHTTPService.h>
 #include <media/IStreamSource.h>
 #include <media/mediaplayer.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -159,7 +160,9 @@
 MyConvertingStreamSource::MyConvertingStreamSource(const char *filename)
     : mCurrentBufferIndex(-1),
       mCurrentBufferOffset(0) {
-    sp<DataSource> dataSource = DataSource::CreateFromURI(filename);
+    sp<DataSource> dataSource =
+        DataSource::CreateFromURI(NULL /* httpService */, filename);
+
     CHECK(dataSource != NULL);
 
     sp<MediaExtractor> extractor = MediaExtractor::Create(dataSource);

drm/libdrmframework/plugins/forward-lock/FwdLockEngine/Android.mk

diff --git a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/Android.mk b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/Android.mk
index e251f82..48b0afe 100644
--- a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/Android.mk
+++ b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/Android.mk
@@ -61,7 +61,7 @@
     $(LOCAL_PATH)/include \
     external/openssl/include
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/drm
+LOCAL_MODULE_RELATIVE_PATH := drm
 
 LOCAL_MODULE_TAGS := optional
 

drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp

diff --git a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp
index 234aef2..f400732 100644
--- a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp
+++ b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp
@@ -316,6 +316,7 @@
 
     if (-1 < fileDesc) {
         if (FwdLockFile_attach(fileDesc) < 0) {
+            close(fileDesc);
             return mimeString;
         }
         const char* pMimeType = FwdLockFile_GetContentType(fileDesc);

drm/mediadrm/plugins/mock/Android.mk

diff --git a/drm/mediadrm/plugins/mock/Android.mk b/drm/mediadrm/plugins/mock/Android.mk
index ada23a2..26c245b 100644
--- a/drm/mediadrm/plugins/mock/Android.mk
+++ b/drm/mediadrm/plugins/mock/Android.mk
@@ -21,7 +21,8 @@
 
 LOCAL_MODULE := libmockdrmcryptoplugin
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_VENDOR_SHARED_LIBRARIES)/mediadrm
+LOCAL_PROPRIETARY_MODULE := true
+LOCAL_MODULE_RELATIVE_PATH := mediadrm
 
 LOCAL_SHARED_LIBRARIES := \
     libutils liblog

drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp

diff --git a/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp b/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp
index f2cadf7..69fa7a0fc 100644
--- a/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp
+++ b/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp
@@ -695,7 +695,7 @@
     {
         String8 result;
         for (size_t i = 0; i < numSubSamples; i++) {
-            result.appendFormat("[%d] {clear:%d, encrypted:%d} ", i,
+            result.appendFormat("[%zu] {clear:%zu, encrypted:%zu} ", i,
                                 subSamples[i].mNumBytesOfClearData,
                                 subSamples[i].mNumBytesOfEncryptedData);
         }

include/camera/CameraMetadata.h

diff --git a/include/camera/CameraMetadata.h b/include/camera/CameraMetadata.h
index fe2bd19..1254d3c 100644
--- a/include/camera/CameraMetadata.h
+++ b/include/camera/CameraMetadata.h
@@ -99,6 +99,11 @@
     status_t append(const CameraMetadata &other);
 
     /**
+     * Append metadata from a raw camera_metadata buffer
+     */
+    status_t append(const camera_metadata* other);
+
+    /**
      * Number of metadata entries.
      */
     size_t entryCount() const;

include/camera/IProCameraCallbacks.h

diff --git a/include/camera/IProCameraCallbacks.h b/include/camera/IProCameraCallbacks.h
index c774698..e8abb89 100644
--- a/include/camera/IProCameraCallbacks.h
+++ b/include/camera/IProCameraCallbacks.h
@@ -51,7 +51,7 @@
     /** Missing by design: implementation is client-side in ProCamera.cpp **/
     // virtual void onBufferReceived(int streamId,
     //                               const CpuConsumer::LockedBufer& buf);
-    virtual void            onResultReceived(int32_t frameId,
+    virtual void            onResultReceived(int32_t requestId,
                                              camera_metadata* result) = 0;
 };
 

include/camera/ProCamera.h

diff --git a/include/camera/ProCamera.h b/include/camera/ProCamera.h
index d9ee662..83a3028 100644
--- a/include/camera/ProCamera.h
+++ b/include/camera/ProCamera.h
@@ -252,7 +252,7 @@
     virtual void        onLockStatusChanged(
                                 IProCameraCallbacks::LockStatus newLockStatus);
 
-    virtual void        onResultReceived(int32_t frameId,
+    virtual void        onResultReceived(int32_t requestId,
                                          camera_metadata* result);
 private:
     ProCamera(int cameraId);

include/camera/camera2/ICameraDeviceCallbacks.h

diff --git a/include/camera/camera2/ICameraDeviceCallbacks.h b/include/camera/camera2/ICameraDeviceCallbacks.h
index 041fa65..8dac4f2 100644
--- a/include/camera/camera2/ICameraDeviceCallbacks.h
+++ b/include/camera/camera2/ICameraDeviceCallbacks.h
@@ -35,13 +35,27 @@
 public:
     DECLARE_META_INTERFACE(CameraDeviceCallbacks);
 
-    // One way
-    virtual void            notifyCallback(int32_t msgType,
-                                           int32_t ext1,
-                                           int32_t ext2) = 0;
+    /**
+     * Error codes for CAMERA_MSG_ERROR
+     */
+    enum CameraErrorCode {
+        ERROR_CAMERA_DISCONNECTED = 0,
+        ERROR_CAMERA_DEVICE = 1,
+        ERROR_CAMERA_SERVICE = 2
+    };
 
     // One way
-    virtual void            onResultReceived(int32_t frameId,
+    virtual void            onDeviceError(CameraErrorCode errorCode) = 0;
+
+    // One way
+    virtual void            onDeviceIdle() = 0;
+
+    // One way
+    virtual void            onCaptureStarted(int32_t requestId,
+                                             int64_t timestamp) = 0;
+
+    // One way
+    virtual void            onResultReceived(int32_t requestId,
                                              const CameraMetadata& result) = 0;
 };
 

include/media/AudioBufferProvider.h

diff --git a/include/media/AudioBufferProvider.h b/include/media/AudioBufferProvider.h
index ef392f0..7be449c 100644
--- a/include/media/AudioBufferProvider.h
+++ b/include/media/AudioBufferProvider.h
@@ -61,6 +61,17 @@
     //  buffer->frameCount  0
     virtual status_t getNextBuffer(Buffer* buffer, int64_t pts = kInvalidPTS) = 0;
 
+    // Release (a portion of) the buffer previously obtained by getNextBuffer().
+    // It is permissible to call releaseBuffer() multiple times per getNextBuffer().
+    // On entry:
+    //  buffer->frameCount  number of frames to release, must be <= number of frames
+    //                      obtained but not yet released
+    //  buffer->raw         unused
+    // On return:
+    //  buffer->frameCount  0; implementation MUST set to zero
+    //  buffer->raw         undefined; implementation is PERMITTED to set to any value,
+    //                      so if caller needs to continue using this buffer it must
+    //                      keep track of the pointer itself
     virtual void releaseBuffer(Buffer* buffer) = 0;
 };
 

include/media/AudioEffect.h

diff --git a/include/media/AudioEffect.h b/include/media/AudioEffect.h
index 05d834d..f3024b7 100644
--- a/include/media/AudioEffect.h
+++ b/include/media/AudioEffect.h
@@ -36,7 +36,7 @@
 
 // ----------------------------------------------------------------------------
 
-class effect_param_cblk_t;
+struct effect_param_cblk_t;
 
 // ----------------------------------------------------------------------------
 

include/media/AudioRecord.h

diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index fb47448..b3c44a8 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -26,7 +26,7 @@
 
 // ----------------------------------------------------------------------------
 
-class audio_track_cblk_t;
+struct audio_track_cblk_t;
 class AudioRecordClientProxy;
 
 // ----------------------------------------------------------------------------
@@ -39,8 +39,12 @@
      * Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*.
      */
     enum event_type {
-        EVENT_MORE_DATA = 0,        // Request to read more data from PCM buffer.
-        EVENT_OVERRUN = 1,          // PCM buffer overrun occurred.
+        EVENT_MORE_DATA = 0,        // Request to read available data from buffer.
+                                    // If this event is delivered but the callback handler
+                                    // does not want to read the available data, the handler must
+                                    // explicitly
+                                    // ignore the event by setting frameCount to zero.
+        EVENT_OVERRUN = 1,          // Buffer overrun occurred.
         EVENT_MARKER = 2,           // Record head is at the specified marker position
                                     // (See setMarkerPosition()).
         EVENT_NEW_POS = 3,          // Record head is at a new position
@@ -63,6 +67,7 @@
                                     // (currently ignored but will make the primary field in future)
 
         size_t      size;           // input/output in bytes == frameCount * frameSize
+                                    // on output is the number of bytes actually drained
                                     // 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
@@ -76,7 +81,7 @@
 
     /* As a convenience, if a callback is supplied, a handler thread
      * is automatically created with the appropriate priority. This thread
-     * invokes the callback when a new buffer becomes ready or various conditions occur.
+     * invokes the callback when a new buffer becomes available or various conditions occur.
      * Parameters:
      *
      * event:   type of event notified (see enum AudioRecord::event_type).
@@ -99,6 +104,8 @@
      *  - NO_ERROR: successful operation
      *  - NO_INIT: audio server or audio hardware not initialized
      *  - BAD_VALUE: unsupported configuration
+     * frameCount is guaranteed to be non-zero if status is NO_ERROR,
+     * and is undefined otherwise.
      */
 
      static status_t getMinFrameCount(size_t* frameCount,
@@ -109,7 +116,7 @@
     /* How data is transferred from AudioRecord
      */
     enum transfer_type {
-        TRANSFER_DEFAULT,   // not specified explicitly; determine from other parameters
+        TRANSFER_DEFAULT,   // not specified explicitly; determine from the other parameters
         TRANSFER_CALLBACK,  // callback EVENT_MORE_DATA
         TRANSFER_OBTAIN,    // FIXME deprecated: call obtainBuffer() and releaseBuffer()
         TRANSFER_SYNC,      // synchronous read()
@@ -137,7 +144,7 @@
      *                     be larger if the requested size is not compatible with current audio HAL
      *                     latency.  Zero means to use a default value.
      * cbf:                Callback function. If not null, this function is called periodically
-     *                     to consume new PCM data and inform of marker, position updates, etc.
+     *                     to consume new data and inform of marker, position updates, etc.
      * user:               Context for use by the callback receiver.
      * notificationFrames: The callback function is called each time notificationFrames PCM
      *                     frames are ready in record track output buffer.
@@ -151,11 +158,11 @@
                                     uint32_t sampleRate,
                                     audio_format_t format,
                                     audio_channel_mask_t channelMask,
-                                    int frameCount      = 0,
+                                    size_t frameCount = 0,
                                     callback_t cbf = NULL,
                                     void* user = NULL,
-                                    int notificationFrames = 0,
-                                    int sessionId = 0,
+                                    uint32_t notificationFrames = 0,
+                                    int sessionId = AUDIO_SESSION_ALLOCATE,
                                     transfer_type transferType = TRANSFER_DEFAULT,
                                     audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE);
 
@@ -171,9 +178,10 @@
      * Returned status (from utils/Errors.h) can be:
      *  - NO_ERROR: successful intialization
      *  - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use
-     *  - BAD_VALUE: invalid parameter (channels, format, sampleRate...)
+     *  - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...)
      *  - NO_INIT: audio server or audio hardware not initialized
      *  - PERMISSION_DENIED: recording is not allowed for the requesting process
+     * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord.
      *
      * Parameters not listed in the AudioRecord constructors above:
      *
@@ -183,16 +191,16 @@
                             uint32_t sampleRate,
                             audio_format_t format,
                             audio_channel_mask_t channelMask,
-                            int frameCount      = 0,
+                            size_t frameCount = 0,
                             callback_t cbf = NULL,
                             void* user = NULL,
-                            int notificationFrames = 0,
+                            uint32_t notificationFrames = 0,
                             bool threadCanCallJava = false,
-                            int sessionId = 0,
+                            int sessionId = AUDIO_SESSION_ALLOCATE,
                             transfer_type transferType = TRANSFER_DEFAULT,
                             audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE);
 
-    /* Result of constructing the AudioRecord. This must be checked
+    /* Result of constructing the AudioRecord. This must be checked for successful initialization
      * before using any AudioRecord API (except for set()), because using
      * an uninitialized AudioRecord produces undefined results.
      * See set() method above for possible return codes.
@@ -221,7 +229,7 @@
             status_t    start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
                               int triggerSession = 0);
 
-    /* Stop a track. If set, the callback will cease being called.  Note that obtainBuffer() still
+    /* Stop a track.  The callback will cease being called.  Note that obtainBuffer() still
      * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK.
      */
             void        stop();
@@ -236,7 +244,7 @@
      * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition
      * with marker == 0 cancels marker notification callback.
      * To set a marker at a position which would compute as 0,
-     * a workaround is to the set the marker at a nearby position such as ~0 or 1.
+     * a workaround is to set the marker at a nearby position such as ~0 or 1.
      * If the AudioRecord has been opened with no callback function associated,
      * the operation will fail.
      *
@@ -378,8 +386,10 @@
      * returning the current value by this function call.  Such loss typically occurs when the
      * user space process is blocked longer than the capacity of audio driver buffers.
      * Units: the number of input audio frames.
+     * FIXME The side-effect of resetting the counter may be incompatible with multi-client.
+     * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects.
      */
-            unsigned int  getInputFramesLost() const;
+            uint32_t    getInputFramesLost() const;
 
 private:
     /* copying audio record objects is not allowed */
@@ -412,6 +422,7 @@
         bool                mPaused;    // whether thread is requested to pause at next loop entry
         bool                mPausedInt; // whether thread internally requests pause
         nsecs_t             mPausedNs;  // if mPausedInt then associated timeout, otherwise ignored
+        bool                mIgnoreNextPausedInt;   // whether to ignore next mPausedInt request
     };
 
             // body of AudioRecordThread::threadLoop()
@@ -422,9 +433,10 @@
             //      NS_INACTIVE inactive so don't run again until re-started
             //      NS_NEVER    never again
             static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3;
-            nsecs_t processAudioBuffer(const sp<AudioRecordThread>& thread);
+            nsecs_t processAudioBuffer();
 
             // caller must hold lock on mLock for all _l methods
+
             status_t openRecord_l(size_t epoch);
 
             // FIXME enum is faster than strcmp() for parameter 'from'
@@ -452,7 +464,7 @@
     // These are private to processAudioBuffer(), and are not protected by a lock
     uint32_t                mRemainingFrames;       // number of frames to request in obtainBuffer()
     bool                    mRetryOnPartialBuffer;  // sleep and retry after partial obtainBuffer()
-    int                     mObservedSequence;      // last observed value of mSequence
+    uint32_t                mObservedSequence;      // last observed value of mSequence
 
     uint32_t                mMarkerPosition;    // in wrapping (overflow) frame units
     bool                    mMarkerReached;
@@ -461,9 +473,13 @@
 
     status_t                mStatus;
 
+    size_t                  mFrameCount;            // corresponds to current IAudioRecord, value is
+                                                    // reported back by AudioFlinger to the client
+    size_t                  mReqFrameCount;         // frame count to request the first or next time
+                                                    // a new IAudioRecord is needed, non-decreasing
+
     // constant after constructor or set()
     uint32_t                mSampleRate;
-    size_t                  mFrameCount;
     audio_format_t          mFormat;
     uint32_t                mChannelCount;
     size_t                  mFrameSize;         // app-level frame size == AudioFlinger frame size
@@ -474,12 +490,11 @@
     int                     mSessionId;
     transfer_type           mTransfer;
 
-    audio_io_handle_t       mInput;             // returned by AudioSystem::getInput()
-
-    // may be changed if IAudioRecord object is re-created
+    // Next 4 fields may be changed if IAudioRecord is re-created, but always != 0
     sp<IAudioRecord>        mAudioRecord;
     sp<IMemory>             mCblkMemory;
     audio_track_cblk_t*     mCblk;              // re-load after mLock.unlock()
+    audio_io_handle_t       mInput;             // returned by AudioSystem::getInput()
 
     int                     mPreviousPriority;  // before start()
     SchedPolicy             mPreviousSchedulingGroup;

include/media/AudioSystem.h

diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index b96b8a1..28fdfd4 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -67,20 +67,24 @@
 
     // returns true in *state if tracks are active on the specified stream or have been active
     // in the past inPastMs milliseconds
-    static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs = 0);
+    static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs);
     // returns true in *state if tracks are active for what qualifies as remote playback
     // on the specified stream or have been active in the past inPastMs milliseconds. Remote
     // playback isn't mutually exclusive with local playback.
     static status_t isStreamActiveRemotely(audio_stream_type_t stream, bool *state,
-            uint32_t inPastMs = 0);
+            uint32_t inPastMs);
     // returns true in *state if a recorder is currently recording with the specified source
     static status_t isSourceActive(audio_source_t source, bool *state);
 
     // set/get audio hardware parameters. The function accepts a list of parameters
     // key value pairs in the form: key1=value1;key2=value2;...
     // Some keys are reserved for standard parameters (See AudioParameter class).
+    // The versions with audio_io_handle_t are intended for internal media framework use only.
     static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
     static String8  getParameters(audio_io_handle_t ioHandle, const String8& keys);
+    // The versions without audio_io_handle_t are intended for JNI.
+    static status_t setParameters(const String8& keyValuePairs);
+    static String8  getParameters(const String8& keys);
 
     static void setErrorCallback(audio_error_callback cb);
 
@@ -90,12 +94,14 @@
     static float linearToLog(int volume);
     static int logToLinear(float volume);
 
+    // Returned samplingRate and frameCount output values are guaranteed
+    // to be non-zero if status == NO_ERROR
     static status_t getOutputSamplingRate(uint32_t* samplingRate,
-            audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+            audio_stream_type_t stream);
     static status_t getOutputFrameCount(size_t* frameCount,
-            audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+            audio_stream_type_t stream);
     static status_t getOutputLatency(uint32_t* latency,
-            audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+            audio_stream_type_t stream);
     static status_t getSamplingRate(audio_io_handle_t output,
                                           audio_stream_type_t streamType,
                                           uint32_t* samplingRate);
@@ -112,6 +118,7 @@
 
     static bool routedToA2dpOutput(audio_stream_type_t streamType);
 
+    // return status NO_ERROR implies *buffSize > 0
     static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
         audio_channel_mask_t channelMask, size_t* buffSize);
 
@@ -127,16 +134,16 @@
     // NOTE: this feature is not supported on all hardware platforms and it is
     // necessary to check returned status before using the returned values.
     static status_t getRenderPosition(audio_io_handle_t output,
-                                      size_t *halFrames,
-                                      size_t *dspFrames,
+                                      uint32_t *halFrames,
+                                      uint32_t *dspFrames,
                                       audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
 
     // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid
-    static size_t getInputFramesLost(audio_io_handle_t ioHandle);
+    static uint32_t getInputFramesLost(audio_io_handle_t ioHandle);
 
     static int newAudioSessionId();
-    static void acquireAudioSessionId(int audioSession);
-    static void releaseAudioSessionId(int audioSession);
+    static void acquireAudioSessionId(int audioSession, pid_t pid);
+    static void releaseAudioSessionId(int audioSession, pid_t pid);
 
     // types of io configuration change events received with ioConfigChanged()
     enum io_config_event {
@@ -194,24 +201,32 @@
     static status_t setPhoneState(audio_mode_t state);
     static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
     static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
+
+    // Client must successfully hand off the handle reference to AudioFlinger via createTrack(),
+    // or release it with releaseOutput().
     static audio_io_handle_t getOutput(audio_stream_type_t stream,
                                         uint32_t samplingRate = 0,
                                         audio_format_t format = AUDIO_FORMAT_DEFAULT,
                                         audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
                                         audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
                                         const audio_offload_info_t *offloadInfo = NULL);
+
     static status_t startOutput(audio_io_handle_t output,
                                 audio_stream_type_t stream,
-                                int session = 0);
+                                int session);
     static status_t stopOutput(audio_io_handle_t output,
                                audio_stream_type_t stream,
-                               int session = 0);
+                               int session);
     static void releaseOutput(audio_io_handle_t output);
+
+    // Client must successfully hand off the handle reference to AudioFlinger via openRecord(),
+    // or release it with releaseInput().
     static audio_io_handle_t getInput(audio_source_t inputSource,
-                                    uint32_t samplingRate = 0,
-                                    audio_format_t format = AUDIO_FORMAT_DEFAULT,
-                                    audio_channel_mask_t channelMask = AUDIO_CHANNEL_IN_MONO,
-                                    int sessionId = 0);
+                                    uint32_t samplingRate,
+                                    audio_format_t format,
+                                    audio_channel_mask_t channelMask,
+                                    int sessionId);
+
     static status_t startInput(audio_io_handle_t input);
     static status_t stopInput(audio_io_handle_t input);
     static void releaseInput(audio_io_handle_t input);

include/media/AudioTimestamp.h

diff --git a/include/media/AudioTimestamp.h b/include/media/AudioTimestamp.h
index c29c7e5..99e9c3e 100644
--- a/include/media/AudioTimestamp.h
+++ b/include/media/AudioTimestamp.h
@@ -19,6 +19,8 @@
 
 #include <time.h>
 
+namespace android {
+
 class AudioTimestamp {
 public:
     AudioTimestamp() : mPosition(0) {
@@ -30,4 +32,6 @@
     struct timespec mTime;     // corresponding CLOCK_MONOTONIC when frame is expected to present
 };
 
+}   // namespace
+
 #endif  // ANDROID_AUDIO_TIMESTAMP_H

include/media/AudioTrack.h

diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index f863e35..7d23d02 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -27,7 +27,7 @@
 
 // ----------------------------------------------------------------------------
 
-class audio_track_cblk_t;
+struct audio_track_cblk_t;
 class AudioTrackClientProxy;
 class StaticAudioTrackClientProxy;
 
@@ -123,6 +123,8 @@
      *  - NO_ERROR: successful operation
      *  - NO_INIT: audio server or audio hardware not initialized
      *  - BAD_VALUE: unsupported configuration
+     * frameCount is guaranteed to be non-zero if status is NO_ERROR,
+     * and is undefined otherwise.
      */
 
     static status_t getMinFrameCount(size_t* frameCount,
@@ -158,7 +160,7 @@
      * sampleRate:         Data source sampling rate in Hz.
      * format:             Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
      *                     16 bits per sample).
-     * channelMask:        Channel mask.
+     * 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
      *                     latency of the track. The actual size selected by the AudioTrack could be
@@ -180,14 +182,16 @@
                                     uint32_t sampleRate,
                                     audio_format_t format,
                                     audio_channel_mask_t,
-                                    int frameCount       = 0,
+                                    size_t frameCount    = 0,
                                     audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
                                     callback_t cbf       = NULL,
                                     void* user           = NULL,
-                                    int notificationFrames = 0,
-                                    int sessionId        = 0,
+                                    uint32_t notificationFrames = 0,
+                                    int sessionId        = AUDIO_SESSION_ALLOCATE,
                                     transfer_type transferType = TRANSFER_DEFAULT,
-                                    const audio_offload_info_t *offloadInfo = NULL);
+                                    const audio_offload_info_t *offloadInfo = NULL,
+                                    int uid = -1,
+                                    pid_t pid = -1);
 
     /* Creates an audio track and registers it with AudioFlinger.
      * With this constructor, the track is configured for static buffer mode.
@@ -208,10 +212,12 @@
                                     audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
                                     callback_t cbf      = NULL,
                                     void* user          = NULL,
-                                    int notificationFrames = 0,
-                                    int sessionId       = 0,
+                                    uint32_t notificationFrames = 0,
+                                    int sessionId       = AUDIO_SESSION_ALLOCATE,
                                     transfer_type transferType = TRANSFER_DEFAULT,
-                                    const audio_offload_info_t *offloadInfo = NULL);
+                                    const audio_offload_info_t *offloadInfo = NULL,
+                                    int uid = -1,
+                                    pid_t pid = -1);
 
     /* Terminates the AudioTrack and unregisters it from AudioFlinger.
      * Also destroys all resources associated with the AudioTrack.
@@ -239,16 +245,18 @@
                             uint32_t sampleRate,
                             audio_format_t format,
                             audio_channel_mask_t channelMask,
-                            int frameCount      = 0,
+                            size_t frameCount   = 0,
                             audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
                             callback_t cbf      = NULL,
                             void* user          = NULL,
-                            int notificationFrames = 0,
+                            uint32_t notificationFrames = 0,
                             const sp<IMemory>& sharedBuffer = 0,
                             bool threadCanCallJava = false,
-                            int sessionId       = 0,
+                            int sessionId       = AUDIO_SESSION_ALLOCATE,
                             transfer_type transferType = TRANSFER_DEFAULT,
-                            const audio_offload_info_t *offloadInfo = NULL);
+                            const audio_offload_info_t *offloadInfo = NULL,
+                            int uid = -1,
+                            pid_t pid = -1);
 
     /* Result of constructing the AudioTrack. This must be checked for successful initialization
      * before using any AudioTrack API (except for set()), because using
@@ -276,7 +284,7 @@
             size_t      frameSize() const   { return mFrameSize; }
 
             uint32_t    channelCount() const { return mChannelCount; }
-            uint32_t    frameCount() const  { return mFrameCount; }
+            size_t      frameCount() const  { return mFrameCount; }
 
     /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */
             sp<IMemory> sharedBuffer() const { return mSharedBuffer; }
@@ -333,7 +341,7 @@
      */
             status_t    setSampleRate(uint32_t sampleRate);
 
-    /* Return current source sample rate in Hz, or 0 if unknown */
+    /* Return current source sample rate in Hz */
             uint32_t    getSampleRate() const;
 
     /* Enables looping and sets the start and end points of looping.
@@ -358,7 +366,7 @@
     /* Sets marker position. When playback reaches the number of frames specified, a callback with
      * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker
      * notification callback.  To set a marker at a position which would compute as 0,
-     * a workaround is to the set the marker at a nearby position such as ~0 or 1.
+     * a workaround is to set the marker at a nearby position such as ~0 or 1.
      * If the AudioTrack has been opened with no callback function associated, the operation will
      * fail.
      *
@@ -449,7 +457,7 @@
      * Returned value:
      *  handle on audio hardware output
      */
-            audio_io_handle_t    getOutput();
+            audio_io_handle_t    getOutput() const;
 
     /* Returns the unique session ID associated with this track.
      *
@@ -563,7 +571,7 @@
             uint32_t    getUnderrunFrames() const;
 
     /* Get the flags */
-            audio_output_flags_t getFlags() const { return mFlags; }
+            audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; }
 
     /* Set parameters - only possible when using direct output */
             status_t    setParameters(const String8& keyValuePairs);
@@ -612,6 +620,7 @@
         bool                mPaused;    // whether thread is requested to pause at next loop entry
         bool                mPausedInt; // whether thread internally requests pause
         nsecs_t             mPausedNs;  // if mPausedInt then associated timeout, otherwise ignored
+        bool                mIgnoreNextPausedInt;   // whether to ignore next mPausedInt request
     };
 
             // body of AudioTrackThread::threadLoop()
@@ -622,52 +631,50 @@
             //      NS_INACTIVE inactive so don't run again until re-started
             //      NS_NEVER    never again
             static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3;
-            nsecs_t processAudioBuffer(const sp<AudioTrackThread>& thread);
-            status_t processStreamEnd(int32_t waitCount);
+            nsecs_t processAudioBuffer();
 
+            bool     isOffloaded() const;
 
             // caller must hold lock on mLock for all _l methods
 
-            status_t createTrack_l(audio_stream_type_t streamType,
-                                 uint32_t sampleRate,
-                                 audio_format_t format,
-                                 size_t frameCount,
-                                 audio_output_flags_t flags,
-                                 const sp<IMemory>& sharedBuffer,
-                                 audio_io_handle_t output,
-                                 size_t epoch);
+            status_t createTrack_l(size_t epoch);
 
             // can only be called when mState != STATE_ACTIVE
             void flush_l();
 
             void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
-            audio_io_handle_t getOutput_l();
 
             // FIXME enum is faster than strcmp() for parameter 'from'
             status_t restoreTrack_l(const char *from);
 
-            bool     isOffloaded() const
+            bool     isOffloaded_l() const
                 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }
 
-    // Next 3 fields may be changed if IAudioTrack is re-created, but always != 0
+    // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0
     sp<IAudioTrack>         mAudioTrack;
     sp<IMemory>             mCblkMemory;
     audio_track_cblk_t*     mCblk;                  // re-load after mLock.unlock()
+    audio_io_handle_t       mOutput;                // returned by AudioSystem::getOutput()
 
     sp<AudioTrackThread>    mAudioTrackThread;
+
     float                   mVolume[2];
     float                   mSendLevel;
-    uint32_t                mSampleRate;
-    size_t                  mFrameCount;            // corresponds to current IAudioTrack
-    size_t                  mReqFrameCount;         // frame count to request the next time a new
-                                                    // IAudioTrack is needed
+    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
+                                                    // a new IAudioTrack is needed, non-decreasing
 
     // constant after constructor or set()
     audio_format_t          mFormat;                // as requested by client, not forced to 16-bit
     audio_stream_type_t     mStreamType;
     uint32_t                mChannelCount;
     audio_channel_mask_t    mChannelMask;
+    sp<IMemory>             mSharedBuffer;
     transfer_type           mTransfer;
+    audio_offload_info_t    mOffloadInfoCopy;
+    const audio_offload_info_t* mOffloadInfo;
 
     // 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.
@@ -708,14 +715,17 @@
     bool                    mRetryOnPartialBuffer;  // sleep and retry after partial obtainBuffer()
     uint32_t                mObservedSequence;      // last observed value of mSequence
 
-    sp<IMemory>             mSharedBuffer;
     uint32_t                mLoopPeriod;            // in frames, zero means looping is disabled
+
     uint32_t                mMarkerPosition;        // in wrapping (overflow) frame units
     bool                    mMarkerReached;
     uint32_t                mNewPosition;           // in frames
     uint32_t                mUpdatePeriod;          // in frames, zero means no EVENT_NEW_POS
 
     audio_output_flags_t    mFlags;
+        // const after set(), except for bits AUDIO_OUTPUT_FLAG_FAST and AUDIO_OUTPUT_FLAG_OFFLOAD.
+        // mLock must be held to read or write those bits reliably.
+
     int                     mSessionId;
     int                     mAuxEffectId;
 
@@ -736,6 +746,7 @@
 
     bool                    mInUnderrun;            // whether track is currently in underrun state
     String8                 mName;                  // server's name for this IAudioTrack
+    uint32_t                mPausedPosition;
 
 private:
     class DeathNotifier : public IBinder::DeathRecipient {
@@ -749,7 +760,8 @@
 
     sp<DeathNotifier>       mDeathNotifier;
     uint32_t                mSequence;              // incremented for each new IAudioTrack attempt
-    audio_io_handle_t       mOutput;                // cached output io handle
+    int                     mClientUid;
+    pid_t                   mClientPid;
 };
 
 class TimedAudioTrack : public AudioTrack

include/media/EffectsFactoryApi.h

diff --git a/include/media/EffectsFactoryApi.h b/include/media/EffectsFactoryApi.h
index b1143b9..b1ed7b0 100644
--- a/include/media/EffectsFactoryApi.h
+++ b/include/media/EffectsFactoryApi.h
@@ -171,30 +171,6 @@
 ////////////////////////////////////////////////////////////////////////////////
 int EffectIsNullUuid(const effect_uuid_t *pEffectUuid);
 
-////////////////////////////////////////////////////////////////////////////////
-//
-//    Function:       EffectGetSubEffects
-//
-//    Description:    Returns the descriptors of the sub effects of the effect
-//                    whose uuid is pointed to by first argument.
-//
-//    Input:
-//          pEffectUuid:    pointer to the effect uuid.
-//          size:           size of the buffer pointed by pDescriptor.
-//
-//    Input/Output:
-//          pDescriptor:    address where to return the sub effect descriptors.
-//
-//    Output:
-//        returned value:    0          successful operation.
-//                          -ENODEV     factory failed to initialize
-//                          -EINVAL     invalid pEffectUuid or pDescriptor
-//                          -ENOENT     no effect with this uuid found
-//        *pDescriptor:     updated with the sub effect descriptors.
-//
-////////////////////////////////////////////////////////////////////////////////
-int EffectGetSubEffects(const effect_uuid_t *pEffectUuid, effect_descriptor_t *pDescriptors, size_t size);
-
 #if __cplusplus
 }  // extern "C"
 #endif

include/media/IAudioFlinger.h

diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 49f921b..7c5f33a 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -53,6 +53,9 @@
     };
     typedef uint32_t track_flags_t;
 
+    // invariant on exit for all APIs that return an sp<>:
+    //   (return value != 0) == (*status == NO_ERROR)
+
     /* create an audio track and registers it with AudioFlinger.
      * return null if the track cannot be created.
      */
@@ -61,9 +64,12 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                size_t frameCount,
+                                size_t *pFrameCount,
                                 track_flags_t *flags,
                                 const sp<IMemory>& sharedBuffer,
+                                // On successful return, AudioFlinger takes over the handle
+                                // reference and will release it when the track is destroyed.
+                                // However on failure, the client is responsible for release.
                                 audio_io_handle_t output,
                                 pid_t tid,  // -1 means unused, otherwise must be valid non-0
                                 int *sessionId,
@@ -71,14 +77,18 @@
                                 // output: server's description of IAudioTrack for display in logs.
                                 // Don't attempt to parse, as the format could change.
                                 String8& name,
+                                int clientUid,
                                 status_t *status) = 0;
 
     virtual sp<IAudioRecord> openRecord(
+                                // On successful return, AudioFlinger takes over the handle
+                                // reference and will release it when the track is destroyed.
+                                // However on failure, the client is responsible for release.
                                 audio_io_handle_t input,
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                size_t frameCount,
+                                size_t *pFrameCount,
                                 track_flags_t *flags,
                                 pid_t tid,  // -1 means unused, otherwise must be valid non-0
                                 int *sessionId,
@@ -159,19 +169,19 @@
                                         audio_channel_mask_t *pChannelMask) = 0;
     virtual status_t closeInput(audio_io_handle_t input) = 0;
 
-    virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) = 0;
+    virtual status_t invalidateStream(audio_stream_type_t stream) = 0;
 
     virtual status_t setVoiceVolume(float volume) = 0;
 
-    virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
+    virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
                                     audio_io_handle_t output) const = 0;
 
-    virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const = 0;
+    virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const = 0;
 
     virtual int newAudioSessionId() = 0;
 
-    virtual void acquireAudioSessionId(int audioSession) = 0;
-    virtual void releaseAudioSessionId(int audioSession) = 0;
+    virtual void acquireAudioSessionId(int audioSession, pid_t pid) = 0;
+    virtual void releaseAudioSessionId(int audioSession, pid_t pid) = 0;
 
     virtual status_t queryNumberEffects(uint32_t *numEffects) const = 0;
 
@@ -184,6 +194,7 @@
                                     effect_descriptor_t *pDesc,
                                     const sp<IEffectClient>& client,
                                     int32_t priority,
+                                    // AudioFlinger doesn't take over handle reference from client
                                     audio_io_handle_t output,
                                     int sessionId,
                                     status_t *status,

include/media/IAudioTrack.h

diff --git a/include/media/IAudioTrack.h b/include/media/IAudioTrack.h
index afac4ae..5c8a484 100644
--- a/include/media/IAudioTrack.h
+++ b/include/media/IAudioTrack.h
@@ -90,6 +90,9 @@
 
     /* Return NO_ERROR if timestamp is valid */
     virtual status_t    getTimestamp(AudioTimestamp& timestamp) = 0;
+
+    /* Signal the playback thread for a change in control block */
+    virtual void        signal() = 0;
 };
 
 // ----------------------------------------------------------------------------

include/media/IMediaHTTPConnection.h

diff --git a/include/media/IMediaHTTPConnection.h b/include/media/IMediaHTTPConnection.h
new file mode 100644
index 0000000..2a63eb7
--- /dev/null
+++ b/include/media/IMediaHTTPConnection.h
@@ -0,0 +1,49 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef I_MEDIA_HTTP_CONNECTION_H_
+
+#define I_MEDIA_HTTP_CONNECTION_H_
+
+#include <binder/IInterface.h>
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/KeyedVector.h>
+
+namespace android {
+
+struct IMediaHTTPConnection;
+
+/** MUST stay in sync with IMediaHTTPConnection.aidl */
+
+struct IMediaHTTPConnection : public IInterface {
+    DECLARE_META_INTERFACE(MediaHTTPConnection);
+
+    virtual bool connect(
+            const char *uri, const KeyedVector<String8, String8> *headers) = 0;
+
+    virtual void disconnect() = 0;
+    virtual ssize_t readAt(off64_t offset, void *data, size_t size) = 0;
+    virtual off64_t getSize() = 0;
+    virtual status_t getMIMEType(String8 *mimeType) = 0;
+    virtual status_t getUri(String8 *uri) = 0;
+
+private:
+    DISALLOW_EVIL_CONSTRUCTORS(IMediaHTTPConnection);
+};
+
+}  // namespace android
+
+#endif  // I_MEDIA_HTTP_CONNECTION_H_

include/media/IMediaHTTPService.h

diff --git a/include/media/IMediaHTTPService.h b/include/media/IMediaHTTPService.h
new file mode 100644
index 0000000..f66d6c8
--- /dev/null
+++ b/include/media/IMediaHTTPService.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef I_MEDIA_HTTP_SERVICE_H_
+
+#define I_MEDIA_HTTP_SERVICE_H_
+
+#include <binder/IInterface.h>
+#include <media/stagefright/foundation/ABase.h>
+
+namespace android {
+
+struct IMediaHTTPConnection;
+
+/** MUST stay in sync with IMediaHTTPService.aidl */
+
+struct IMediaHTTPService : public IInterface {
+    DECLARE_META_INTERFACE(MediaHTTPService);
+
+    virtual sp<IMediaHTTPConnection> makeHTTPConnection() = 0;
+
+private:
+    DISALLOW_EVIL_CONSTRUCTORS(IMediaHTTPService);
+};
+
+}  // namespace android
+
+#endif  // I_MEDIA_HTTP_SERVICE_H_

include/media/IMediaMetadataRetriever.h

diff --git a/include/media/IMediaMetadataRetriever.h b/include/media/IMediaMetadataRetriever.h
index 6dbb2d7..2529800 100644
--- a/include/media/IMediaMetadataRetriever.h
+++ b/include/media/IMediaMetadataRetriever.h
@@ -26,6 +26,8 @@
 
 namespace android {
 
+struct IMediaHTTPService;
+
 class IMediaMetadataRetriever: public IInterface
 {
 public:
@@ -33,6 +35,7 @@
     virtual void            disconnect() = 0;
 
     virtual status_t        setDataSource(
+            const sp<IMediaHTTPService> &httpService,
             const char *srcUrl,
             const KeyedVector<String8, String8> *headers = NULL) = 0;
 

include/media/IMediaPlayer.h

diff --git a/include/media/IMediaPlayer.h b/include/media/IMediaPlayer.h
index 0cbd269..db62cd5 100644
--- a/include/media/IMediaPlayer.h
+++ b/include/media/IMediaPlayer.h
@@ -33,6 +33,7 @@
 class Surface;
 class IStreamSource;
 class IGraphicBufferProducer;
+struct IMediaHTTPService;
 
 class IMediaPlayer: public IInterface
 {
@@ -41,8 +42,11 @@
 
     virtual void            disconnect() = 0;
 
-    virtual status_t        setDataSource(const char *url,
-                                    const KeyedVector<String8, String8>* headers) = 0;
+    virtual status_t        setDataSource(
+            const sp<IMediaHTTPService> &httpService,
+            const char *url,
+            const KeyedVector<String8, String8>* headers) = 0;
+
     virtual status_t        setDataSource(int fd, int64_t offset, int64_t length) = 0;
     virtual status_t        setDataSource(const sp<IStreamSource>& source) = 0;
     virtual status_t        setVideoSurfaceTexture(

include/media/IMediaPlayerService.h

diff --git a/include/media/IMediaPlayerService.h b/include/media/IMediaPlayerService.h
index 2998b37..5b45376 100644
--- a/include/media/IMediaPlayerService.h
+++ b/include/media/IMediaPlayerService.h
@@ -34,6 +34,7 @@
 struct ICrypto;
 struct IDrm;
 struct IHDCP;
+struct IMediaHTTPService;
 class IMediaRecorder;
 class IOMX;
 class IRemoteDisplay;
@@ -49,9 +50,14 @@
     virtual sp<IMediaMetadataRetriever> createMetadataRetriever() = 0;
     virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0) = 0;
 
-    virtual status_t         decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
-                                    audio_format_t* pFormat,
-                                    const sp<IMemoryHeap>& heap, size_t *pSize) = 0;
+    virtual status_t         decode(
+            const sp<IMediaHTTPService> &httpService,
+            const char* url,
+            uint32_t *pSampleRate,
+            int* pNumChannels,
+            audio_format_t* pFormat,
+            const sp<IMemoryHeap>& heap, size_t *pSize) = 0;
+
     virtual status_t         decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate,
                                     int* pNumChannels, audio_format_t* pFormat,
                                     const sp<IMemoryHeap>& heap, size_t *pSize) = 0;
@@ -93,9 +99,6 @@
 
     virtual void addBatteryData(uint32_t params) = 0;
     virtual status_t pullBatteryData(Parcel* reply) = 0;
-
-    virtual status_t updateProxyConfig(
-            const char *host, int32_t port, const char *exclusionList) = 0;
 };
 
 // ----------------------------------------------------------------------------

include/media/IOMX.h

diff --git a/include/media/IOMX.h b/include/media/IOMX.h
index db9093a..f6f9e7a 100644
--- a/include/media/IOMX.h
+++ b/include/media/IOMX.h
@@ -83,6 +83,10 @@
     virtual status_t storeMetaDataInBuffers(
             node_id node, OMX_U32 port_index, OMX_BOOL enable) = 0;
 
+    virtual status_t prepareForAdaptivePlayback(
+            node_id node, OMX_U32 portIndex, OMX_BOOL enable,
+            OMX_U32 maxFrameWidth, OMX_U32 maxFrameHeight) = 0;
+
     virtual status_t enableGraphicBuffers(
             node_id node, OMX_U32 port_index, OMX_BOOL enable) = 0;
 
@@ -138,6 +142,9 @@
     enum InternalOptionType {
         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_START_TIME, // data is an int64_t
+        INTERNAL_OPTION_TIME_LAPSE, // data is an int64_t[2]
     };
     virtual status_t setInternalOption(
             node_id node,

include/media/MediaMetadataRetrieverInterface.h

diff --git a/include/media/MediaMetadataRetrieverInterface.h b/include/media/MediaMetadataRetrieverInterface.h
index ecc3b65..bb6b97b 100644
--- a/include/media/MediaMetadataRetrieverInterface.h
+++ b/include/media/MediaMetadataRetrieverInterface.h
@@ -24,6 +24,8 @@
 
 namespace android {
 
+struct IMediaHTTPService;
+
 // Abstract base class
 class MediaMetadataRetrieverBase : public RefBase
 {
@@ -32,6 +34,7 @@
     virtual             ~MediaMetadataRetrieverBase() {}
 
     virtual status_t    setDataSource(
+            const sp<IMediaHTTPService> &httpService,
             const char *url,
             const KeyedVector<String8, String8> *headers = NULL) = 0;
 

include/media/MediaPlayerInterface.h

diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index 3b151ef..87717da 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -99,6 +99,8 @@
         virtual status_t    getPosition(uint32_t *position) const = 0;
         virtual status_t    getFramesWritten(uint32_t *frameswritten) const = 0;
         virtual int         getSessionId() const = 0;
+        virtual audio_stream_type_t getAudioStreamType() const = 0;
+        virtual uint32_t    getSampleRate() const = 0;
 
         // If no callback is specified, use the "write" API below to submit
         // audio data.
@@ -135,6 +137,7 @@
     }
 
     virtual status_t    setDataSource(
+            const sp<IMediaHTTPService> &httpService,
             const char *url,
             const KeyedVector<String8, String8> *headers = NULL) = 0;
 
@@ -211,11 +214,6 @@
         return INVALID_OPERATION;
     }
 
-    virtual status_t updateProxyConfig(
-            const char *host, int32_t port, const char *exclusionList) {
-        return INVALID_OPERATION;
-    }
-
 private:
     friend class MediaPlayerService;
 

include/media/mediametadataretriever.h

diff --git a/include/media/mediametadataretriever.h b/include/media/mediametadataretriever.h
index 0df77c1..b35cf32 100644
--- a/include/media/mediametadataretriever.h
+++ b/include/media/mediametadataretriever.h
@@ -25,6 +25,7 @@
 
 namespace android {
 
+struct IMediaHTTPService;
 class IMediaPlayerService;
 class IMediaMetadataRetriever;
 
@@ -68,6 +69,7 @@
     void disconnect();
 
     status_t setDataSource(
+            const sp<IMediaHTTPService> &httpService,
             const char *dataSourceUrl,
             const KeyedVector<String8, String8> *headers = NULL);
 

include/media/mediaplayer.h

diff --git a/include/media/mediaplayer.h b/include/media/mediaplayer.h
index 2177c4c..f8e4e3b 100644
--- a/include/media/mediaplayer.h
+++ b/include/media/mediaplayer.h
@@ -45,6 +45,7 @@
     MEDIA_STARTED           = 6,
     MEDIA_PAUSED            = 7,
     MEDIA_STOPPED           = 8,
+    MEDIA_SKIPPED           = 9,
     MEDIA_TIMED_TEXT        = 99,
     MEDIA_ERROR             = 100,
     MEDIA_INFO              = 200,
@@ -188,6 +189,8 @@
     virtual void notify(int msg, int ext1, int ext2, const Parcel *obj) = 0;
 };
 
+struct IMediaHTTPService;
+
 class MediaPlayer : public BnMediaPlayerClient,
                     public virtual IMediaDeathNotifier
 {
@@ -198,6 +201,7 @@
             void            disconnect();
 
             status_t        setDataSource(
+                    const sp<IMediaHTTPService> &httpService,
                     const char *url,
                     const KeyedVector<String8, String8> *headers);
 
@@ -223,9 +227,14 @@
             bool            isLooping();
             status_t        setVolume(float leftVolume, float rightVolume);
             void            notify(int msg, int ext1, int ext2, const Parcel *obj = NULL);
-    static  status_t        decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
-                                   audio_format_t* pFormat,
-                                   const sp<IMemoryHeap>& heap, size_t *pSize);
+    static  status_t        decode(
+            const sp<IMediaHTTPService> &httpService,
+            const char* url,
+            uint32_t *pSampleRate,
+            int* pNumChannels,
+            audio_format_t* pFormat,
+            const sp<IMemoryHeap>& heap,
+            size_t *pSize);
     static  status_t        decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate,
                                    int* pNumChannels, audio_format_t* pFormat,
                                    const sp<IMemoryHeap>& heap, size_t *pSize);
@@ -241,9 +250,6 @@
             status_t        setRetransmitEndpoint(const char* addrString, uint16_t port);
             status_t        setNextMediaPlayer(const sp<MediaPlayer>& player);
 
-            status_t updateProxyConfig(
-                    const char *host, int32_t port, const char *exclusionList);
-
 private:
             void            clear_l();
             status_t        seekTo_l(int msec);

include/media/mediarecorder.h

diff --git a/include/media/mediarecorder.h b/include/media/mediarecorder.h
index 88a42a0..142cb90 100644
--- a/include/media/mediarecorder.h
+++ b/include/media/mediarecorder.h
@@ -39,7 +39,7 @@
 enum video_source {
     VIDEO_SOURCE_DEFAULT = 0,
     VIDEO_SOURCE_CAMERA = 1,
-    VIDEO_SOURCE_GRALLOC_BUFFER = 2,
+    VIDEO_SOURCE_SURFACE = 2,
 
     VIDEO_SOURCE_LIST_END  // must be last - used to validate audio source type
 };

include/media/mediascanner.h

diff --git a/include/media/mediascanner.h b/include/media/mediascanner.h
index a73403b..4537679 100644
--- a/include/media/mediascanner.h
+++ b/include/media/mediascanner.h
@@ -21,6 +21,7 @@
 #include <utils/threads.h>
 #include <utils/List.h>
 #include <utils/Errors.h>
+#include <utils/String8.h>
 #include <pthread.h>
 
 struct dirent;
@@ -29,6 +30,7 @@
 
 class MediaScannerClient;
 class StringArray;
+class CharacterEncodingDetector;
 
 enum MediaScanResult {
     // This file or directory was scanned successfully.
@@ -94,15 +96,9 @@
     virtual status_t setMimeType(const char* mimeType) = 0;
 
 protected:
-    void convertValues(uint32_t encoding);
-
-protected:
-    // cached name and value strings, for native encoding support.
-    StringArray*    mNames;
-    StringArray*    mValues;
-
-    // default encoding based on MediaScanner::mLocale string
-    uint32_t        mLocaleEncoding;
+    // default encoding from MediaScanner::mLocale
+    String8 mLocale;
+    CharacterEncodingDetector *mEncodingDetector;
 };
 
 }; // namespace android

include/media/nbaio/AudioBufferProviderSource.h

diff --git a/include/media/nbaio/AudioBufferProviderSource.h b/include/media/nbaio/AudioBufferProviderSource.h
index 2c4aaff..b16e20a 100644
--- a/include/media/nbaio/AudioBufferProviderSource.h
+++ b/include/media/nbaio/AudioBufferProviderSource.h
@@ -27,7 +27,7 @@
 class AudioBufferProviderSource : public NBAIO_Source {
 
 public:
-    AudioBufferProviderSource(AudioBufferProvider *provider, NBAIO_Format format);
+    AudioBufferProviderSource(AudioBufferProvider *provider, const NBAIO_Format& format);
     virtual ~AudioBufferProviderSource();
 
     // NBAIO_Port interface

include/media/nbaio/AudioStreamInSource.h

diff --git a/include/media/nbaio/AudioStreamInSource.h b/include/media/nbaio/AudioStreamInSource.h
index 07d8c89..eaea63c 100644
--- a/include/media/nbaio/AudioStreamInSource.h
+++ b/include/media/nbaio/AudioStreamInSource.h
@@ -43,7 +43,7 @@
 
     // This is an over-estimate, and could dupe the caller into making a blocking read()
     // FIXME Use an audio HAL API to query the buffer filling status when it's available.
-    virtual ssize_t availableToRead() { return mStreamBufferSizeBytes >> mBitShift; }
+    virtual ssize_t availableToRead() { return mStreamBufferSizeBytes / mFrameSize; }
 
     virtual ssize_t read(void *buffer, size_t count);
 

include/media/nbaio/AudioStreamOutSink.h

diff --git a/include/media/nbaio/AudioStreamOutSink.h b/include/media/nbaio/AudioStreamOutSink.h
index 7948d40..9949b88 100644
--- a/include/media/nbaio/AudioStreamOutSink.h
+++ b/include/media/nbaio/AudioStreamOutSink.h
@@ -43,7 +43,7 @@
 
     // This is an over-estimate, and could dupe the caller into making a blocking write()
     // FIXME Use an audio HAL API to query the buffer emptying status when it's available.
-    virtual ssize_t availableToWrite() const { return mStreamBufferSizeBytes >> mBitShift; }
+    virtual ssize_t availableToWrite() const { return mStreamBufferSizeBytes / mFrameSize; }
 
     virtual ssize_t write(const void *buffer, size_t count);
 

include/media/nbaio/MonoPipe.h

diff --git a/include/media/nbaio/MonoPipe.h b/include/media/nbaio/MonoPipe.h
index d3802fe..b09b35f 100644
--- a/include/media/nbaio/MonoPipe.h
+++ b/include/media/nbaio/MonoPipe.h
@@ -41,7 +41,7 @@
     // Note: whatever shares this object with another thread needs to do so in an SMP-safe way (like
     // creating it the object before creating the other thread, or storing the object with a
     // release_store). Otherwise the other thread could see a partially-constructed object.
-    MonoPipe(size_t reqFrames, NBAIO_Format format, bool writeCanBlock = false);
+    MonoPipe(size_t reqFrames, const NBAIO_Format& format, bool writeCanBlock = false);
     virtual ~MonoPipe();
 
     // NBAIO_Port interface

include/media/nbaio/NBAIO.h

diff --git a/include/media/nbaio/NBAIO.h b/include/media/nbaio/NBAIO.h
index 1da0c73..ab9a4a5 100644
--- a/include/media/nbaio/NBAIO.h
+++ b/include/media/nbaio/NBAIO.h
@@ -29,6 +29,7 @@
 #include <utils/Errors.h>
 #include <utils/RefBase.h>
 #include <media/AudioTimestamp.h>
+#include <system/audio.h>
 
 namespace android {
 
@@ -52,31 +53,41 @@
 // the combinations that are actually needed within AudioFlinger.  If the list of combinations grows
 // too large, then this decision should be re-visited.
 // Sample rate and channel count are explicit, PCM interleaved 16-bit is assumed.
-typedef unsigned NBAIO_Format;
-enum {
-    Format_Invalid
+struct NBAIO_Format {
+//private:
+    unsigned    mSampleRate;
+    unsigned    mChannelCount;
+    audio_format_t  mFormat;
+    ssize_t     mFrameSize;
 };
 
-// Return the frame size of an NBAIO_Format in bytes
-size_t Format_frameSize(NBAIO_Format format);
+extern const NBAIO_Format Format_Invalid;
 
-// Return the frame size of an NBAIO_Format as a bit shift
-size_t Format_frameBitShift(NBAIO_Format format);
+// Return the frame size of an NBAIO_Format in bytes
+size_t Format_frameSize(const NBAIO_Format& format);
 
 // Convert a sample rate in Hz and channel count to an NBAIO_Format
-NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount);
+// FIXME Remove the default value of AUDIO_FORMAT_PCM_16_BIT, and rename
+NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount,
+        audio_format_t format = AUDIO_FORMAT_PCM_16_BIT);
 
 // Return the sample rate in Hz of an NBAIO_Format
-unsigned Format_sampleRate(NBAIO_Format format);
+unsigned Format_sampleRate(const NBAIO_Format& format);
 
 // Return the channel count of an NBAIO_Format
-unsigned Format_channelCount(NBAIO_Format format);
+unsigned Format_channelCount(const NBAIO_Format& format);
 
 // Callbacks used by NBAIO_Sink::writeVia() and NBAIO_Source::readVia() below.
 typedef ssize_t (*writeVia_t)(void *user, void *buffer, size_t count);
 typedef ssize_t (*readVia_t)(void *user, const void *buffer,
                              size_t count, int64_t readPTS);
 
+// Check whether an NBAIO_Format is valid
+bool Format_isValid(const NBAIO_Format& format);
+
+// Compare two NBAIO_Format values
+bool Format_isEqual(const NBAIO_Format& format1, const NBAIO_Format& format2);
+
 // Abstract class (interface) representing a data port.
 class NBAIO_Port : public RefBase {
 
@@ -115,15 +126,15 @@
     virtual NBAIO_Format format() const { return mNegotiated ? mFormat : Format_Invalid; }
 
 protected:
-    NBAIO_Port(NBAIO_Format format) : mNegotiated(false), mFormat(format),
-                                      mBitShift(Format_frameBitShift(format)) { }
+    NBAIO_Port(const NBAIO_Format& format) : mNegotiated(false), mFormat(format),
+                                             mFrameSize(Format_frameSize(format)) { }
     virtual ~NBAIO_Port() { }
 
     // Implementations are free to ignore these if they don't need them
 
     bool            mNegotiated;    // mNegotiated implies (mFormat != Format_Invalid)
     NBAIO_Format    mFormat;        // (mFormat != Format_Invalid) does not imply mNegotiated
-    size_t          mBitShift;      // assign in parallel with any assignment to mFormat
+    size_t          mFrameSize;     // assign in parallel with any assignment to mFormat
 };
 
 // Abstract class (interface) representing a non-blocking data sink, for use by a data provider.
@@ -220,7 +231,7 @@
     virtual status_t getTimestamp(AudioTimestamp& timestamp) { return INVALID_OPERATION; }
 
 protected:
-    NBAIO_Sink(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
@@ -311,7 +322,7 @@
     virtual void    onTimestamp(const AudioTimestamp& timestamp) { }
 
 protected:
-    NBAIO_Source(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

include/media/nbaio/NBLog.h

diff --git a/include/media/nbaio/NBLog.h b/include/media/nbaio/NBLog.h
index 6d59ea7..bcbbc04 100644
--- a/include/media/nbaio/NBLog.h
+++ b/include/media/nbaio/NBLog.h
@@ -25,6 +25,8 @@
 
 namespace android {
 
+class String8;
+
 class NBLog {
 
 public:
@@ -187,6 +189,10 @@
     const Shared* const mShared; // raw pointer to shared memory
     const sp<IMemory> mIMemory; // ref-counted version
     int32_t     mFront;         // index of oldest acknowledged Entry
+    int     mFd;                // file descriptor
+    int     mIndent;            // indentation level
+
+    void    dumpLine(const String8& timestamp, String8& body);
 
     static const size_t kSquashTimestamp = 5; // squash this many or more adjacent timestamps
 };

include/media/nbaio/Pipe.h

diff --git a/include/media/nbaio/Pipe.h b/include/media/nbaio/Pipe.h
index 79a4eee..c784129 100644
--- a/include/media/nbaio/Pipe.h
+++ b/include/media/nbaio/Pipe.h
@@ -30,7 +30,7 @@
 
 public:
     // maxFrames will be rounded up to a power of 2, and all slots are available. Must be >= 2.
-    Pipe(size_t maxFrames, NBAIO_Format format);
+    Pipe(size_t maxFrames, const NBAIO_Format& format);
     virtual ~Pipe();
 
     // NBAIO_Port interface

include/media/nbaio/SourceAudioBufferProvider.h

diff --git a/include/media/nbaio/SourceAudioBufferProvider.h b/include/media/nbaio/SourceAudioBufferProvider.h
index cdfb6fe..daf6bc3 100644
--- a/include/media/nbaio/SourceAudioBufferProvider.h
+++ b/include/media/nbaio/SourceAudioBufferProvider.h
@@ -41,7 +41,7 @@
 
 private:
     const sp<NBAIO_Source> mSource;     // the wrapped source
-    /*const*/ size_t    mFrameBitShift; // log2(frame size in bytes)
+    /*const*/ size_t    mFrameSize; // frame size in bytes
     void*               mAllocated; // pointer to base of allocated memory
     size_t              mSize;      // size of mAllocated in frames
     size_t              mOffset;    // frame offset within mAllocated of valid data

include/media/stagefright/ACodec.h

diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h
index a8ffd4a..36f2a67 100644
--- a/include/media/stagefright/ACodec.h
+++ b/include/media/stagefright/ACodec.h
@@ -67,6 +67,8 @@
 
     void signalRequestIDRFrame();
 
+    bool isConfiguredForAdaptivePlayback() { return mIsConfiguredForAdaptivePlayback; }
+
     struct PortDescription : public RefBase {
         size_t countBuffers();
         IOMX::buffer_id bufferIDAt(size_t index) const;
@@ -116,6 +118,7 @@
         kWhatStart                   = 'star',
         kWhatRequestIDRFrame         = 'ridr',
         kWhatSetParameters           = 'setP',
+        kWhatSubmitOutputMetaDataBufferIfEOS = 'subm',
     };
 
     enum {
@@ -186,6 +189,7 @@
     bool mIsEncoder;
     bool mUseMetadataOnEncoderOutput;
     bool mShutdownInProgress;
+    bool mIsConfiguredForAdaptivePlayback;
 
     // If "mKeepComponentAllocated" we only transition back to Loaded state
     // and do not release the component instance.
@@ -201,6 +205,12 @@
     int32_t mMetaDataBuffersToSubmit;
 
     int64_t mRepeatFrameDelayUs;
+    int64_t mMaxPtsGapUs;
+
+    int64_t mTimePerFrameUs;
+    int64_t mTimePerCaptureUs;
+
+    bool mCreateInputBuffersSuspended;
 
     status_t setCyclicIntraMacroblockRefresh(const sp<AMessage> &msg, int32_t mode);
     status_t allocateBuffersOnPort(OMX_U32 portIndex);
@@ -212,6 +222,7 @@
             OMX_U32 *nMinUndequeuedBuffers);
     status_t allocateOutputMetaDataBuffers();
     status_t submitOutputMetaDataBuffer();
+    void signalSubmitOutputMetaDataBufferIfEOS_workaround();
     status_t allocateOutputBuffersFromNativeWindow();
     status_t cancelBufferToNativeWindow(BufferInfo *info);
     status_t freeOutputBuffersNotOwnedByComponent();
@@ -247,6 +258,8 @@
             int32_t numChannels, int32_t sampleRate, int32_t bitRate,
             int32_t aacProfile, bool isADTS);
 
+    status_t setupAC3Codec(bool encoder, int32_t numChannels, int32_t sampleRate);
+
     status_t selectAudioPortFormat(
             OMX_U32 portIndex, OMX_AUDIO_CODINGTYPE desiredFormat);
 
@@ -264,6 +277,7 @@
     status_t setupMPEG4EncoderParameters(const sp<AMessage> &msg);
     status_t setupH263EncoderParameters(const sp<AMessage> &msg);
     status_t setupAVCEncoderParameters(const sp<AMessage> &msg);
+    status_t setupVPXEncoderParameters(const sp<AMessage> &msg);
 
     status_t verifySupportForProfileAndLevel(int32_t profile, int32_t level);
 

include/media/stagefright/AudioPlayer.h

diff --git a/include/media/stagefright/AudioPlayer.h b/include/media/stagefright/AudioPlayer.h
index 912a43c..14afb85 100644
--- a/include/media/stagefright/AudioPlayer.h
+++ b/include/media/stagefright/AudioPlayer.h
@@ -129,7 +129,7 @@
     void reset();
 
     uint32_t getNumFramesPendingPlayout() const;
-    int64_t getOutputPlayPositionUs_l() const;
+    int64_t getOutputPlayPositionUs_l();
 
     bool allowDeepBuffering() const { return (mCreateFlags & ALLOW_DEEP_BUFFERING) != 0; }
     bool useOffload() const { return (mCreateFlags & USE_OFFLOAD) != 0; }

include/media/stagefright/CameraSource.h

diff --git a/include/media/stagefright/CameraSource.h b/include/media/stagefright/CameraSource.h
index a829916..dd0a106 100644
--- a/include/media/stagefright/CameraSource.h
+++ b/include/media/stagefright/CameraSource.h
@@ -172,7 +172,7 @@
                  const sp<IGraphicBufferProducer>& surface,
                  bool storeMetaDataInVideoBuffers);
 
-    virtual void startCameraRecording();
+    virtual status_t startCameraRecording();
     virtual void releaseRecordingFrame(const sp<IMemory>& frame);
 
     // Returns true if need to skip the current frame.
@@ -185,6 +185,8 @@
     virtual void dataCallbackTimestamp(int64_t timestampUs, int32_t msgType,
             const sp<IMemory> &data);
 
+    void releaseCamera();
+
 private:
     friend class CameraSourceListener;
 
@@ -233,7 +235,6 @@
                     int32_t frameRate);
 
     void stopCameraRecording();
-    void releaseCamera();
     status_t reset();
 
     CameraSource(const CameraSource &);

include/media/stagefright/CameraSourceTimeLapse.h

diff --git a/include/media/stagefright/CameraSourceTimeLapse.h b/include/media/stagefright/CameraSourceTimeLapse.h
index 6b7a63c..34213be 100644
--- a/include/media/stagefright/CameraSourceTimeLapse.h
+++ b/include/media/stagefright/CameraSourceTimeLapse.h
@@ -41,7 +41,8 @@
         Size videoSize,
         int32_t videoFrameRate,
         const sp<IGraphicBufferProducer>& surface,
-        int64_t timeBetweenTimeLapseFrameCaptureUs);
+        int64_t timeBetweenTimeLapseFrameCaptureUs,
+        bool storeMetaDataInVideoBuffers = true);
 
     virtual ~CameraSourceTimeLapse();
 
@@ -116,7 +117,8 @@
         Size videoSize,
         int32_t videoFrameRate,
         const sp<IGraphicBufferProducer>& surface,
-        int64_t timeBetweenTimeLapseFrameCaptureUs);
+        int64_t timeBetweenTimeLapseFrameCaptureUs,
+        bool storeMetaDataInVideoBuffers = true);
 
     // Wrapper over CameraSource::signalBufferReturned() to implement quick stop.
     // It only handles the case when mLastReadBufferCopy is signalled. Otherwise

include/media/stagefright/DataSource.h

diff --git a/include/media/stagefright/DataSource.h b/include/media/stagefright/DataSource.h
index 742bc0e..f8787dd 100644
--- a/include/media/stagefright/DataSource.h
+++ b/include/media/stagefright/DataSource.h
@@ -31,6 +31,7 @@
 namespace android {
 
 struct AMessage;
+struct IMediaHTTPService;
 class String8;
 
 class DataSource : public RefBase {
@@ -43,6 +44,7 @@
     };
 
     static sp<DataSource> CreateFromURI(
+            const sp<IMediaHTTPService> &httpService,
             const char *uri,
             const KeyedVector<String8, String8> *headers = NULL);
 
@@ -80,7 +82,6 @@
             const sp<DataSource> &source, String8 *mimeType,
             float *confidence, sp<AMessage> *meta);
 
-    static void RegisterSniffer(SnifferFunc func);
     static void RegisterDefaultSniffers();
 
     // for DRM
@@ -101,6 +102,9 @@
 private:
     static Mutex gSnifferMutex;
     static List<SnifferFunc> gSniffers;
+    static bool gSniffersRegistered;
+
+    static void RegisterSniffer_l(SnifferFunc func);
 
     DataSource(const DataSource &);
     DataSource &operator=(const DataSource &);

include/media/stagefright/DataURISource.h

diff --git a/include/media/stagefright/DataURISource.h b/include/media/stagefright/DataURISource.h
new file mode 100644
index 0000000..693562e
--- /dev/null
+++ b/include/media/stagefright/DataURISource.h
@@ -0,0 +1,49 @@
+/*
+ * 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 DATA_URI_SOURCE_H_
+
+#define DATA_URI_SOURCE_H_
+
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/foundation/ABase.h>
+
+namespace android {
+
+struct ABuffer;
+
+struct DataURISource : public DataSource {
+    static sp<DataURISource> Create(const char *uri);
+
+    virtual status_t initCheck() const;
+    virtual ssize_t readAt(off64_t offset, void *data, size_t size);
+    virtual status_t getSize(off64_t *size);
+
+protected:
+    virtual ~DataURISource();
+
+private:
+    sp<ABuffer> mBuffer;
+
+    DataURISource(const sp<ABuffer> &buffer);
+
+    DISALLOW_EVIL_CONSTRUCTORS(DataURISource);
+};
+
+}  // namespace android
+
+#endif  // DATA_URI_SOURCE_H_
+

include/media/stagefright/MediaCodecSource.h

diff --git a/include/media/stagefright/MediaCodecSource.h b/include/media/stagefright/MediaCodecSource.h
new file mode 100644
index 0000000..4b18a0b
--- /dev/null
+++ b/include/media/stagefright/MediaCodecSource.h
@@ -0,0 +1,134 @@
+/*
+ * 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 MediaCodecSource_H_
+#define MediaCodecSource_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <media/stagefright/foundation/AHandlerReflector.h>
+#include <media/stagefright/MediaSource.h>
+
+namespace android {
+
+class ALooper;
+class AMessage;
+class IGraphicBufferProducer;
+class MediaCodec;
+class MetaData;
+
+struct MediaCodecSource : public MediaSource,
+                          public MediaBufferObserver {
+    enum FlagBits {
+        FLAG_USE_SURFACE_INPUT      = 1,
+        FLAG_USE_METADATA_INPUT     = 2,
+    };
+
+    static sp<MediaCodecSource> Create(
+            const sp<ALooper> &looper,
+            const sp<AMessage> &format,
+            const sp<MediaSource> &source,
+            uint32_t flags = 0);
+
+    bool isVideo() const { return mIsVideo; }
+    sp<IGraphicBufferProducer> getGraphicBufferProducer();
+
+    // MediaSource
+    virtual status_t start(MetaData *params = NULL);
+    virtual status_t stop();
+    virtual status_t pause();
+    virtual sp<MetaData> getFormat() { return mMeta; }
+    virtual status_t read(
+            MediaBuffer **buffer,
+            const ReadOptions *options = NULL);
+
+    // MediaBufferObserver
+    virtual void signalBufferReturned(MediaBuffer *buffer);
+
+    // for AHandlerReflector
+    void onMessageReceived(const sp<AMessage> &msg);
+
+protected:
+    virtual ~MediaCodecSource();
+
+private:
+    struct Puller;
+
+    enum {
+        kWhatPullerNotify,
+        kWhatEncoderActivity,
+        kWhatStart,
+        kWhatStop,
+        kWhatPause,
+    };
+
+    MediaCodecSource(
+            const sp<ALooper> &looper,
+            const sp<AMessage> &outputFormat,
+            const sp<MediaSource> &source,
+            uint32_t flags = 0);
+
+    status_t onStart(MetaData *params);
+    status_t init();
+    status_t initEncoder();
+    void releaseEncoder();
+    status_t feedEncoderInputBuffers();
+    void scheduleDoMoreWork();
+    status_t doMoreWork();
+    void suspend();
+    void resume(int64_t skipFramesBeforeUs = -1ll);
+    void signalEOS(status_t err = ERROR_END_OF_STREAM);
+    bool reachedEOS();
+    status_t postSynchronouslyAndReturnError(const sp<AMessage> &msg);
+
+    sp<ALooper> mLooper;
+    sp<ALooper> mCodecLooper;
+    sp<AHandlerReflector<MediaCodecSource> > mReflector;
+    sp<AMessage> mOutputFormat;
+    sp<MetaData> mMeta;
+    sp<Puller> mPuller;
+    sp<MediaCodec> mEncoder;
+    uint32_t mFlags;
+    List<uint32_t> mStopReplyIDQueue;
+    bool mIsVideo;
+    bool mStarted;
+    bool mStopping;
+    bool mDoMoreWorkPending;
+    bool mPullerReachedEOS;
+    sp<AMessage> mEncoderActivityNotify;
+    sp<IGraphicBufferProducer> mGraphicBufferProducer;
+    Vector<sp<ABuffer> > mEncoderInputBuffers;
+    Vector<sp<ABuffer> > mEncoderOutputBuffers;
+    List<MediaBuffer *> mInputBufferQueue;
+    List<size_t> mAvailEncoderInputIndices;
+    List<int64_t> mDecodingTimeQueue; // decoding time (us) for video
+
+    // audio drift time
+    int64_t mFirstSampleTimeUs;
+    List<int64_t> mDriftTimeQueue;
+
+    // following variables are protected by mOutputBufferLock
+    Mutex mOutputBufferLock;
+    Condition mOutputBufferCond;
+    List<MediaBuffer*> mOutputBufferQueue;
+    bool mEncodedReachedEOS;
+    status_t mErrorCode;
+
+    DISALLOW_EVIL_CONSTRUCTORS(MediaCodecSource);
+};
+
+} // namespace android
+
+#endif /* MediaCodecSource_H_ */

include/media/stagefright/MediaDefs.h

diff --git a/include/media/stagefright/MediaDefs.h b/include/media/stagefright/MediaDefs.h
index 85693d4..678d642 100644
--- a/include/media/stagefright/MediaDefs.h
+++ b/include/media/stagefright/MediaDefs.h
@@ -38,12 +38,14 @@
 extern const char *MEDIA_MIMETYPE_AUDIO_AAC;
 extern const char *MEDIA_MIMETYPE_AUDIO_QCELP;
 extern const char *MEDIA_MIMETYPE_AUDIO_VORBIS;
+extern const char *MEDIA_MIMETYPE_AUDIO_OPUS;
 extern const char *MEDIA_MIMETYPE_AUDIO_G711_ALAW;
 extern const char *MEDIA_MIMETYPE_AUDIO_G711_MLAW;
 extern const char *MEDIA_MIMETYPE_AUDIO_RAW;
 extern const char *MEDIA_MIMETYPE_AUDIO_FLAC;
 extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS;
 extern const char *MEDIA_MIMETYPE_AUDIO_MSGSM;
+extern const char *MEDIA_MIMETYPE_AUDIO_AC3;
 
 extern const char *MEDIA_MIMETYPE_CONTAINER_MPEG4;
 extern const char *MEDIA_MIMETYPE_CONTAINER_WAV;

include/media/stagefright/MediaHTTP.h

diff --git a/include/media/stagefright/MediaHTTP.h b/include/media/stagefright/MediaHTTP.h
new file mode 100644
index 0000000..006d8d8
--- /dev/null
+++ b/include/media/stagefright/MediaHTTP.h
@@ -0,0 +1,77 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MEDIA_HTTP_H_
+
+#define MEDIA_HTTP_H_
+
+#include <media/stagefright/foundation/AString.h>
+
+#include "include/HTTPBase.h"
+
+namespace android {
+
+struct IMediaHTTPConnection;
+
+struct MediaHTTP : public HTTPBase {
+    MediaHTTP(const sp<IMediaHTTPConnection> &conn);
+
+    virtual status_t connect(
+            const char *uri,
+            const KeyedVector<String8, String8> *headers,
+            off64_t offset);
+
+    virtual void disconnect();
+
+    virtual status_t initCheck() const;
+
+    virtual ssize_t readAt(off64_t offset, void *data, size_t size);
+
+    virtual status_t getSize(off64_t *size);
+
+    virtual uint32_t flags();
+
+    virtual status_t reconnectAtOffset(off64_t offset);
+
+protected:
+    virtual ~MediaHTTP();
+
+    virtual sp<DecryptHandle> DrmInitialization(const char* mime);
+    virtual void getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client);
+    virtual String8 getUri();
+    virtual String8 getMIMEType() const;
+
+private:
+    status_t mInitCheck;
+    sp<IMediaHTTPConnection> mHTTPConnection;
+
+    KeyedVector<String8, String8> mLastHeaders;
+    AString mLastURI;
+
+    bool mCachedSizeValid;
+    off64_t mCachedSize;
+
+    sp<DecryptHandle> mDecryptHandle;
+    DrmManagerClient *mDrmManagerClient;
+
+    void clearDRMState_l();
+
+    DISALLOW_EVIL_CONSTRUCTORS(MediaHTTP);
+};
+
+}  // namespace android
+
+#endif  // MEDIA_HTTP_H_

include/media/stagefright/MediaSource.h

diff --git a/include/media/stagefright/MediaSource.h b/include/media/stagefright/MediaSource.h
index 3818e63..204d1c6 100644
--- a/include/media/stagefright/MediaSource.h
+++ b/include/media/stagefright/MediaSource.h
@@ -105,7 +105,7 @@
     // This will be called after a successful start() and before the
     // first read() call.
     // Callee assumes ownership of the buffers if no error is returned.
-    virtual status_t setBuffers(const Vector<MediaBuffer *> &buffers) {
+    virtual status_t setBuffers(const Vector<MediaBuffer *> & /* buffers */) {
         return ERROR_UNSUPPORTED;
     }
 

include/media/stagefright/MetaData.h

diff --git a/include/media/stagefright/MetaData.h b/include/media/stagefright/MetaData.h
index de3fc36..e862ec3 100644
--- a/include/media/stagefright/MetaData.h
+++ b/include/media/stagefright/MetaData.h
@@ -56,6 +56,9 @@
     kKeyD263              = 'd263',  // raw data
     kKeyVorbisInfo        = 'vinf',  // raw data
     kKeyVorbisBooks       = 'vboo',  // raw data
+    kKeyOpusHeader        = 'ohdr',  // raw data
+    kKeyOpusCodecDelay    = 'ocod',  // uint64_t (codec delay in ns)
+    kKeyOpusSeekPreRoll   = 'ospr',  // uint64_t (seek preroll in ns)
     kKeyWantsNALFragments = 'NALf',
     kKeyIsSyncFrame       = 'sync',  // int32_t (bool)
     kKeyIsCodecConfig     = 'conf',  // int32_t (bool)
@@ -134,6 +137,7 @@
     kKeyRequiresSecureBuffers = 'secu',  // bool (int32_t)
 
     kKeyIsADTS            = 'adts',  // bool (int32_t)
+    kKeyAACAOT            = 'aaot',  // int32_t
 
     // If a MediaBuffer's data represents (at least partially) encrypted
     // data, the following fields aid in decryption.
@@ -214,6 +218,8 @@
     bool findData(uint32_t key, uint32_t *type,
                   const void **data, size_t *size) const;
 
+    bool hasData(uint32_t key) const;
+
     void dumpToLog() const;
 
 protected:

include/media/stagefright/NuMediaExtractor.h

diff --git a/include/media/stagefright/NuMediaExtractor.h b/include/media/stagefright/NuMediaExtractor.h
index 5ae6f6b..402e7f8 100644
--- a/include/media/stagefright/NuMediaExtractor.h
+++ b/include/media/stagefright/NuMediaExtractor.h
@@ -31,6 +31,7 @@
 struct ABuffer;
 struct AMessage;
 struct DataSource;
+struct IMediaHTTPService;
 struct MediaBuffer;
 struct MediaExtractor;
 struct MediaSource;
@@ -45,6 +46,7 @@
     NuMediaExtractor();
 
     status_t setDataSource(
+            const sp<IMediaHTTPService> &httpService,
             const char *path,
             const KeyedVector<String8, String8> *headers = NULL);
 

include/media/stagefright/OMXCodec.h

diff --git a/include/media/stagefright/OMXCodec.h b/include/media/stagefright/OMXCodec.h
index daaf20f..5121c17 100644
--- a/include/media/stagefright/OMXCodec.h
+++ b/include/media/stagefright/OMXCodec.h
@@ -248,6 +248,8 @@
             int32_t numChannels, int32_t sampleRate, int32_t bitRate,
             int32_t aacProfile, bool isADTS);
 
+    status_t setAC3Format(int32_t numChannels, int32_t sampleRate);
+
     void setG711Format(int32_t numChannels);
 
     status_t setVideoPortFormatType(

include/media/stagefright/Utils.h

diff --git a/include/media/stagefright/Utils.h b/include/media/stagefright/Utils.h
index c24f612..bbad271 100644
--- a/include/media/stagefright/Utils.h
+++ b/include/media/stagefright/Utils.h
@@ -57,7 +57,8 @@
 status_t sendMetaDataToHal(sp<MediaPlayerBase::AudioSink>& sink, const sp<MetaData>& meta);
 
 // Check whether the stream defined by meta can be offloaded to hardware
-bool canOffloadStream(const sp<MetaData>& meta, bool hasVideo, bool isStreaming);
+bool canOffloadStream(const sp<MetaData>& meta, bool hasVideo,
+                      bool isStreaming, audio_stream_type_t streamType);
 
 }  // namespace android
 

include/media/stagefright/timedtext/TimedTextDriver.h

diff --git a/include/media/stagefright/timedtext/TimedTextDriver.h b/include/media/stagefright/timedtext/TimedTextDriver.h
index f23c337..37ef674 100644
--- a/include/media/stagefright/timedtext/TimedTextDriver.h
+++ b/include/media/stagefright/timedtext/TimedTextDriver.h
@@ -25,6 +25,7 @@
 namespace android {
 
 class ALooper;
+struct IMediaHTTPService;
 class MediaPlayerBase;
 class MediaSource;
 class Parcel;
@@ -34,7 +35,9 @@
 
 class TimedTextDriver {
 public:
-    TimedTextDriver(const wp<MediaPlayerBase> &listener);
+    TimedTextDriver(
+            const wp<MediaPlayerBase> &listener,
+            const sp<IMediaHTTPService> &httpService);
 
     ~TimedTextDriver();
 
@@ -77,6 +80,7 @@
     sp<ALooper> mLooper;
     sp<TimedTextPlayer> mPlayer;
     wp<MediaPlayerBase> mListener;
+    sp<IMediaHTTPService> mHTTPService;
 
     // Variables to be guarded by mLock.
     State mState;

include/private/media/AudioTrackShared.h

diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 4b13887..3901e79 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -65,7 +65,9 @@
 struct AudioTrackSharedStatic {
     StaticAudioTrackSingleStateQueue::Shared
                     mSingleStateQueue;
-    size_t          mBufferPosition;    // updated asynchronously by server,
+    // 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"
 };
 
@@ -96,11 +98,7 @@
                                         // The value should be used "for entertainment purposes only",
                                         // which means don't make important decisions based on it.
 
-                size_t      frameCount_;    // used during creation to pass actual track buffer size
-                                            // from AudioFlinger to client, and not referenced again
-                                            // FIXME remove here and replace by createTrack() in/out
-                                            // parameter
-                                            // renamed to "_" to detect incorrect use
+                uint32_t    mPad1;      // unused
 
     volatile    int32_t     mFutex;     // event flag: down (P) by client,
                                         // up (V) by server or binderDied() or interrupt()
@@ -108,7 +106,9 @@
 
 private:
 
-                size_t      mMinimum;       // server wakes up client if available >= mMinimum
+                // 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    mMinimum;       // server wakes up client if available >= mMinimum
 
                 // Channel volumes are fixed point U4.12, so 0x1000 means 1.0.
                 // Left channel is in [0:15], right channel is in [16:31].
@@ -245,7 +245,11 @@
     }
 
     void        setMinimum(size_t minimum) {
-        mCblk->mMinimum = minimum;
+        // This can only happen on a 64-bit client
+        if (minimum > UINT32_MAX) {
+            minimum = UINT32_MAX;
+        }
+        mCblk->mMinimum = (uint32_t) minimum;
     }
 
     // Return the number of frames that would need to be obtained and released
@@ -256,6 +260,8 @@
         return mEpoch;
     }
 
+    size_t      getFramesFilled();
+
 private:
     size_t      mEpoch;
 };
@@ -358,6 +364,7 @@
     //      which must be > 0.
     //  buffer->mNonContig is unused.
     //  buffer->mRaw is unused.
+    //  ackFlush is true iff being called from Track::start to acknowledge a pending flush.
     // On exit:
     //  buffer->mFrameCount has the actual number of contiguous available frames,
     //      which is always 0 when the return status != NO_ERROR.
@@ -368,7 +375,7 @@
     //  NO_ERROR    Success, buffer->mFrameCount > 0.
     //  WOULD_BLOCK No frames are available.
     //  NO_INIT     Shared memory is corrupt.
-    virtual status_t    obtainBuffer(Buffer* buffer);
+    virtual status_t    obtainBuffer(Buffer* buffer, bool ackFlush = false);
 
     // Release (some of) the frames last obtained.
     // On entry, buffer->mFrameCount should have the number of frames to release,
@@ -435,7 +442,7 @@
 public:
     virtual size_t      framesReady();
     virtual void        framesReadyIsCalledByMultipleThreads();
-    virtual status_t    obtainBuffer(Buffer* buffer);
+    virtual status_t    obtainBuffer(Buffer* buffer, bool ackFlush);
     virtual void        releaseBuffer(Buffer* buffer);
     virtual void        tallyUnderrunFrames(uint32_t frameCount);
     virtual uint32_t    getUnderrunFrames() const { return 0; }

include/private/media/StaticAudioTrackState.h

diff --git a/include/private/media/StaticAudioTrackState.h b/include/private/media/StaticAudioTrackState.h
index 46a5946..d483061 100644
--- a/include/private/media/StaticAudioTrackState.h
+++ b/include/private/media/StaticAudioTrackState.h
@@ -25,9 +25,13 @@
 // state is wrapped by a SingleStateQueue.
 struct StaticAudioTrackState {
     // do not define constructors, destructors, or virtual methods
-    size_t  mLoopStart;
-    size_t  mLoopEnd;
-    int     mLoopCount;
+
+    // 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

libvideoeditor/lvpp/Android.mk

diff --git a/libvideoeditor/lvpp/Android.mk b/libvideoeditor/lvpp/Android.mk
index 2286827..860d351 100755
--- a/libvideoeditor/lvpp/Android.mk
+++ b/libvideoeditor/lvpp/Android.mk
@@ -71,7 +71,6 @@
     $(TOP)/frameworks/av/media/libstagefright \
     $(TOP)/frameworks/av/media/libstagefright/include \
     $(TOP)/frameworks/av/media/libstagefright/rtsp \
-    $(call include-path-for, corecg graphics) \
     $(TOP)/frameworks/av/libvideoeditor/osal/inc \
     $(TOP)/frameworks/av/libvideoeditor/vss/common/inc \
     $(TOP)/frameworks/av/libvideoeditor/vss/mcs/inc \

libvideoeditor/lvpp/PreviewPlayer.cpp

diff --git a/libvideoeditor/lvpp/PreviewPlayer.cpp b/libvideoeditor/lvpp/PreviewPlayer.cpp
index 2bd9f84..b36fe0a 100755
--- a/libvideoeditor/lvpp/PreviewPlayer.cpp
+++ b/libvideoeditor/lvpp/PreviewPlayer.cpp
@@ -21,6 +21,7 @@
 
 #include <binder/IPCThreadState.h>
 #include <binder/IServiceManager.h>
+#include <media/IMediaHTTPService.h>
 #include <media/IMediaPlayerService.h>
 #include <media/stagefright/DataSource.h>
 #include <media/stagefright/MediaBuffer.h>
@@ -1160,7 +1161,8 @@
     sp<DataSource> dataSource;
     sp<MediaExtractor> extractor;
 
-    dataSource = DataSource::CreateFromURI(mUri.string(), NULL);
+    dataSource = DataSource::CreateFromURI(
+            NULL /* httpService */, mUri.string(), NULL);
 
     if (dataSource == NULL) {
         return UNKNOWN_ERROR;

libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp

diff --git a/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp b/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp
index 176f8e9..cb4b23e 100755
--- a/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp
+++ b/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp
@@ -671,8 +671,9 @@
 
 
                         M4OSA_Void* ptr;
-                        ptr = (M4OSA_Void*)((unsigned int)mInputBuffer->data() +
-                        mInputBuffer->range_offset());
+                        ptr = reinterpret_cast<M4OSA_Void*>(
+                                reinterpret_cast<uintptr_t>(mInputBuffer->data()) +
+                                mInputBuffer->range_offset());
 
                         M4OSA_UInt32 len = mInputBuffer->range_length();
                         M4OSA_Context fp = M4OSA_NULL;
@@ -700,8 +701,8 @@
                              mBGAudioPCMFileOriginalSeekPoint <=
                               (mBGAudioPCMFileTrimmedLength - len)) {
 
-                            ALOGV("Checking mBGAudioPCMFileHandle %d",
-                                (unsigned int)mBGAudioPCMFileHandle);
+                            ALOGV("Checking mBGAudioPCMFileHandle %p",
+                                  mBGAudioPCMFileHandle);
 
                             if (mBGAudioPCMFileHandle != M4OSA_NULL) {
                                 ALOGV("fillBuffer seeking file to %lld",

libvideoeditor/lvpp/VideoEditorPlayer.cpp

diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.cpp b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
index 5aeba4f..f9c3879 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.cpp
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
@@ -57,6 +57,7 @@
 
 
 status_t VideoEditorPlayer::setDataSource(
+        const sp<IMediaHTTPService> &httpService,
         const char *url, const KeyedVector<String8, String8> *headers) {
     ALOGI("setDataSource('%s')", url);
     if (headers != NULL) {
@@ -585,4 +586,11 @@
     return mSessionId;
 }
 
+uint32_t VideoEditorPlayer::VeAudioOutput::getSampleRate() const {
+    if (mMsecsPerFrame == 0) {
+        return 0;
+    }
+    return (uint32_t)(1.e3 / mMsecsPerFrame);
+}
+
 }  // namespace android

libvideoeditor/lvpp/VideoEditorPlayer.h

diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.h b/libvideoeditor/lvpp/VideoEditorPlayer.h
index ab6d731..781e4bc 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.h
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.h
@@ -48,6 +48,7 @@
         virtual status_t        getPosition(uint32_t *position) const;
         virtual status_t        getFramesWritten(uint32_t*) const;
         virtual int             getSessionId() const;
+        virtual uint32_t        getSampleRate() const;
 
         virtual status_t        open(
                 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
@@ -62,6 +63,7 @@
         virtual void            pause();
         virtual void            close();
         void setAudioStreamType(audio_stream_type_t streamType) { mStreamType = streamType; }
+        virtual audio_stream_type_t getAudioStreamType() const { return mStreamType; }
                 void            setVolume(float left, float right);
         virtual status_t        dump(int fd,const Vector<String16>& args) const;
 
@@ -96,6 +98,7 @@
     virtual status_t initCheck();
 
     virtual status_t setDataSource(
+            const sp<IMediaHTTPService> &httpService,
             const char *url, const KeyedVector<String8, String8> *headers);
 
     virtual status_t setDataSource(int fd, int64_t offset, int64_t length);

libvideoeditor/lvpp/VideoEditorPreviewController.cpp

diff --git a/libvideoeditor/lvpp/VideoEditorPreviewController.cpp b/libvideoeditor/lvpp/VideoEditorPreviewController.cpp
index 149c4ea..953f35a 100755
--- a/libvideoeditor/lvpp/VideoEditorPreviewController.cpp
+++ b/libvideoeditor/lvpp/VideoEditorPreviewController.cpp
@@ -19,6 +19,7 @@
 #include <utils/Log.h>
 
 #include <gui/Surface.h>
+#include <media/IMediaHTTPService.h>
 
 #include "VideoEditorAudioPlayer.h"
 #include "PreviewRenderer.h"
@@ -967,7 +968,8 @@
     ALOGV("preparePlayer: instance %d file %d", playerInstance, index);
 
     const char* fileName = (const char*) pController->mClipList[index]->pFile;
-    pController->mVePlayer[playerInstance]->setDataSource(fileName, NULL);
+    pController->mVePlayer[playerInstance]->setDataSource(
+            NULL /* httpService */, fileName, NULL);
 
     ALOGV("preparePlayer: setDataSource instance %s",
      (const char *)pController->mClipList[index]->pFile);
@@ -1248,7 +1250,7 @@
         case MEDIA_SET_VIDEO_SIZE:
             ALOGV("MEDIA_SET_VIDEO_SIZE; New video size %d x %d", ext1, ext2);
             break;
-        case 0xAAAAAAAA:
+        case static_cast<int>(0xAAAAAAAA):
             ALOGV("VIDEO PLAYBACK ALMOST over, prepare next player");
             // Select next player and prepare it
             // If there is a clip after this one
@@ -1268,7 +1270,7 @@
                 }
             }
             break;
-        case 0xBBBBBBBB:
+        case static_cast<int>(0xBBBBBBBB):
         {
             ALOGV("VIDEO PLAYBACK, Update Overlay");
             int overlayIndex = ext2;

libvideoeditor/lvpp/VideoEditorSRC.cpp

diff --git a/libvideoeditor/lvpp/VideoEditorSRC.cpp b/libvideoeditor/lvpp/VideoEditorSRC.cpp
index 36d0812..6beabfa 100755
--- a/libvideoeditor/lvpp/VideoEditorSRC.cpp
+++ b/libvideoeditor/lvpp/VideoEditorSRC.cpp
@@ -284,7 +284,7 @@
 
 
 void VideoEditorSRC::releaseBuffer(AudioBufferProvider::Buffer *pBuffer) {
-    ALOGV("releaseBuffer: %p", pBuffers);
+    ALOGV("releaseBuffer: %p", pBuffer);
     free(pBuffer->raw);
     pBuffer->raw = NULL;
     pBuffer->frameCount = 0;

libvideoeditor/osal/inc/M4OSA_Clock.h

diff --git a/libvideoeditor/osal/inc/M4OSA_Clock.h b/libvideoeditor/osal/inc/M4OSA_Clock.h
index db753a5..52ea696 100755
--- a/libvideoeditor/osal/inc/M4OSA_Clock.h
+++ b/libvideoeditor/osal/inc/M4OSA_Clock.h
@@ -21,7 +21,7 @@
  ************************************************************************
 */
 
-#ifndef M4OSA_CLOCH_H
+#ifndef M4OSA_CLOCK_H
 #define M4OSA_CLOCK_H
 
 #include "M4OSA_Types.h"

libvideoeditor/osal/inc/M4OSA_Error.h

diff --git a/libvideoeditor/osal/inc/M4OSA_Error.h b/libvideoeditor/osal/inc/M4OSA_Error.h
index 4d59529..75c3177 100755
--- a/libvideoeditor/osal/inc/M4OSA_Error.h
+++ b/libvideoeditor/osal/inc/M4OSA_Error.h
@@ -57,7 +57,7 @@
   * @arg coreID: (IN) [M4OSA_UInt32] CoreID to put in the error code
   * @arg errorID: (IN) [M4OSA_UInt32] ErrorID to put in the error code*/
 #define M4OSA_ERR_CREATE(severity, coreID, errorID)\
-   (M4OSA_Int32)((((M4OSA_UInt32)severity)<<30)+((((M4OSA_UInt32)coreID)&0x003FFF)<<16)+(((M4OSA_UInt32)errorID)&0x00FFFF))
+   (M4OSA_UInt32)((((M4OSA_UInt32)severity)<<30)+((((M4OSA_UInt32)coreID)&0x003FFF)<<16)+(((M4OSA_UInt32)errorID)&0x00FFFF))
 
 /** This macro extracts the 3 fields from the error:
   * @arg error: (IN) [M4OSA_ERR] Error code

libvideoeditor/vss/src/M4DECODER_Null.c

diff --git a/libvideoeditor/vss/src/M4DECODER_Null.c b/libvideoeditor/vss/src/M4DECODER_Null.c
index a0dad30..ce260e5 100755
--- a/libvideoeditor/vss/src/M4DECODER_Null.c
+++ b/libvideoeditor/vss/src/M4DECODER_Null.c
@@ -210,7 +210,7 @@
             break;

 

         case M4DECODER_kOptionID_EnableYuvWithEffect:

-            pStreamContext->bYuvWithEffectSet = (M4OSA_Bool)pValue;

+            pStreamContext->bYuvWithEffectSet = (M4OSA_Bool)(intptr_t)pValue;

             break;

 

         case M4DECODER_kOptionID_YuvWithEffectNonContiguous:

libvideoeditor/vss/stagefrightshells/src/Android.mk

diff --git a/libvideoeditor/vss/stagefrightshells/src/Android.mk b/libvideoeditor/vss/stagefrightshells/src/Android.mk
index e30b85d..9188942 100755
--- a/libvideoeditor/vss/stagefrightshells/src/Android.mk
+++ b/libvideoeditor/vss/stagefrightshells/src/Android.mk
@@ -33,7 +33,6 @@
     $(TOP)/frameworks/av/media/libstagefright \
     $(TOP)/frameworks/av/media/libstagefright/include \
     $(TOP)/frameworks/av/media/libstagefright/rtsp \
-    $(call include-path-for, corecg graphics) \
     $(TOP)/frameworks/av/libvideoeditor/lvpp \
     $(TOP)/frameworks/av/libvideoeditor/osal/inc \
     $(TOP)/frameworks/av/libvideoeditor/vss/inc \

libvideoeditor/vss/stagefrightshells/src/VideoEditor3gpReader.cpp

diff --git a/libvideoeditor/vss/stagefrightshells/src/VideoEditor3gpReader.cpp b/libvideoeditor/vss/stagefrightshells/src/VideoEditor3gpReader.cpp
index 3c8915a..99cf9ec 100755
--- a/libvideoeditor/vss/stagefrightshells/src/VideoEditor3gpReader.cpp
+++ b/libvideoeditor/vss/stagefrightshells/src/VideoEditor3gpReader.cpp
@@ -776,16 +776,16 @@
         case M4READER_kOptionID_SetOsaFileReaderFctsPtr:
         break;
 
-        case M4READER_3GP_kOptionID_AudioOnly:
+        case static_cast<M4OSA_OptionID>(M4READER_3GP_kOptionID_AudioOnly):
         break;
 
-        case M4READER_3GP_kOptionID_VideoOnly:
+        case static_cast<M4OSA_OptionID>(M4READER_3GP_kOptionID_VideoOnly):
         break;
 
-        case M4READER_3GP_kOptionID_FastOpenMode:
+        case static_cast<M4OSA_OptionID>(M4READER_3GP_kOptionID_FastOpenMode):
         break;
 
-        case M4READER_kOptionID_MaxMetadataSize:
+        case static_cast<M4OSA_OptionID>(M4READER_kOptionID_MaxMetadataSize):
         break;
 
         default:

libvideoeditor/vss/stagefrightshells/src/VideoEditorAudioDecoder.cpp

diff --git a/libvideoeditor/vss/stagefrightshells/src/VideoEditorAudioDecoder.cpp b/libvideoeditor/vss/stagefrightshells/src/VideoEditorAudioDecoder.cpp
index 9b35d07..e4c7ea1 100755
--- a/libvideoeditor/vss/stagefrightshells/src/VideoEditorAudioDecoder.cpp
+++ b/libvideoeditor/vss/stagefrightshells/src/VideoEditorAudioDecoder.cpp
@@ -809,7 +809,7 @@
     pDecoderContext = (VideoEditorAudioDecoder_Context*)pContext;
 
     switch( optionID ) {
-        case M4AD_kOptionID_UserParam:
+        case static_cast<M4OSA_UInt32>(M4AD_kOptionID_UserParam):
             ALOGV("VideoEditorAudioDecodersetOption UserParam is not supported");
             err = M4ERR_NOT_IMPLEMENTED;
             break;

libvideoeditor/vss/stagefrightshells/src/VideoEditorMp3Reader.cpp

diff --git a/libvideoeditor/vss/stagefrightshells/src/VideoEditorMp3Reader.cpp b/libvideoeditor/vss/stagefrightshells/src/VideoEditorMp3Reader.cpp
index af53c54..2e0d05d 100755
--- a/libvideoeditor/vss/stagefrightshells/src/VideoEditorMp3Reader.cpp
+++ b/libvideoeditor/vss/stagefrightshells/src/VideoEditorMp3Reader.cpp
@@ -334,8 +334,8 @@
         (VideoEditorMp3Reader_Context*)context;
     M4OSA_ERR err = M4NO_ERROR;
 
-    ALOGV("VideoEditorMp3Reader_Context begin: optionId: %d Value: %d ",
-        (int)optionId,(int)pValue);
+    ALOGV("VideoEditorMp3Reader_Context begin: optionId: %u Value: %p ",
+          optionId, pValue);
 
     M4OSA_DEBUG_IF1((M4OSA_NULL == pReaderContext), M4ERR_PARAMETER,
         "invalid context pointer");

libvideoeditor/vss/stagefrightshells/src/VideoEditorUtils.cpp

diff --git a/libvideoeditor/vss/stagefrightshells/src/VideoEditorUtils.cpp b/libvideoeditor/vss/stagefrightshells/src/VideoEditorUtils.cpp
index 5a7237d..d264a2e 100755
--- a/libvideoeditor/vss/stagefrightshells/src/VideoEditorUtils.cpp
+++ b/libvideoeditor/vss/stagefrightshells/src/VideoEditorUtils.cpp
@@ -84,17 +84,17 @@
         LOG1("displayMetaData kKeyBitRate %d", int32Data);
     }
     if (meta->findData(kKeyESDS, &type, &data, &size)) {
-        LOG1("displayMetaData kKeyESDS type=%d size=%d", type, size);
+        LOG1("displayMetaData kKeyESDS type=%d size=%zu", type, size);
     }
     if (meta->findData(kKeyAVCC, &type, &data, &size)) {
-        LOG1("displayMetaData kKeyAVCC data=0x%X type=%d size=%d",
+        LOG1("displayMetaData kKeyAVCC data=0x%X type=%d size=%zu",
             *((unsigned int*)data), type, size);
     }
     if (meta->findData(kKeyVorbisInfo, &type, &data, &size)) {
-        LOG1("displayMetaData kKeyVorbisInfo type=%d size=%d", type, size);
+        LOG1("displayMetaData kKeyVorbisInfo type=%d size=%zu", type, size);
     }
     if (meta->findData(kKeyVorbisBooks, &type, &data, &size)) {
-        LOG1("displayMetaData kKeyVorbisBooks type=%d size=%d", type, size);
+        LOG1("displayMetaData kKeyVorbisBooks type=%d size=%zu", type, size);
     }
     if (meta->findInt32(kKeyWantsNALFragments, &int32Data)) {
         LOG1("displayMetaData kKeyWantsNALFragments %d", int32Data);
@@ -115,7 +115,7 @@
         LOG1("displayMetaData kKeyColorFormat %d", int32Data);
     }
     if (meta->findPointer(kKeyPlatformPrivate, &ptr)) {
-        LOG1("displayMetaData kKeyPlatformPrivate pointer=0x%x", (int32_t) ptr);
+        LOG1("displayMetaData kKeyPlatformPrivate pointer=%p", ptr);
     }
     if (meta->findCString(kKeyDecoderComponent, &charData)) {
         LOG1("displayMetaData kKeyDecoderComponent %s", charData);
@@ -151,7 +151,7 @@
         LOG1("displayMetaData kKeyYear %s", charData);
     }
     if (meta->findData(kKeyAlbumArt, &type, &data, &size)) {
-        LOG1("displayMetaData kKeyAlbumArt type=%d size=%d", type, size);
+        LOG1("displayMetaData kKeyAlbumArt type=%d size=%zu", type, size);
     }
     if (meta->findCString(kKeyAlbumArtMIME, &charData)) {
         LOG1("displayMetaData kKeyAlbumArtMIME %s", charData);
@@ -277,7 +277,7 @@
     }
 
     if (size < 4) {
-        ALOGE("Codec specific data length too short: %d", size);
+        ALOGE("Codec specific data length too short: %zu", size);
         return ERROR_MALFORMED;
     }
 
@@ -286,7 +286,7 @@
         // 2 bytes for each of the parameter set length field
         // plus the 7 bytes for the header
         if (size < 4 + 7) {
-            ALOGE("Codec specific data length too short: %d", size);
+            ALOGE("Codec specific data length too short: %zu", size);
             return ERROR_MALFORMED;
         }
 
@@ -355,7 +355,7 @@
         }
 
         if (nSeqParamSets > 0x1F) {
-            ALOGE("Too many seq parameter sets (%d) found", nSeqParamSets);
+            ALOGE("Too many seq parameter sets (%zu) found", nSeqParamSets);
             return ERROR_MALFORMED;
         }
     }
@@ -368,7 +368,7 @@
             return ERROR_MALFORMED;
         }
         if (nPicParamSets > 0xFF) {
-            ALOGE("Too many pic parameter sets (%d) found", nPicParamSets);
+            ALOGE("Too many pic parameter sets (%zu) found", nPicParamSets);
             return ERROR_MALFORMED;
         }
     }

media/libeffects/downmix/Android.mk

diff --git a/media/libeffects/downmix/Android.mk b/media/libeffects/downmix/Android.mk
index 5d0a87c..2bb6dbe 100644
--- a/media/libeffects/downmix/Android.mk
+++ b/media/libeffects/downmix/Android.mk
@@ -13,7 +13,7 @@
 
 LOCAL_MODULE_TAGS := optional
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
 
 ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
 LOCAL_LDLIBS += -ldl

media/libeffects/downmix/EffectDownmix.c

diff --git a/media/libeffects/downmix/EffectDownmix.c b/media/libeffects/downmix/EffectDownmix.c
index f17a6e8..d25dc9b 100644
--- a/media/libeffects/downmix/EffectDownmix.c
+++ b/media/libeffects/downmix/EffectDownmix.c
@@ -61,13 +61,13 @@
 // This is the only symbol that needs to be exported
 __attribute__ ((visibility ("default")))
 audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = {
-    tag : AUDIO_EFFECT_LIBRARY_TAG,
-    version : EFFECT_LIBRARY_API_VERSION,
-    name : "Downmix Library",
-    implementor : "The Android Open Source Project",
-    create_effect : DownmixLib_Create,
-    release_effect : DownmixLib_Release,
-    get_descriptor : DownmixLib_GetDescriptor,
+    .tag = AUDIO_EFFECT_LIBRARY_TAG,
+    .version = EFFECT_LIBRARY_API_VERSION,
+    .name = "Downmix Library",
+    .implementor = "The Android Open Source Project",
+    .create_effect = DownmixLib_Create,
+    .release_effect = DownmixLib_Release,
+    .get_descriptor = DownmixLib_GetDescriptor,
 };
 
 
@@ -629,7 +629,9 @@
         return -EINVAL;
     }
 
-    memcpy(&pDwmModule->config, pConfig, sizeof(effect_config_t));
+    if (&pDwmModule->config != pConfig) {
+        memcpy(&pDwmModule->config, pConfig, sizeof(effect_config_t));
+    }
 
     if (init) {
         pDownmixer->type = DOWNMIX_TYPE_FOLD;
@@ -707,7 +709,7 @@
 
       case DOWNMIX_PARAM_TYPE:
         if (size != sizeof(downmix_type_t)) {
-            ALOGE("Downmix_setParameter(DOWNMIX_PARAM_TYPE) invalid size %d, should be %d",
+            ALOGE("Downmix_setParameter(DOWNMIX_PARAM_TYPE) invalid size %zu, should be %zu",
                     size, sizeof(downmix_type_t));
             return -EINVAL;
         }
@@ -760,7 +762,7 @@
 
     case DOWNMIX_PARAM_TYPE:
       if (*pSize < sizeof(int16_t)) {
-          ALOGE("Downmix_getParameter invalid parameter size %d for DOWNMIX_PARAM_TYPE", *pSize);
+          ALOGE("Downmix_getParameter invalid parameter size %zu for DOWNMIX_PARAM_TYPE", *pSize);
           return -EINVAL;
       }
       pValue16 = (int16_t *)pValue;

media/libeffects/factory/Android.mk

diff --git a/media/libeffects/factory/Android.mk b/media/libeffects/factory/Android.mk
index 60a6ce5..a932af7 100644
--- a/media/libeffects/factory/Android.mk
+++ b/media/libeffects/factory/Android.mk
@@ -9,7 +9,6 @@
 LOCAL_SHARED_LIBRARIES := \
 	libcutils liblog
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)
 LOCAL_MODULE:= libeffects
 
 LOCAL_SHARED_LIBRARIES += libdl

media/libeffects/factory/EffectsFactory.c

diff --git a/media/libeffects/factory/EffectsFactory.c b/media/libeffects/factory/EffectsFactory.c
index f8d6041..6d30d64 100644
--- a/media/libeffects/factory/EffectsFactory.c
+++ b/media/libeffects/factory/EffectsFactory.c
@@ -368,27 +368,21 @@
     }
     if (e1 == NULL) {
         ret = -ENOENT;
-        pthread_mutex_unlock(&gLibLock);
         goto exit;
     }
 
     // release effect in library
     if (fx->lib == NULL) {
         ALOGW("EffectRelease() fx %p library already unloaded", handle);
-        pthread_mutex_unlock(&gLibLock);
     } else {
         pthread_mutex_lock(&fx->lib->lock);
-        // Releasing the gLibLock here as the list access is over as the
-        // effect is removed from the list.
-        // If the gLibLock is not released, we will have a deadlock situation
-        // since we call the sub effect release inside the EffectRelease of Proxy
-        pthread_mutex_unlock(&gLibLock);
         fx->lib->desc->release_effect(fx->subItfe);
         pthread_mutex_unlock(&fx->lib->lock);
     }
     free(fx);
 
 exit:
+    pthread_mutex_unlock(&gLibLock);
     return ret;
 }
 
@@ -404,8 +398,8 @@
 // is pointed by the first argument. It searches the gSubEffectList for the
 // matching uuid and then copies the corresponding sub effect descriptors
 // to the inout param
-int EffectGetSubEffects(const effect_uuid_t *uuid,
-                        effect_descriptor_t *pDescriptors, size_t size)
+int EffectGetSubEffects(const effect_uuid_t *uuid, sub_effect_entry_t **pSube,
+                        size_t size)
 {
    ALOGV("EffectGetSubEffects() UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X"
           "%02X\n",uuid->timeLow, uuid->timeMid, uuid->timeHiAndVersion,
@@ -413,8 +407,7 @@
           uuid->node[3],uuid->node[4],uuid->node[5]);
 
    // Check if the size of the desc buffer is large enough for 2 subeffects
-   if ((uuid == NULL) || (pDescriptors == NULL) ||
-       (size < 2*sizeof(effect_descriptor_t))) {
+   if ((uuid == NULL) || (pSube == NULL) || (size < 2)) {
        ALOGW("NULL pointer or insufficient memory. Cannot query subeffects");
        return -EINVAL;
    }
@@ -432,11 +425,10 @@
            list_elem_t *subefx = e->sub_elem;
            while (subefx != NULL) {
                subeffect = (sub_effect_entry_t*)subefx->object;
-               d = (effect_descriptor_t*)(subeffect->object);
-               pDescriptors[count++] = *d;
+               pSube[count++] = subeffect;
                subefx = subefx->next;
            }
-           ALOGV("EffectGetSubEffects end - copied the sub effect descriptors");
+           ALOGV("EffectGetSubEffects end - copied the sub effect structures");
            return count;
        }
        e = e->next;

media/libeffects/factory/EffectsFactory.h

diff --git a/media/libeffects/factory/EffectsFactory.h b/media/libeffects/factory/EffectsFactory.h
index 147ff18..560b485 100644
--- a/media/libeffects/factory/EffectsFactory.h
+++ b/media/libeffects/factory/EffectsFactory.h
@@ -20,7 +20,7 @@
 #include <cutils/log.h>
 #include <pthread.h>
 #include <dirent.h>
-#include <media/EffectsFactoryApi.h>
+#include <hardware/audio_effect.h>
 
 #if __cplusplus
 extern "C" {
@@ -66,6 +66,32 @@
     void *object;
 } sub_effect_entry_t;
 
+
+////////////////////////////////////////////////////////////////////////////////
+//
+//    Function:       EffectGetSubEffects
+//
+//    Description:    Returns the descriptors of the sub effects of the effect
+//                    whose uuid is pointed to by first argument.
+//
+//    Input:
+//          pEffectUuid:    pointer to the effect uuid.
+//          size:           max number of sub_effect_entry_t * in pSube.
+//
+//    Input/Output:
+//          pSube:          address where to return the sub effect structures.
+//    Output:
+//        returned value:    0          successful operation.
+//                          -ENODEV     factory failed to initialize
+//                          -EINVAL     invalid pEffectUuid or pDescriptor
+//                          -ENOENT     no effect with this uuid found
+//        *pDescriptor:     updated with the sub effect descriptors.
+//
+////////////////////////////////////////////////////////////////////////////////
+int EffectGetSubEffects(const effect_uuid_t *pEffectUuid,
+                        sub_effect_entry_t **pSube,
+                        size_t size);
+
 #if __cplusplus
 }  // extern "C"
 #endif

media/libeffects/loudness/Android.mk

diff --git a/media/libeffects/loudness/Android.mk b/media/libeffects/loudness/Android.mk
index dcb7b27..edf964e 100644
--- a/media/libeffects/loudness/Android.mk
+++ b/media/libeffects/loudness/Android.mk
@@ -14,7 +14,7 @@
 	liblog \
 	libstlport
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
 LOCAL_MODULE:= libldnhncr
 
 LOCAL_C_INCLUDES := \

media/libeffects/loudness/EffectLoudnessEnhancer.cpp

diff --git a/media/libeffects/loudness/EffectLoudnessEnhancer.cpp b/media/libeffects/loudness/EffectLoudnessEnhancer.cpp
index dfc25db..3c2b320 100644
--- a/media/libeffects/loudness/EffectLoudnessEnhancer.cpp
+++ b/media/libeffects/loudness/EffectLoudnessEnhancer.cpp
@@ -56,8 +56,7 @@
     int32_t mTargetGainmB;// target gain in mB
     // in this implementation, there is no coupling between the compression on the left and right
     // channels
-    le_fx::AdaptiveDynamicRangeCompression* mCompressorL;
-    le_fx::AdaptiveDynamicRangeCompression* mCompressorR;
+    le_fx::AdaptiveDynamicRangeCompression* mCompressor;
 };
 
 //
@@ -68,11 +67,10 @@
 {
     ALOGV("  > LE_reset(%p)", pContext);
 
-    if ((pContext->mCompressorL != NULL) && (pContext->mCompressorR != NULL)) {
+    if (pContext->mCompressor != NULL) {
         float targetAmp = pow(10, pContext->mTargetGainmB/2000.0f); // mB to linear amplification
         ALOGV("LE_reset(): Target gain=%dmB <=> factor=%.2fX", pContext->mTargetGainmB, targetAmp);
-        pContext->mCompressorL->Initialize(targetAmp, pContext->mConfig.inputCfg.samplingRate);
-        pContext->mCompressorR->Initialize(targetAmp, pContext->mConfig.inputCfg.samplingRate);
+        pContext->mCompressor->Initialize(targetAmp, pContext->mConfig.inputCfg.samplingRate);
     } else {
         ALOGE("LE_reset(%p): null compressors, can't apply target gain", pContext);
     }
@@ -176,13 +174,9 @@
     float targetAmp = pow(10, pContext->mTargetGainmB/2000.0f); // mB to linear amplification
     ALOGV("LE_init(): Target gain=%dmB <=> factor=%.2fX", pContext->mTargetGainmB, targetAmp);
 
-    if (pContext->mCompressorL == NULL) {
-        pContext->mCompressorL = new le_fx::AdaptiveDynamicRangeCompression();
-        pContext->mCompressorL->Initialize(targetAmp, pContext->mConfig.inputCfg.samplingRate);
-    }
-    if (pContext->mCompressorR == NULL) {
-        pContext->mCompressorR = new le_fx::AdaptiveDynamicRangeCompression();
-        pContext->mCompressorR->Initialize(targetAmp, pContext->mConfig.inputCfg.samplingRate);
+    if (pContext->mCompressor == NULL) {
+        pContext->mCompressor = new le_fx::AdaptiveDynamicRangeCompression();
+        pContext->mCompressor->Initialize(targetAmp, pContext->mConfig.inputCfg.samplingRate);
     }
 
     LE_setConfig(pContext, &pContext->mConfig);
@@ -215,8 +209,7 @@
     pContext->mItfe = &gLEInterface;
     pContext->mState = LOUDNESS_ENHANCER_STATE_UNINITIALIZED;
 
-    pContext->mCompressorL = NULL;
-    pContext->mCompressorR = NULL;
+    pContext->mCompressor = NULL;
     ret = LE_init(pContext);
     if (ret < 0) {
         ALOGW("LELib_Create() init failed");
@@ -242,13 +235,9 @@
         return -EINVAL;
     }
     pContext->mState = LOUDNESS_ENHANCER_STATE_UNINITIALIZED;
-    if (pContext->mCompressorL != NULL) {
-        delete pContext->mCompressorL;
-        pContext->mCompressorL = NULL;
-    }
-    if (pContext->mCompressorR != NULL) {
-        delete pContext->mCompressorR;
-        pContext->mCompressorR = NULL;
+    if (pContext->mCompressor != NULL) {
+        delete pContext->mCompressor;
+        pContext->mCompressor = NULL;
     }
     delete pContext;
 
@@ -293,11 +282,14 @@
     //ALOGV("LE about to process %d samples", inBuffer->frameCount);
     uint16_t inIdx;
     float inputAmp = pow(10, pContext->mTargetGainmB/2000.0f);
+    float leftSample, rightSample;
     for (inIdx = 0 ; inIdx < inBuffer->frameCount ; inIdx++) {
-        inBuffer->s16[2*inIdx] = pContext->mCompressorL->Compress(
-                inputAmp * (float)inBuffer->s16[2*inIdx]);
-        inBuffer->s16[2*inIdx +1] = pContext->mCompressorR->Compress(
-                inputAmp * (float)inBuffer->s16[2*inIdx +1]);
+        // makeup gain is applied on the input of the compressor
+        leftSample  = inputAmp * (float)inBuffer->s16[2*inIdx];
+        rightSample = inputAmp * (float)inBuffer->s16[2*inIdx +1];
+        pContext->mCompressor->Compress(&leftSample, &rightSample);
+        inBuffer->s16[2*inIdx]    = (int16_t) leftSample;
+        inBuffer->s16[2*inIdx +1] = (int16_t) rightSample;
     }
 
     if (inBuffer->raw != outBuffer->raw) {
@@ -461,13 +453,13 @@
 // This is the only symbol that needs to be exported
 __attribute__ ((visibility ("default")))
 audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = {
-    tag : AUDIO_EFFECT_LIBRARY_TAG,
-    version : EFFECT_LIBRARY_API_VERSION,
-    name : "Loudness Enhancer Library",
-    implementor : "The Android Open Source Project",
-    create_effect : LELib_Create,
-    release_effect : LELib_Release,
-    get_descriptor : LELib_GetDescriptor,
+    .tag = AUDIO_EFFECT_LIBRARY_TAG,
+    .version = EFFECT_LIBRARY_API_VERSION,
+    .name = "Loudness Enhancer Library",
+    .implementor = "The Android Open Source Project",
+    .create_effect = LELib_Create,
+    .release_effect = LELib_Release,
+    .get_descriptor = LELib_GetDescriptor,
 };
 
 }; // extern "C"

media/libeffects/loudness/dsp/core/dynamic_range_compression-inl.h

diff --git a/media/libeffects/loudness/dsp/core/dynamic_range_compression-inl.h b/media/libeffects/loudness/dsp/core/dynamic_range_compression-inl.h
index fed8c2a..da75ceb 100644
--- a/media/libeffects/loudness/dsp/core/dynamic_range_compression-inl.h
+++ b/media/libeffects/loudness/dsp/core/dynamic_range_compression-inl.h
@@ -35,7 +35,7 @@
     float target_gain) {
   const float decibel = target_gain_to_knee_threshold_.Interpolate(
         target_gain);
-  ALOGE("set_knee_threshold_via_target_gain: decibel =%.3f", decibel);
+  ALOGV("set_knee_threshold_via_target_gain: decibel =%.3fdB", decibel);
   set_knee_threshold(decibel);
 }
 

media/libeffects/loudness/dsp/core/dynamic_range_compression.cpp

diff --git a/media/libeffects/loudness/dsp/core/dynamic_range_compression.cpp b/media/libeffects/loudness/dsp/core/dynamic_range_compression.cpp
index 2bbd043..7bd068e 100644
--- a/media/libeffects/loudness/dsp/core/dynamic_range_compression.cpp
+++ b/media/libeffects/loudness/dsp/core/dynamic_range_compression.cpp
@@ -102,5 +102,40 @@
   return x;
 }
 
+void AdaptiveDynamicRangeCompression::Compress(float *x1, float *x2) {
+  // Taking the maximum amplitude of both channels
+  const float max_abs_x = std::max(std::fabs(*x1),
+    std::max(std::fabs(*x2), kMinLogAbsValue));
+  const float max_abs_x_dB = math::fast_log(max_abs_x);
+  // Subtract Threshold from log-encoded input to get the amount of overshoot
+  const float overshoot = max_abs_x_dB - knee_threshold_;
+  // Hard half-wave rectifier
+  const float rect = std::max(overshoot, 0.0f);
+  // Multiply rectified overshoot with slope
+  const float cv = rect * slope_;
+  const float prev_state = state_;
+  if (cv <= state_) {
+    state_ = alpha_attack_ * state_ + (1.0f - alpha_attack_) * cv;
+  } else {
+    state_ = alpha_release_ * state_ + (1.0f - alpha_release_) * cv;
+  }
+  compressor_gain_ *=
+      math::ExpApproximationViaTaylorExpansionOrder5(state_ - prev_state);
+  *x1 *= compressor_gain_;
+  if (*x1 > kFixedPointLimit) {
+    *x1 = kFixedPointLimit;
+  }
+  if (*x1 < -kFixedPointLimit) {
+    *x1 = -kFixedPointLimit;
+  }
+  *x2 *= compressor_gain_;
+  if (*x2 > kFixedPointLimit) {
+    *x2 = kFixedPointLimit;
+  }
+  if (*x2 < -kFixedPointLimit) {
+    *x2 = -kFixedPointLimit;
+  }
+}
+
 }  // namespace le_fx
 

media/libeffects/loudness/dsp/core/dynamic_range_compression.h

diff --git a/media/libeffects/loudness/dsp/core/dynamic_range_compression.h b/media/libeffects/loudness/dsp/core/dynamic_range_compression.h
index 4c015df..2821a78 100644
--- a/media/libeffects/loudness/dsp/core/dynamic_range_compression.h
+++ b/media/libeffects/loudness/dsp/core/dynamic_range_compression.h
@@ -55,6 +55,9 @@
   // log(.) and exp(.).
   float Compress(float x);
 
+  // Stereo channel version of the compressor
+  void Compress(float *x1, float *x2);
+
   // This version is slower than Compress(.) but faster than CompressSlow(.)
   float CompressNormalSpeed(float x);
 

media/libeffects/lvm/lib/Eq/src/LVEQNB_Init.c

diff --git a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Init.c b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Init.c
index c4767a8..e01c1c5 100644
--- a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Init.c
+++ b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Init.c
@@ -25,6 +25,7 @@
 #include "LVEQNB.h"
 #include "LVEQNB_Private.h"
 #include "InstAlloc.h"
+#include <string.h> /* For memset */
 
 /****************************************************************************************/
 /*                                                                                      */

media/libeffects/lvm/wrapper/Android.mk

diff --git a/media/libeffects/lvm/wrapper/Android.mk b/media/libeffects/lvm/wrapper/Android.mk
index f1af389..68ba34c 100644
--- a/media/libeffects/lvm/wrapper/Android.mk
+++ b/media/libeffects/lvm/wrapper/Android.mk
@@ -13,7 +13,7 @@
 
 LOCAL_MODULE:= libbundlewrapper
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
 
 LOCAL_STATIC_LIBRARIES += libmusicbundle
 
@@ -42,7 +42,7 @@
 
 LOCAL_MODULE:= libreverbwrapper
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
 
 LOCAL_STATIC_LIBRARIES += libreverb
 

media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp

diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
index 85232e7..58d7767 100644
--- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
+++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
@@ -138,22 +138,22 @@
 int  BassBoost_setParameter    (EffectContext *pContext, void *pParam, void *pValue);
 int  BassBoost_getParameter    (EffectContext *pContext,
                                void           *pParam,
-                               size_t         *pValueSize,
+                               uint32_t       *pValueSize,
                                void           *pValue);
 int  Virtualizer_setParameter  (EffectContext *pContext, void *pParam, void *pValue);
 int  Virtualizer_getParameter  (EffectContext *pContext,
                                void           *pParam,
-                               size_t         *pValueSize,
+                               uint32_t       *pValueSize,
                                void           *pValue);
 int  Equalizer_setParameter    (EffectContext *pContext, void *pParam, void *pValue);
 int  Equalizer_getParameter    (EffectContext *pContext,
                                 void          *pParam,
-                                size_t        *pValueSize,
+                                uint32_t      *pValueSize,
                                 void          *pValue);
 int  Volume_setParameter       (EffectContext *pContext, void *pParam, void *pValue);
 int  Volume_getParameter       (EffectContext *pContext,
                                 void          *pParam,
-                                size_t        *pValueSize,
+                                uint32_t      *pValueSize,
                                 void          *pValue);
 int Effect_setEnabled(EffectContext *pContext, bool enabled);
 
@@ -1758,7 +1758,7 @@
 
 int BassBoost_getParameter(EffectContext     *pContext,
                            void              *pParam,
-                           size_t            *pValueSize,
+                           uint32_t          *pValueSize,
                            void              *pValue){
     int status = 0;
     int32_t *pParamTemp = (int32_t *)pParam;
@@ -1876,7 +1876,7 @@
 
 int Virtualizer_getParameter(EffectContext        *pContext,
                              void                 *pParam,
-                             size_t               *pValueSize,
+                             uint32_t             *pValueSize,
                              void                 *pValue){
     int status = 0;
     int32_t *pParamTemp = (int32_t *)pParam;
@@ -1994,7 +1994,7 @@
 //----------------------------------------------------------------------------
 int Equalizer_getParameter(EffectContext     *pContext,
                            void              *pParam,
-                           size_t            *pValueSize,
+                           uint32_t          *pValueSize,
                            void              *pValue){
     int status = 0;
     int bMute = 0;
@@ -2252,7 +2252,7 @@
 
 int Volume_getParameter(EffectContext     *pContext,
                         void              *pParam,
-                        size_t            *pValueSize,
+                        uint32_t          *pValueSize,
                         void              *pValue){
     int status = 0;
     int bMute = 0;
@@ -2830,7 +2830,7 @@
 
                 p->status = android::BassBoost_getParameter(pContext,
                                                             p->data,
-                                                            (size_t  *)&p->vsize,
+                                                            &p->vsize,
                                                             p->data + voffset);
 
                 *replySize = sizeof(effect_param_t) + voffset + p->vsize;
@@ -2860,8 +2860,8 @@
                 int voffset = ((p->psize - 1) / sizeof(int32_t) + 1) * sizeof(int32_t);
 
                 p->status = android::Virtualizer_getParameter(pContext,
-                                                             (void *)p->data,
-                                                             (size_t  *)&p->vsize,
+                                                              (void *)p->data,
+                                                              &p->vsize,
                                                               p->data + voffset);
 
                 *replySize = sizeof(effect_param_t) + voffset + p->vsize;
@@ -2925,7 +2925,7 @@
 
                 p->status = android::Volume_getParameter(pContext,
                                                          (void *)p->data,
-                                                         (size_t  *)&p->vsize,
+                                                         &p->vsize,
                                                          p->data + voffset);
 
                 *replySize = sizeof(effect_param_t) + voffset + p->vsize;
@@ -3267,13 +3267,13 @@
 // This is the only symbol that needs to be exported
 __attribute__ ((visibility ("default")))
 audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = {
-    tag : AUDIO_EFFECT_LIBRARY_TAG,
-    version : EFFECT_LIBRARY_API_VERSION,
-    name : "Effect Bundle Library",
-    implementor : "NXP Software Ltd.",
-    create_effect : android::EffectCreate,
-    release_effect : android::EffectRelease,
-    get_descriptor : android::EffectGetDescriptor,
+    .tag = AUDIO_EFFECT_LIBRARY_TAG,
+    .version = EFFECT_LIBRARY_API_VERSION,
+    .name = "Effect Bundle Library",
+    .implementor = "NXP Software Ltd.",
+    .create_effect = android::EffectCreate,
+    .release_effect = android::EffectRelease,
+    .get_descriptor = android::EffectGetDescriptor,
 };
 
 }

media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp

diff --git a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
index 8a96212..0367302 100644
--- a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
+++ b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
@@ -2149,13 +2149,13 @@
 // This is the only symbol that needs to be exported
 __attribute__ ((visibility ("default")))
 audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = {
-    tag : AUDIO_EFFECT_LIBRARY_TAG,
-    version : EFFECT_LIBRARY_API_VERSION,
-    name : "Reverb Library",
-    implementor : "NXP Software Ltd.",
-    create_effect : android::EffectCreate,
-    release_effect : android::EffectRelease,
-    get_descriptor : android::EffectGetDescriptor,
+    .tag = AUDIO_EFFECT_LIBRARY_TAG,
+    .version = EFFECT_LIBRARY_API_VERSION,
+    .name = "Reverb Library",
+    .implementor = "NXP Software Ltd.",
+    .create_effect = android::EffectCreate,
+    .release_effect = android::EffectRelease,
+    .get_descriptor = android::EffectGetDescriptor,
 };
 
 }

media/libeffects/preprocessing/Android.mk

diff --git a/media/libeffects/preprocessing/Android.mk b/media/libeffects/preprocessing/Android.mk
index c344352..9e8cb83 100644
--- a/media/libeffects/preprocessing/Android.mk
+++ b/media/libeffects/preprocessing/Android.mk
@@ -5,7 +5,7 @@
 
 LOCAL_MODULE:= libaudiopreprocessing
 LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
 
 LOCAL_SRC_FILES:= \
     PreProcessing.cpp

media/libeffects/preprocessing/PreProcessing.cpp

diff --git a/media/libeffects/preprocessing/PreProcessing.cpp b/media/libeffects/preprocessing/PreProcessing.cpp
index 25586e8..c56ff72 100644
--- a/media/libeffects/preprocessing/PreProcessing.cpp
+++ b/media/libeffects/preprocessing/PreProcessing.cpp
@@ -1233,8 +1233,8 @@
             if (session->framesIn < session->frameCount) {
                 return 0;
             }
-            size_t frIn = session->framesIn;
-            size_t frOut = session->apmFrameCount;
+            spx_uint32_t frIn = session->framesIn;
+            spx_uint32_t frOut = session->apmFrameCount;
             if (session->inChannelCount == 1) {
                 speex_resampler_process_int(session->inResampler,
                                             0,
@@ -1290,8 +1290,8 @@
         }
 
         if (session->outResampler != NULL) {
-            size_t frIn = session->apmFrameCount;
-            size_t frOut = session->frameCount;
+            spx_uint32_t frIn = session->apmFrameCount;
+            spx_uint32_t frOut = session->frameCount;
             if (session->inChannelCount == 1) {
                 speex_resampler_process_int(session->outResampler,
                                     0,
@@ -1754,8 +1754,8 @@
             if (session->framesRev < session->frameCount) {
                 return 0;
             }
-            size_t frIn = session->framesRev;
-            size_t frOut = session->apmFrameCount;
+            spx_uint32_t frIn = session->framesRev;
+            spx_uint32_t frOut = session->apmFrameCount;
             if (session->inChannelCount == 1) {
                 speex_resampler_process_int(session->revResampler,
                                             0,
@@ -1892,13 +1892,13 @@
 // This is the only symbol that needs to be exported
 __attribute__ ((visibility ("default")))
 audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = {
-    tag : AUDIO_EFFECT_LIBRARY_TAG,
-    version : EFFECT_LIBRARY_API_VERSION,
-    name : "Audio Preprocessing Library",
-    implementor : "The Android Open Source Project",
-    create_effect : PreProcessingLib_Create,
-    release_effect : PreProcessingLib_Release,
-    get_descriptor : PreProcessingLib_GetDescriptor
+    .tag = AUDIO_EFFECT_LIBRARY_TAG,
+    .version = EFFECT_LIBRARY_API_VERSION,
+    .name = "Audio Preprocessing Library",
+    .implementor = "The Android Open Source Project",
+    .create_effect = PreProcessingLib_Create,
+    .release_effect = PreProcessingLib_Release,
+    .get_descriptor = PreProcessingLib_GetDescriptor
 };
 
 }; // extern "C"

media/libeffects/proxy/Android.mk

diff --git a/media/libeffects/proxy/Android.mk b/media/libeffects/proxy/Android.mk
index 01b3be1..b438796 100644
--- a/media/libeffects/proxy/Android.mk
+++ b/media/libeffects/proxy/Android.mk
@@ -15,7 +15,7 @@
 LOCAL_PATH:= $(call my-dir)
 include $(CLEAR_VARS)
 LOCAL_MODULE:= libeffectproxy
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
 LOCAL_MODULE_TAGS := optional
 
 
@@ -28,7 +28,8 @@
 
 LOCAL_C_INCLUDES := \
         system/media/audio_effects/include \
-        bionic/libc/include
+        bionic/libc/include \
+        frameworks/av/media/libeffects/factory
 
 include $(BUILD_SHARED_LIBRARY)
 

media/libeffects/proxy/EffectProxy.cpp

diff --git a/media/libeffects/proxy/EffectProxy.cpp b/media/libeffects/proxy/EffectProxy.cpp
index b3304b7..62d3fd3 100644
--- a/media/libeffects/proxy/EffectProxy.cpp
+++ b/media/libeffects/proxy/EffectProxy.cpp
@@ -30,9 +30,10 @@
 // This is a dummy proxy descriptor just to return to Factory during the initial
 // GetDescriptor call. Later in the factory, it is replaced with the
 // SW sub effect descriptor
+// proxy UUID af8da7e0-2ca1-11e3-b71d-0002a5d5c51b
 const effect_descriptor_t gProxyDescriptor = {
         EFFECT_UUID_INITIALIZER, // type
-        EFFECT_UUID_INITIALIZER, // uuid
+        {0xaf8da7e0, 0x2ca1, 0x11e3, 0xb71d, { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b }}, // uuid
         EFFECT_CONTROL_API_VERSION, //version of effect control API
         (EFFECT_FLAG_TYPE_INSERT | EFFECT_FLAG_INSERT_LAST |
          EFFECT_FLAG_VOLUME_CTRL), // effect capability flags
@@ -55,6 +56,8 @@
                            effect_handle_t  *pHandle) {
 
     effect_descriptor_t* desc;
+    audio_effect_library_t** aeli;
+    sub_effect_entry_t** sube;
     EffectContext* pContext;
     if (pHandle == NULL || uuid == NULL) {
         ALOGE("EffectProxyCreate() called with NULL pointer");
@@ -73,31 +76,52 @@
 
     // Get the HW and SW sub effect descriptors from the effects factory
     desc = new effect_descriptor_t[SUB_FX_COUNT];
+    aeli = new audio_effect_library_t*[SUB_FX_COUNT];
+    sube = new sub_effect_entry_t*[SUB_FX_COUNT];
+    pContext->sube = new sub_effect_entry_t*[SUB_FX_COUNT];
     pContext->desc = new effect_descriptor_t[SUB_FX_COUNT];
-    int retValue = EffectGetSubEffects(uuid, desc,
-                                sizeof(effect_descriptor_t) * SUB_FX_COUNT);
+    pContext->aeli = new audio_effect_library_t*[SUB_FX_COUNT];
+    int retValue = EffectGetSubEffects(uuid, sube, SUB_FX_COUNT);
     // EffectGetSubEffects returns the number of sub-effects copied.
     if (retValue != SUB_FX_COUNT) {
        ALOGE("EffectCreate() could not get the sub effects");
-       delete desc;
-       delete pContext->desc;
+       delete[] sube;
+       delete[] desc;
+       delete[] aeli;
+       delete[] pContext->sube;
+       delete[] pContext->desc;
+       delete[] pContext->aeli;
        return -EINVAL;
     }
     // Check which is the HW descriptor and copy the descriptors
     // to the Context desc array
     // Also check if there is only one HW and one SW descriptor.
     // HW descriptor alone has the HW_TUNNEL flag.
+    desc[0] = *(effect_descriptor_t*)(sube[0])->object;
+    desc[1] = *(effect_descriptor_t*)(sube[1])->object;
+    aeli[0] = sube[0]->lib->desc;
+    aeli[1] = sube[1]->lib->desc;
     if ((desc[0].flags & EFFECT_FLAG_HW_ACC_TUNNEL) &&
        !(desc[1].flags & EFFECT_FLAG_HW_ACC_TUNNEL)) {
+        pContext->sube[SUB_FX_OFFLOAD] = sube[0];
         pContext->desc[SUB_FX_OFFLOAD] = desc[0];
+        pContext->aeli[SUB_FX_OFFLOAD] = aeli[0];
+        pContext->sube[SUB_FX_HOST] = sube[1];
         pContext->desc[SUB_FX_HOST] = desc[1];
+        pContext->aeli[SUB_FX_HOST] = aeli[1];
     }
     else if ((desc[1].flags & EFFECT_FLAG_HW_ACC_TUNNEL) &&
              !(desc[0].flags & EFFECT_FLAG_HW_ACC_TUNNEL)) {
+        pContext->sube[SUB_FX_HOST] = sube[0];
         pContext->desc[SUB_FX_HOST] = desc[0];
+        pContext->aeli[SUB_FX_HOST] = aeli[0];
+        pContext->sube[SUB_FX_OFFLOAD] = sube[1];
         pContext->desc[SUB_FX_OFFLOAD] = desc[1];
+        pContext->aeli[SUB_FX_OFFLOAD] = aeli[1];
     }
-    delete desc;
+    delete[] desc;
+    delete[] aeli;
+    delete[] sube;
 #if (LOG_NDEBUG == 0)
     effect_uuid_t uuid_print = pContext->desc[SUB_FX_HOST].uuid;
     ALOGV("EffectCreate() UUID of HOST: %08X-%04X-%04X-%04X-%02X%02X%02X%02X"
@@ -127,13 +151,15 @@
         return -EINVAL;
     }
     ALOGV("EffectRelease");
-    delete pContext->desc;
+    delete[] pContext->desc;
     free(pContext->replyData);
 
     if (pContext->eHandle[SUB_FX_HOST])
-       EffectRelease(pContext->eHandle[SUB_FX_HOST]);
+       pContext->aeli[SUB_FX_HOST]->release_effect(pContext->eHandle[SUB_FX_HOST]);
     if (pContext->eHandle[SUB_FX_OFFLOAD])
-       EffectRelease(pContext->eHandle[SUB_FX_OFFLOAD]);
+       pContext->aeli[SUB_FX_OFFLOAD]->release_effect(pContext->eHandle[SUB_FX_OFFLOAD]);
+    delete[] pContext->aeli;
+    delete[] pContext->sube;
     delete pContext;
     pContext = NULL;
     return 0;
@@ -186,7 +212,8 @@
     }
     if (pContext->eHandle[SUB_FX_HOST] == NULL) {
         ALOGV("Effect_command() Calling HOST EffectCreate");
-        status = EffectCreate(&pContext->desc[SUB_FX_HOST].uuid,
+        status = pContext->aeli[SUB_FX_HOST]->create_effect(
+                              &pContext->desc[SUB_FX_HOST].uuid,
                               pContext->sessionId, pContext->ioId,
                               &(pContext->eHandle[SUB_FX_HOST]));
         if (status != NO_ERROR || (pContext->eHandle[SUB_FX_HOST] == NULL)) {
@@ -196,11 +223,13 @@
     }
     if (pContext->eHandle[SUB_FX_OFFLOAD] == NULL) {
         ALOGV("Effect_command() Calling OFFLOAD EffectCreate");
-        status = EffectCreate(&pContext->desc[SUB_FX_OFFLOAD].uuid,
+        status = pContext->aeli[SUB_FX_OFFLOAD]->create_effect(
+                              &pContext->desc[SUB_FX_OFFLOAD].uuid,
                               pContext->sessionId, pContext->ioId,
                               &(pContext->eHandle[SUB_FX_OFFLOAD]));
         if (status != NO_ERROR || (pContext->eHandle[SUB_FX_OFFLOAD] == NULL)) {
             ALOGV("Effect_command() Error creating HW effect");
+            pContext->eHandle[SUB_FX_OFFLOAD] = NULL;
             // Do not return error here as SW effect is created
             // Return error if the CMD_OFFLOAD sends the index as OFFLOAD
         }
@@ -232,11 +261,17 @@
         // Update the DSP wrapper with the new ioHandle.
         // Pass the OFFLOAD command to the wrapper.
         // The DSP wrapper needs to handle this CMD
-        if (pContext->eHandle[SUB_FX_OFFLOAD])
-            status = (*pContext->eHandle[SUB_FX_OFFLOAD])->command(
-                             pContext->eHandle[SUB_FX_OFFLOAD], cmdCode, cmdSize,
-                             pCmdData, replySize, pReplyData);
-        return status;
+        if (pContext->eHandle[SUB_FX_OFFLOAD]) {
+            ALOGV("Effect_command: Calling OFFLOAD command");
+            return (*pContext->eHandle[SUB_FX_OFFLOAD])->command(
+                           pContext->eHandle[SUB_FX_OFFLOAD], cmdCode, cmdSize,
+                           pCmdData, replySize, pReplyData);
+        }
+        *(int*)pReplyData = NO_ERROR;
+        ALOGV("Effect_command OFFLOAD return 0, replyData %d",
+                                                *(int*)pReplyData);
+
+        return NO_ERROR;
     }
 
     int index = pContext->index;
@@ -328,11 +363,11 @@
 
 __attribute__ ((visibility ("default")))
 audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = {
-    tag : AUDIO_EFFECT_LIBRARY_TAG,
-    version : EFFECT_LIBRARY_API_VERSION,
-    name : "Effect Proxy",
-    implementor : "AOSP",
-    create_effect : android::EffectProxyCreate,
-    release_effect : android::EffectProxyRelease,
-    get_descriptor : android::EffectProxyGetDescriptor,
+    .tag = AUDIO_EFFECT_LIBRARY_TAG,
+    .version = EFFECT_LIBRARY_API_VERSION,
+    .name = "Effect Proxy",
+    .implementor = "AOSP",
+    .create_effect = android::EffectProxyCreate,
+    .release_effect = android::EffectProxyRelease,
+    .get_descriptor = android::EffectProxyGetDescriptor,
 };

media/libeffects/proxy/EffectProxy.h

diff --git a/media/libeffects/proxy/EffectProxy.h b/media/libeffects/proxy/EffectProxy.h
index acbe17e..046b93e 100644
--- a/media/libeffects/proxy/EffectProxy.h
+++ b/media/libeffects/proxy/EffectProxy.h
@@ -16,6 +16,8 @@
 
 #include <hardware/audio.h>
 #include <hardware/audio_effect.h>
+#include "EffectsFactory.h"
+
 namespace android {
 enum {
     SUB_FX_HOST,       // Index of HOST in the descriptor and handle arrays
@@ -62,7 +64,9 @@
 
 struct EffectContext {
   const struct effect_interface_s  *common_itfe; // Holds the itfe of the Proxy
+  sub_effect_entry_t** sube;                     // Points to the sub effects
   effect_descriptor_t*  desc;                    // Points to the sub effect descriptors
+  audio_effect_library_t**  aeli;                // Points to the sub effect aeli
   effect_handle_t       eHandle[SUB_FX_COUNT];   // The effect handles of the sub effects
   int                   index;       // The index that is currently active - HOST or OFFLOAD
   int32_t               sessionId;   // The sessiond in which the effect is created.

media/libeffects/testlibs/Android.mk_

diff --git a/media/libeffects/testlibs/Android.mk_ b/media/libeffects/testlibs/Android.mk_
index 2954908..672ebba 100644
--- a/media/libeffects/testlibs/Android.mk_
+++ b/media/libeffects/testlibs/Android.mk_
@@ -11,7 +11,7 @@
 LOCAL_SHARED_LIBRARIES := \
 	libcutils
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
 LOCAL_MODULE:= libreverbtest
 
 ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
@@ -47,7 +47,7 @@
 LOCAL_SHARED_LIBRARIES := \
 	libcutils
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
 LOCAL_MODULE:= libequalizertest
 
 ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)

media/libeffects/visualizer/Android.mk

diff --git a/media/libeffects/visualizer/Android.mk b/media/libeffects/visualizer/Android.mk
index e196eb2..c92c543 100644
--- a/media/libeffects/visualizer/Android.mk
+++ b/media/libeffects/visualizer/Android.mk
@@ -13,11 +13,10 @@
 	liblog \
 	libdl
 
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
 LOCAL_MODULE:= libvisualizer
 
 LOCAL_C_INCLUDES := \
-	$(call include-path-for, graphics corecg) \
 	$(call include-path-for, audio-effects)
 
 

media/libeffects/visualizer/EffectVisualizer.cpp

diff --git a/media/libeffects/visualizer/EffectVisualizer.cpp b/media/libeffects/visualizer/EffectVisualizer.cpp
index dc403ab..5bdaa03 100644
--- a/media/libeffects/visualizer/EffectVisualizer.cpp
+++ b/media/libeffects/visualizer/EffectVisualizer.cpp
@@ -544,56 +544,57 @@
         break;
 
 
-    case VISUALIZER_CMD_CAPTURE:
-        if (pReplyData == NULL || *replySize != pContext->mCaptureSize) {
-            ALOGV("VISUALIZER_CMD_CAPTURE() error *replySize %d pContext->mCaptureSize %d",
-                    *replySize, pContext->mCaptureSize);
+    case VISUALIZER_CMD_CAPTURE: {
+        int32_t captureSize = pContext->mCaptureSize;
+        if (pReplyData == NULL || *replySize != captureSize) {
+            ALOGV("VISUALIZER_CMD_CAPTURE() error *replySize %d captureSize %d",
+                    *replySize, captureSize);
             return -EINVAL;
         }
         if (pContext->mState == VISUALIZER_STATE_ACTIVE) {
-            int32_t latencyMs = pContext->mLatency;
             const uint32_t deltaMs = Visualizer_getDeltaTimeMsFromUpdatedTime(pContext);
-            latencyMs -= deltaMs;
-            if (latencyMs < 0) {
-                latencyMs = 0;
-            }
-            const uint32_t deltaSmpl = pContext->mConfig.inputCfg.samplingRate * latencyMs / 1000;
-
-            int32_t capturePoint = pContext->mCaptureIdx - pContext->mCaptureSize - deltaSmpl;
-            int32_t captureSize = pContext->mCaptureSize;
-            if (capturePoint < 0) {
-                int32_t size = -capturePoint;
-                if (size > captureSize) {
-                    size = captureSize;
-                }
-                memcpy(pReplyData,
-                       pContext->mCaptureBuf + CAPTURE_BUF_SIZE + capturePoint,
-                       size);
-                pReplyData = (char *)pReplyData + size;
-                captureSize -= size;
-                capturePoint = 0;
-            }
-            memcpy(pReplyData,
-                   pContext->mCaptureBuf + capturePoint,
-                   captureSize);
-
 
             // if audio framework has stopped playing audio although the effect is still
             // active we must clear the capture buffer to return silence
             if ((pContext->mLastCaptureIdx == pContext->mCaptureIdx) &&
-                    (pContext->mBufferUpdateTime.tv_sec != 0)) {
-                if (deltaMs > MAX_STALL_TIME_MS) {
+                    (pContext->mBufferUpdateTime.tv_sec != 0) &&
+                    (deltaMs > MAX_STALL_TIME_MS)) {
                     ALOGV("capture going to idle");
                     pContext->mBufferUpdateTime.tv_sec = 0;
-                    memset(pReplyData, 0x80, pContext->mCaptureSize);
+                    memset(pReplyData, 0x80, captureSize);
+            } else {
+                int32_t latencyMs = pContext->mLatency;
+                latencyMs -= deltaMs;
+                if (latencyMs < 0) {
+                    latencyMs = 0;
                 }
+                const uint32_t deltaSmpl =
+                    pContext->mConfig.inputCfg.samplingRate * latencyMs / 1000;
+                int32_t capturePoint = pContext->mCaptureIdx - captureSize - deltaSmpl;
+
+                if (capturePoint < 0) {
+                    int32_t size = -capturePoint;
+                    if (size > captureSize) {
+                        size = captureSize;
+                    }
+                    memcpy(pReplyData,
+                           pContext->mCaptureBuf + CAPTURE_BUF_SIZE + capturePoint,
+                           size);
+                    pReplyData = (char *)pReplyData + size;
+                    captureSize -= size;
+                    capturePoint = 0;
+                }
+                memcpy(pReplyData,
+                       pContext->mCaptureBuf + capturePoint,
+                       captureSize);
             }
+
             pContext->mLastCaptureIdx = pContext->mCaptureIdx;
         } else {
-            memset(pReplyData, 0x80, pContext->mCaptureSize);
+            memset(pReplyData, 0x80, captureSize);
         }
 
-        break;
+        } break;
 
     case VISUALIZER_CMD_MEASURE: {
         uint16_t peakU16 = 0;
@@ -678,13 +679,13 @@
 // This is the only symbol that needs to be exported
 __attribute__ ((visibility ("default")))
 audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = {
-    tag : AUDIO_EFFECT_LIBRARY_TAG,
-    version : EFFECT_LIBRARY_API_VERSION,
-    name : "Visualizer Library",
-    implementor : "The Android Open Source Project",
-    create_effect : VisualizerLib_Create,
-    release_effect : VisualizerLib_Release,
-    get_descriptor : VisualizerLib_GetDescriptor,
+    .tag = AUDIO_EFFECT_LIBRARY_TAG,
+    .version = EFFECT_LIBRARY_API_VERSION,
+    .name = "Visualizer Library",
+    .implementor = "The Android Open Source Project",
+    .create_effect = VisualizerLib_Create,
+    .release_effect = VisualizerLib_Release,
+    .get_descriptor = VisualizerLib_GetDescriptor,
 };
 
 }; // extern "C"

media/libmedia/Android.mk

diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index 56e7787..f3770e4 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -25,6 +25,8 @@
     AudioRecord.cpp \
     AudioSystem.cpp \
     mediaplayer.cpp \
+    IMediaHTTPConnection.cpp \
+    IMediaHTTPService.cpp \
     IMediaLogService.cpp \
     IMediaPlayerService.cpp \
     IMediaPlayerClient.cpp \
@@ -44,7 +46,7 @@
     IAudioPolicyService.cpp \
     MediaScanner.cpp \
     MediaScannerClient.cpp \
-    autodetect.cpp \
+    CharacterEncodingDetector.cpp \
     IMediaDeathNotifier.cpp \
     MediaProfiles.cpp \
     IEffect.cpp \
@@ -58,26 +60,34 @@
 
 LOCAL_SRC_FILES += ../libnbaio/roundup.c
 
-# for <cutils/atomic-inline.h>
-LOCAL_CFLAGS += -DANDROID_SMP=$(if $(findstring true,$(TARGET_CPU_SMP)),1,0)
-LOCAL_SRC_FILES += SingleStateQueue.cpp
-LOCAL_CFLAGS += -DSINGLE_STATE_QUEUE_INSTANTIATIONS='"SingleStateQueueInstantiations.cpp"'
-# Consider a separate a library for SingleStateQueueInstantiations.
-
 LOCAL_SHARED_LIBRARIES := \
-	libui liblog libcutils libutils libbinder libsonivox libicuuc libexpat \
+	libui liblog libcutils libutils libbinder libsonivox libicuuc libicui18n libexpat \
         libcamera_client libstagefright_foundation \
-        libgui libdl libaudioutils
+        libgui libdl libaudioutils libnbaio
+
+LOCAL_STATIC_LIBRARIES += libinstantssq
 
 LOCAL_WHOLE_STATIC_LIBRARY := libmedia_helper
 
 LOCAL_MODULE:= libmedia
 
 LOCAL_C_INCLUDES := \
-    $(call include-path-for, graphics corecg) \
     $(TOP)/frameworks/native/include/media/openmax \
     external/icu4c/common \
+    external/icu4c/i18n \
     $(call include-path-for, audio-effects) \
     $(call include-path-for, audio-utils)
 
 include $(BUILD_SHARED_LIBRARY)
+
+include $(CLEAR_VARS)
+
+# for <cutils/atomic-inline.h>
+LOCAL_CFLAGS += -DANDROID_SMP=$(if $(findstring true,$(TARGET_CPU_SMP)),1,0)
+LOCAL_SRC_FILES += SingleStateQueue.cpp
+LOCAL_CFLAGS += -DSINGLE_STATE_QUEUE_INSTANTIATIONS='"SingleStateQueueInstantiations.cpp"'
+
+LOCAL_MODULE := libinstantssq
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_STATIC_LIBRARY)

media/libmedia/AudioEffect.cpp

diff --git a/media/libmedia/AudioEffect.cpp b/media/libmedia/AudioEffect.cpp
index 8dfffb3..35f6557 100644
--- a/media/libmedia/AudioEffect.cpp
+++ b/media/libmedia/AudioEffect.cpp
@@ -380,9 +380,9 @@
 }
 
 void AudioEffect::commandExecuted(uint32_t cmdCode,
-                                  uint32_t cmdSize,
+                                  uint32_t cmdSize __unused,
                                   void *cmdData,
-                                  uint32_t replySize,
+                                  uint32_t replySize __unused,
                                   void *replyData)
 {
     if (cmdData == NULL || replyData == NULL) {

media/libmedia/AudioRecord.cpp

diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index c5a7777..961b0a2 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -41,37 +41,29 @@
         return BAD_VALUE;
     }
 
-    // default to 0 in case of error
-    *frameCount = 0;
-
-    size_t size = 0;
+    size_t size;
     status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size);
     if (status != NO_ERROR) {
-        ALOGE("AudioSystem could not query the input buffer size; status %d", status);
-        return NO_INIT;
+        ALOGE("AudioSystem could not query the input buffer size for sampleRate %u, format %#x, "
+              "channelMask %#x; status %d", sampleRate, format, channelMask, status);
+        return status;
     }
 
-    if (size == 0) {
-        ALOGE("Unsupported configuration: sampleRate %u, format %d, channelMask %#x",
+    // We double the size of input buffer for ping pong use of record buffer.
+    // Assumes audio_is_linear_pcm(format)
+    if ((*frameCount = (size * 2) / (popcount(channelMask) * audio_bytes_per_sample(format))) == 0) {
+        ALOGE("Unsupported configuration: sampleRate %u, format %#x, channelMask %#x",
             sampleRate, format, channelMask);
         return BAD_VALUE;
     }
 
-    // We double the size of input buffer for ping pong use of record buffer.
-    size <<= 1;
-
-    // Assumes audio_is_linear_pcm(format)
-    uint32_t channelCount = popcount(channelMask);
-    size /= channelCount * audio_bytes_per_sample(format);
-
-    *frameCount = size;
     return NO_ERROR;
 }
 
 // ---------------------------------------------------------------------------
 
 AudioRecord::AudioRecord()
-    : mStatus(NO_INIT), mSessionId(0),
+    : mStatus(NO_INIT), mSessionId(AUDIO_SESSION_ALLOCATE),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT)
 {
 }
@@ -81,14 +73,14 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        int frameCount,
+        size_t frameCount,
         callback_t cbf,
         void* user,
-        int notificationFrames,
+        uint32_t notificationFrames,
         int sessionId,
         transfer_type transferType,
-        audio_input_flags_t flags)
-    : mStatus(NO_INIT), mSessionId(0),
+        audio_input_flags_t flags __unused)
+    : mStatus(NO_INIT), mSessionId(AUDIO_SESSION_ALLOCATE),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
       mPreviousSchedulingGroup(SP_DEFAULT),
       mProxy(NULL)
@@ -110,12 +102,10 @@
             mAudioRecordThread->requestExitAndWait();
             mAudioRecordThread.clear();
         }
-        if (mAudioRecord != 0) {
-            mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
-            mAudioRecord.clear();
-        }
+        mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
+        mAudioRecord.clear();
         IPCThreadState::self()->flushCommands();
-        AudioSystem::releaseAudioSessionId(mSessionId);
+        AudioSystem::releaseAudioSessionId(mSessionId, -1);
     }
 }
 
@@ -124,15 +114,20 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        int frameCountInt,
+        size_t frameCount,
         callback_t cbf,
         void* user,
-        int notificationFrames,
+        uint32_t notificationFrames,
         bool threadCanCallJava,
         int sessionId,
         transfer_type transferType,
         audio_input_flags_t flags)
 {
+    ALOGV("set(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
+          "notificationFrames %u, sessionId %d, transferType %d, flags %#x",
+          inputSource, sampleRate, format, channelMask, frameCount, notificationFrames,
+          sessionId, transferType, flags);
+
     switch (transferType) {
     case TRANSFER_DEFAULT:
         if (cbf == NULL || threadCanCallJava) {
@@ -156,23 +151,15 @@
     }
     mTransfer = transferType;
 
-    // FIXME "int" here is legacy and will be replaced by size_t later
-    if (frameCountInt < 0) {
-        ALOGE("Invalid frame count %d", frameCountInt);
-        return BAD_VALUE;
-    }
-    size_t frameCount = frameCountInt;
-
-    ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask,
-            frameCount);
-
     AutoMutex lock(mLock);
 
+    // invariant that mAudioRecord != 0 is true only after set() returns successfully
     if (mAudioRecord != 0) {
         ALOGE("Track already in use");
         return INVALID_OPERATION;
     }
 
+    // handle default values first.
     if (inputSource == AUDIO_SOURCE_DEFAULT) {
         inputSource = AUDIO_SOURCE_MIC;
     }
@@ -191,12 +178,12 @@
 
     // validate parameters
     if (!audio_is_valid_format(format)) {
-        ALOGE("Invalid format %d", format);
+        ALOGE("Invalid format %#x", format);
         return BAD_VALUE;
     }
     // Temporary restriction: AudioFlinger currently supports 16-bit PCM only
     if (format != AUDIO_FORMAT_PCM_16_BIT) {
-        ALOGE("Format %d is not supported", format);
+        ALOGE("Format %#x is not supported", format);
         return BAD_VALUE;
     }
     mFormat = format;
@@ -209,15 +196,19 @@
     uint32_t channelCount = popcount(channelMask);
     mChannelCount = channelCount;
 
-    // Assumes audio_is_linear_pcm(format), else sizeof(uint8_t)
-    mFrameSize = channelCount * audio_bytes_per_sample(format);
+    if (audio_is_linear_pcm(format)) {
+        mFrameSize = channelCount * audio_bytes_per_sample(format);
+    } else {
+        mFrameSize = sizeof(uint8_t);
+    }
 
     // validate framecount
-    size_t minFrameCount = 0;
+    size_t minFrameCount;
     status_t status = AudioRecord::getMinFrameCount(&minFrameCount,
             sampleRate, format, channelMask);
     if (status != NO_ERROR) {
-        ALOGE("getMinFrameCount() failed; status %d", status);
+        ALOGE("getMinFrameCount() failed for sampleRate %u, format %#x, channelMask %#x; status %d",
+                sampleRate, format, channelMask, status);
         return status;
     }
     ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);
@@ -228,12 +219,13 @@
         ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
         return BAD_VALUE;
     }
-    mFrameCount = frameCount;
+    // mFrameCount is initialized in openRecord_l
+    mReqFrameCount = frameCount;
 
     mNotificationFramesReq = notificationFrames;
     mNotificationFramesAct = 0;
 
-    if (sessionId == 0 ) {
+    if (sessionId == AUDIO_SESSION_ALLOCATE) {
         mSessionId = AudioSystem::newAudioSessionId();
     } else {
         mSessionId = sessionId;
@@ -241,26 +233,27 @@
     ALOGV("set(): mSessionId %d", mSessionId);
 
     mFlags = flags;
-
-    // create the IAudioRecord
-    status = openRecord_l(0 /*epoch*/);
-    if (status) {
-        return status;
-    }
+    mCbf = cbf;
 
     if (cbf != NULL) {
         mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava);
         mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
     }
 
+    // create the IAudioRecord
+    status = openRecord_l(0 /*epoch*/);
+
+    if (status != NO_ERROR) {
+        if (mAudioRecordThread != 0) {
+            mAudioRecordThread->requestExit();   // see comment in AudioRecord.h
+            mAudioRecordThread->requestExitAndWait();
+            mAudioRecordThread.clear();
+        }
+        return status;
+    }
+
     mStatus = NO_ERROR;
-
-    // Update buffer size in case it has been limited by AudioFlinger during track creation
-    mFrameCount = mCblk->frameCount_;
-
     mActive = false;
-    mCbf = cbf;
-    mRefreshRemaining = true;
     mUserData = user;
     // TODO: add audio hardware input latency here
     mLatency = (1000*mFrameCount) / sampleRate;
@@ -268,7 +261,7 @@
     mMarkerReached = false;
     mNewPosition = 0;
     mUpdatePeriod = 0;
-    AudioSystem::acquireAudioSessionId(mSessionId);
+    AudioSystem::acquireAudioSessionId(mSessionId, -1);
     mSequence = 1;
     mObservedSequence = mSequence;
     mInOverrun = false;
@@ -289,6 +282,9 @@
 
     // reset current position as seen by client to 0
     mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition());
+    // force refresh of remaining frames by processAudioBuffer() as last
+    // read before stop could be partial.
+    mRefreshRemaining = true;
 
     mNewPosition = mProxy->getPosition() + mUpdatePeriod;
     int32_t flags = android_atomic_acquire_load(&mCblk->mFlags);
@@ -352,6 +348,7 @@
 
 status_t AudioRecord::setMarkerPosition(uint32_t marker)
 {
+    // The only purpose of setting marker position is to get a callback
     if (mCbf == NULL) {
         return INVALID_OPERATION;
     }
@@ -377,6 +374,7 @@
 
 status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
 {
+    // The only purpose of setting position update period is to get a callback
     if (mCbf == NULL) {
         return INVALID_OPERATION;
     }
@@ -412,7 +410,7 @@
     return NO_ERROR;
 }
 
-unsigned int AudioRecord::getInputFramesLost() const
+uint32_t AudioRecord::getInputFramesLost() const
 {
     // no need to check mActive, because if inactive this will return 0, which is what we want
     return AudioSystem::getInputFramesLost(getInput());
@@ -430,92 +428,142 @@
         return NO_INIT;
     }
 
-    IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;
-    pid_t tid = -1;
+    // Fast tracks must be at the primary _output_ [sic] sampling rate,
+    // because there is currently no concept of a primary input sampling rate
+    uint32_t afSampleRate = AudioSystem::getPrimaryOutputSamplingRate();
+    if (afSampleRate == 0) {
+        ALOGW("getPrimaryOutputSamplingRate failed");
+    }
 
     // Client can only express a preference for FAST.  Server will perform additional tests.
-    // The only supported use case for FAST is callback transfer mode.
+    if ((mFlags & AUDIO_INPUT_FLAG_FAST) && !(
+            // use case: callback transfer mode
+            (mTransfer == TRANSFER_CALLBACK) &&
+            // matching sample rate
+            (mSampleRate == afSampleRate))) {
+        ALOGW("AUDIO_INPUT_FLAG_FAST denied by client");
+        // once denied, do not request again if IAudioRecord is re-created
+        mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
+    }
+
+    IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;
+
+    pid_t tid = -1;
     if (mFlags & AUDIO_INPUT_FLAG_FAST) {
-        if ((mTransfer != TRANSFER_CALLBACK) || (mAudioRecordThread == 0)) {
-            ALOGW("AUDIO_INPUT_FLAG_FAST denied by client");
-            // once denied, do not request again if IAudioRecord is re-created
-            mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
-        } else {
-            trackFlags |= IAudioFlinger::TRACK_FAST;
+        trackFlags |= IAudioFlinger::TRACK_FAST;
+        if (mAudioRecordThread != 0) {
             tid = mAudioRecordThread->getTid();
         }
     }
 
+    // FIXME Assume double buffering, because we don't know the true HAL sample rate
+    const uint32_t nBuffering = 2;
+
     mNotificationFramesAct = mNotificationFramesReq;
+    size_t frameCount = mReqFrameCount;
 
     if (!(mFlags & AUDIO_INPUT_FLAG_FAST)) {
         // Make sure that application is notified with sufficient margin before overrun
-        if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount/2) {
-            mNotificationFramesAct = mFrameCount/2;
+        if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) {
+            mNotificationFramesAct = frameCount/2;
         }
     }
 
     audio_io_handle_t input = AudioSystem::getInput(mInputSource, mSampleRate, mFormat,
             mChannelMask, mSessionId);
     if (input == 0) {
-        ALOGE("Could not get audio input for record source %d", mInputSource);
+        ALOGE("Could not get audio input for record source %d, sample rate %u, format %#x, "
+              "channel mask %#x, session %d",
+              mInputSource, mSampleRate, mFormat, mChannelMask, mSessionId);
         return BAD_VALUE;
     }
+    {
+    // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
+    // we must release it ourselves if anything goes wrong.
 
+    size_t temp = frameCount;   // temp may be replaced by a revised value of frameCount,
+                                // but we will still need the original value also
     int originalSessionId = mSessionId;
     sp<IAudioRecord> record = audioFlinger->openRecord(input,
                                                        mSampleRate, mFormat,
                                                        mChannelMask,
-                                                       mFrameCount,
+                                                       &temp,
                                                        &trackFlags,
                                                        tid,
                                                        &mSessionId,
                                                        &status);
-    ALOGE_IF(originalSessionId != 0 && mSessionId != originalSessionId,
+    ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId,
             "session ID changed from %d to %d", originalSessionId, mSessionId);
 
-    if (record == 0) {
+    if (status != NO_ERROR) {
         ALOGE("AudioFlinger could not create record track, status: %d", status);
-        AudioSystem::releaseInput(input);
-        return status;
+        goto release;
     }
+    ALOG_ASSERT(record != 0);
+
+    // AudioFlinger now owns the reference to the I/O handle,
+    // so we are no longer responsible for releasing it.
+
     sp<IMemory> iMem = record->getCblk();
     if (iMem == 0) {
         ALOGE("Could not get control block");
         return NO_INIT;
     }
+    void *iMemPointer = iMem->pointer();
+    if (iMemPointer == NULL) {
+        ALOGE("Could not get control block pointer");
+        return NO_INIT;
+    }
+    // invariant that mAudioRecord != 0 is true only after set() returns successfully
     if (mAudioRecord != 0) {
         mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
         mDeathNotifier.clear();
     }
-    mInput = input;
     mAudioRecord = record;
+
     mCblkMemory = iMem;
-    audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer());
+    audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
     mCblk = cblk;
-    // FIXME missing fast track frameCount logic
+    // note that temp is the (possibly revised) value of frameCount
+    if (temp < frameCount || (frameCount == 0 && temp == 0)) {
+        ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp);
+    }
+    frameCount = temp;
+
     mAwaitBoost = false;
     if (mFlags & AUDIO_INPUT_FLAG_FAST) {
         if (trackFlags & IAudioFlinger::TRACK_FAST) {
-            ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %u", mFrameCount);
+            ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %u", frameCount);
             mAwaitBoost = true;
-            // double-buffering is not required for fast tracks, due to tighter scheduling
-            if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount) {
-                mNotificationFramesAct = mFrameCount;
-            }
         } else {
-            ALOGV("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %u", mFrameCount);
+            ALOGV("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %u", frameCount);
             // once denied, do not request again if IAudioRecord is re-created
             mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
-            if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount/2) {
-                mNotificationFramesAct = mFrameCount/2;
-            }
+        }
+        // Theoretically double-buffering is not required for fast tracks,
+        // due to tighter scheduling.  But in practice, to accomodate kernels with
+        // scheduling jitter, and apps with computation jitter, we use double-buffering.
+        if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {
+            mNotificationFramesAct = frameCount/nBuffering;
         }
     }
 
-    // starting address of buffers in shared memory
+    // We retain a copy of the I/O handle, but don't own the reference
+    mInput = input;
+    mRefreshRemaining = true;
+
+    // Starting address of buffers in shared memory, immediately after the control block.  This
+    // address is for the mapping within client address space.  AudioFlinger::TrackBase::mBuffer
+    // is for the server address space.
     void *buffers = (char*)cblk + sizeof(audio_track_cblk_t);
 
+    mFrameCount = frameCount;
+    // If IAudioRecord is re-created, don't let the requested frameCount
+    // decrease.  This can confuse clients that cache frameCount().
+    if (frameCount > mReqFrameCount) {
+        mReqFrameCount = frameCount;
+    }
+
     // update proxy
     mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize);
     mProxy->setEpoch(epoch);
@@ -525,6 +573,14 @@
     mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this);
 
     return NO_ERROR;
+    }
+
+release:
+    AudioSystem::releaseInput(input);
+    if (status == NO_ERROR) {
+        status = NO_INIT;
+    }
+    return status;
 }
 
 status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
@@ -540,13 +596,13 @@
     }
 
     const struct timespec *requested;
+    struct timespec timeout;
     if (waitCount == -1) {
         requested = &ClientProxy::kForever;
     } else if (waitCount == 0) {
         requested = &ClientProxy::kNonBlocking;
     } else if (waitCount > 0) {
         long long ms = WAIT_PERIOD_MS * (long long) waitCount;
-        struct timespec timeout;
         timeout.tv_sec = ms / 1000;
         timeout.tv_nsec = (int) (ms % 1000) * 1000000;
         requested = &timeout;
@@ -586,6 +642,9 @@
                 if (newSequence == oldSequence) {
                     status = restoreRecord_l("obtainBuffer");
                     if (status != NO_ERROR) {
+                        buffer.mFrameCount = 0;
+                        buffer.mRaw = NULL;
+                        buffer.mNonContig = 0;
                         break;
                     }
                 }
@@ -687,7 +746,7 @@
 
 // -------------------------------------------------------------------------
 
-nsecs_t AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread)
+nsecs_t AudioRecord::processAudioBuffer()
 {
     mLock.lock();
     if (mAwaitBoost) {
@@ -755,17 +814,17 @@
     }
 
     // Cache other fields that will be needed soon
-    size_t notificationFrames = mNotificationFramesAct;
+    uint32_t notificationFrames = mNotificationFramesAct;
     if (mRefreshRemaining) {
         mRefreshRemaining = false;
         mRemainingFrames = notificationFrames;
         mRetryOnPartialBuffer = false;
     }
     size_t misalignment = mProxy->getMisalignment();
-    int32_t sequence = mSequence;
+    uint32_t sequence = mSequence;
 
     // These fields don't need to be cached, because they are assigned only by set():
-    //      mTransfer, mCbf, mUserData, mSampleRate
+    //      mTransfer, mCbf, mUserData, mSampleRate, mFrameSize
 
     mLock.unlock();
 
@@ -839,8 +898,8 @@
                 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount);
         requested = &ClientProxy::kNonBlocking;
         size_t avail = audioBuffer.frameCount + nonContig;
-        ALOGV("obtainBuffer(%u) returned %u = %u + %u",
-                mRemainingFrames, avail, audioBuffer.frameCount, nonContig);
+        ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d",
+                mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
         if (err != NO_ERROR) {
             if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) {
                 break;
@@ -949,7 +1008,7 @@
 
 // =========================================================================
 
-void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who)
+void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
 {
     sp<AudioRecord> audioRecord = mAudioRecord.promote();
     if (audioRecord != 0) {
@@ -961,7 +1020,8 @@
 // =========================================================================
 
 AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver, bool bCanCallJava)
-    : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL)
+    : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
+      mIgnoreNextPausedInt(false)
 {
 }
 
@@ -978,6 +1038,10 @@
             // caller will check for exitPending()
             return true;
         }
+        if (mIgnoreNextPausedInt) {
+            mIgnoreNextPausedInt = false;
+            mPausedInt = false;
+        }
         if (mPausedInt) {
             if (mPausedNs > 0) {
                 (void) mMyCond.waitRelative(mMyLock, mPausedNs);
@@ -988,7 +1052,7 @@
             return true;
         }
     }
-    nsecs_t ns =  mReceiver.processAudioBuffer(this);
+    nsecs_t ns =  mReceiver.processAudioBuffer();
     switch (ns) {
     case 0:
         return true;
@@ -1012,12 +1076,7 @@
 {
     // must be in this order to avoid a race condition
     Thread::requestExit();
-    AutoMutex _l(mMyLock);
-    if (mPaused || mPausedInt) {
-        mPaused = false;
-        mPausedInt = false;
-        mMyCond.signal();
-    }
+    resume();
 }
 
 void AudioRecord::AudioRecordThread::pause()
@@ -1029,6 +1088,7 @@
 void AudioRecord::AudioRecordThread::resume()
 {
     AutoMutex _l(mMyLock);
+    mIgnoreNextPausedInt = true;
     if (mPaused || mPausedInt) {
         mPaused = false;
         mPausedInt = false;

media/libmedia/AudioSystem.cpp

diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 8033c2c..140fb66 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -40,10 +40,10 @@
 DefaultKeyedVector<audio_io_handle_t, AudioSystem::OutputDescriptor *> AudioSystem::gOutputs(0);
 
 // Cached values for recording queries, all protected by gLock
-uint32_t AudioSystem::gPrevInSamplingRate = 16000;
-audio_format_t AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT;
-audio_channel_mask_t AudioSystem::gPrevInChannelMask = AUDIO_CHANNEL_IN_MONO;
-size_t AudioSystem::gInBuffSize = 0;
+uint32_t AudioSystem::gPrevInSamplingRate;
+audio_format_t AudioSystem::gPrevInFormat;
+audio_channel_mask_t AudioSystem::gPrevInChannelMask;
+size_t AudioSystem::gInBuffSize = 0;    // zero indicates cache is invalid
 
 
 // establish binder interface to AudioFlinger service
@@ -84,13 +84,15 @@
     return DEAD_OBJECT;
 }
 
-status_t AudioSystem::muteMicrophone(bool state) {
+status_t AudioSystem::muteMicrophone(bool state)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
     return af->setMicMute(state);
 }
 
-status_t AudioSystem::isMicrophoneMuted(bool* state) {
+status_t AudioSystem::isMicrophoneMuted(bool* state)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
     *state = af->getMicMute();
@@ -175,13 +177,15 @@
     return af->setMode(mode);
 }
 
-status_t AudioSystem::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) {
+status_t AudioSystem::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
     return af->setParameters(ioHandle, keyValuePairs);
 }
 
-String8 AudioSystem::getParameters(audio_io_handle_t ioHandle, const String8& keys) {
+String8 AudioSystem::getParameters(audio_io_handle_t ioHandle, const String8& keys)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     String8 result = String8("");
     if (af == 0) return result;
@@ -190,6 +194,16 @@
     return result;
 }
 
+status_t AudioSystem::setParameters(const String8& keyValuePairs)
+{
+    return setParameters((audio_io_handle_t) 0, keyValuePairs);
+}
+
+String8 AudioSystem::getParameters(const String8& keys)
+{
+    return getParameters((audio_io_handle_t) 0, keys);
+}
+
 // convert volume steps to natural log scale
 
 // change this value to change volume scaling
@@ -249,6 +263,11 @@
         *samplingRate = outputDesc->samplingRate;
         gLock.unlock();
     }
+    if (*samplingRate == 0) {
+        ALOGE("AudioSystem::getSamplingRate failed for output %d stream type %d",
+                output, streamType);
+        return BAD_VALUE;
+    }
 
     ALOGV("getSamplingRate() streamType %d, output %d, sampling rate %u", streamType, output,
             *samplingRate);
@@ -289,6 +308,11 @@
         *frameCount = outputDesc->frameCount;
         gLock.unlock();
     }
+    if (*frameCount == 0) {
+        ALOGE("AudioSystem::getFrameCount failed for output %d stream type %d",
+                output, streamType);
+        return BAD_VALUE;
+    }
 
     ALOGV("getFrameCount() streamType %d, output %d, frameCount %d", streamType, output,
             *frameCount);
@@ -313,7 +337,7 @@
 }
 
 status_t AudioSystem::getLatency(audio_io_handle_t output,
-                                 audio_stream_type_t streamType,
+                                 audio_stream_type_t streamType __unused,
                                  uint32_t* latency)
 {
     OutputDescriptor *outputDesc;
@@ -349,6 +373,12 @@
             return PERMISSION_DENIED;
         }
         inBuffSize = af->getInputBufferSize(sampleRate, format, channelMask);
+        if (inBuffSize == 0) {
+            ALOGE("AudioSystem::getInputBufferSize failed sampleRate %d format %#x channelMask %x",
+                    sampleRate, format, channelMask);
+            return BAD_VALUE;
+        }
+        // A benign race is possible here: we could overwrite a fresher cache entry
         gLock.lock();
         // save the request params
         gPrevInSamplingRate = sampleRate;
@@ -370,8 +400,8 @@
     return af->setVoiceVolume(value);
 }
 
-status_t AudioSystem::getRenderPosition(audio_io_handle_t output, size_t *halFrames,
-                                        size_t *dspFrames, audio_stream_type_t stream)
+status_t AudioSystem::getRenderPosition(audio_io_handle_t output, uint32_t *halFrames,
+                                        uint32_t *dspFrames, audio_stream_type_t stream)
 {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
@@ -387,9 +417,10 @@
     return af->getRenderPosition(halFrames, dspFrames, output);
 }
 
-size_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
+uint32_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
-    unsigned int result = 0;
+    uint32_t result = 0;
     if (af == 0) return result;
     if (ioHandle == 0) return result;
 
@@ -397,29 +428,33 @@
     return result;
 }
 
-int AudioSystem::newAudioSessionId() {
+int AudioSystem::newAudioSessionId()
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return 0;
     return af->newAudioSessionId();
 }
 
-void AudioSystem::acquireAudioSessionId(int audioSession) {
+void AudioSystem::acquireAudioSessionId(int audioSession, pid_t pid)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af != 0) {
-        af->acquireAudioSessionId(audioSession);
+        af->acquireAudioSessionId(audioSession, pid);
     }
 }
 
-void AudioSystem::releaseAudioSessionId(int audioSession) {
+void AudioSystem::releaseAudioSessionId(int audioSession, pid_t pid)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af != 0) {
-        af->releaseAudioSessionId(audioSession);
+        af->releaseAudioSessionId(audioSession, pid);
     }
 }
 
 // ---------------------------------------------------------------------------
 
-void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who) {
+void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who __unused)
+{
     Mutex::Autolock _l(AudioSystem::gLock);
 
     AudioSystem::gAudioFlinger.clear();
@@ -455,7 +490,7 @@
 
         OutputDescriptor *outputDesc =  new OutputDescriptor(*desc);
         gOutputs.add(ioHandle, outputDesc);
-        ALOGV("ioConfigChanged() new output samplingRate %u, format %d channel mask %#x frameCount %u "
+        ALOGV("ioConfigChanged() new output samplingRate %u, format %#x channel mask %#x frameCount %u "
                 "latency %d",
                 outputDesc->samplingRate, outputDesc->format, outputDesc->channelMask,
                 outputDesc->frameCount, outputDesc->latency);
@@ -479,7 +514,7 @@
         if (param2 == NULL) break;
         desc = (const OutputDescriptor *)param2;
 
-        ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %d channel mask %#x "
+        ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %#x channel mask %#x "
                 "frameCount %d latency %d",
                 ioHandle, desc->samplingRate, desc->format,
                 desc->channelMask, desc->frameCount, desc->latency);
@@ -496,12 +531,14 @@
     }
 }
 
-void AudioSystem::setErrorCallback(audio_error_callback cb) {
+void AudioSystem::setErrorCallback(audio_error_callback cb)
+{
     Mutex::Autolock _l(gLock);
     gAudioErrorCallback = cb;
 }
 
-bool AudioSystem::routedToA2dpOutput(audio_stream_type_t streamType) {
+bool AudioSystem::routedToA2dpOutput(audio_stream_type_t streamType)
+{
     switch (streamType) {
     case AUDIO_STREAM_MUSIC:
     case AUDIO_STREAM_VOICE_CALL:
@@ -709,7 +746,8 @@
 audio_io_handle_t AudioSystem::getOutputForEffect(const effect_descriptor_t *desc)
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
-    if (aps == 0) return PERMISSION_DENIED;
+    // FIXME change return type to status_t, and return PERMISSION_DENIED here
+    if (aps == 0) return 0;
     return aps->getOutputForEffect(desc);
 }
 
@@ -804,7 +842,8 @@
 
 // ---------------------------------------------------------------------------
 
-void AudioSystem::AudioPolicyServiceClient::binderDied(const wp<IBinder>& who) {
+void AudioSystem::AudioPolicyServiceClient::binderDied(const wp<IBinder>& who __unused)
+{
     Mutex::Autolock _l(AudioSystem::gLock);
     AudioSystem::gAudioPolicyService.clear();
 

media/libmedia/AudioTrack.cpp

diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 5e805c9..3217171 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -44,9 +44,6 @@
         return BAD_VALUE;
     }
 
-    // default to 0 in case of error
-    *frameCount = 0;
-
     // FIXME merge with similar code in createTrack_l(), except we're missing
     //       some information here that is available in createTrack_l():
     //          audio_io_handle_t output
@@ -54,16 +51,26 @@
     //          audio_channel_mask_t channelMask
     //          audio_output_flags_t flags
     uint32_t afSampleRate;
-    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
-        return NO_INIT;
+    status_t status;
+    status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType);
+    if (status != NO_ERROR) {
+        ALOGE("Unable to query output sample rate for stream type %d; status %d",
+                streamType, status);
+        return status;
     }
     size_t afFrameCount;
-    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
-        return NO_INIT;
+    status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType);
+    if (status != NO_ERROR) {
+        ALOGE("Unable to query output frame count for stream type %d; status %d",
+                streamType, status);
+        return status;
     }
     uint32_t afLatency;
-    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
-        return NO_INIT;
+    status = AudioSystem::getOutputLatency(&afLatency, streamType);
+    if (status != NO_ERROR) {
+        ALOGE("Unable to query output latency for stream type %d; status %d",
+                streamType, status);
+        return status;
     }
 
     // Ensure that buffer depth covers at least audio hardware latency
@@ -74,6 +81,13 @@
 
     *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
             afFrameCount * minBufCount * sampleRate / afSampleRate;
+    // The formula above should always produce a non-zero value, but return an error
+    // in the unlikely event that it does not, as that's part of the API contract.
+    if (*frameCount == 0) {
+        ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d",
+                streamType, sampleRate);
+        return BAD_VALUE;
+    }
     ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d",
             *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency);
     return NO_ERROR;
@@ -85,7 +99,8 @@
     : mStatus(NO_INIT),
       mIsTimed(false),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
-      mPreviousSchedulingGroup(SP_DEFAULT)
+      mPreviousSchedulingGroup(SP_DEFAULT),
+      mPausedPosition(0)
 {
 }
 
@@ -94,22 +109,26 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        int frameCount,
+        size_t frameCount,
         audio_output_flags_t flags,
         callback_t cbf,
         void* user,
-        int notificationFrames,
+        uint32_t notificationFrames,
         int sessionId,
         transfer_type transferType,
-        const audio_offload_info_t *offloadInfo)
+        const audio_offload_info_t *offloadInfo,
+        int uid,
+        pid_t pid)
     : mStatus(NO_INIT),
       mIsTimed(false),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
-      mPreviousSchedulingGroup(SP_DEFAULT)
+      mPreviousSchedulingGroup(SP_DEFAULT),
+      mPausedPosition(0)
 {
     mStatus = set(streamType, sampleRate, format, channelMask,
             frameCount, flags, cbf, user, notificationFrames,
-            0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo);
+            0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType,
+            offloadInfo, uid, pid);
 }
 
 AudioTrack::AudioTrack(
@@ -121,18 +140,22 @@
         audio_output_flags_t flags,
         callback_t cbf,
         void* user,
-        int notificationFrames,
+        uint32_t notificationFrames,
         int sessionId,
         transfer_type transferType,
-        const audio_offload_info_t *offloadInfo)
+        const audio_offload_info_t *offloadInfo,
+        int uid,
+        pid_t pid)
     : mStatus(NO_INIT),
       mIsTimed(false),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
-      mPreviousSchedulingGroup(SP_DEFAULT)
+      mPreviousSchedulingGroup(SP_DEFAULT),
+      mPausedPosition(0)
 {
     mStatus = set(streamType, sampleRate, format, channelMask,
             0 /*frameCount*/, flags, cbf, user, notificationFrames,
-            sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo);
+            sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
+            uid, pid);
 }
 
 AudioTrack::~AudioTrack()
@@ -151,7 +174,9 @@
         mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this);
         mAudioTrack.clear();
         IPCThreadState::self()->flushCommands();
-        AudioSystem::releaseAudioSessionId(mSessionId);
+        ALOGV("~AudioTrack, releasing session id from %d on behalf of %d",
+                IPCThreadState::self()->getCallingPid(), mClientPid);
+        AudioSystem::releaseAudioSessionId(mSessionId, mClientPid);
     }
 }
 
@@ -160,17 +185,24 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        int frameCountInt,
+        size_t frameCount,
         audio_output_flags_t flags,
         callback_t cbf,
         void* user,
-        int notificationFrames,
+        uint32_t notificationFrames,
         const sp<IMemory>& sharedBuffer,
         bool threadCanCallJava,
         int sessionId,
         transfer_type transferType,
-        const audio_offload_info_t *offloadInfo)
+        const audio_offload_info_t *offloadInfo,
+        int uid,
+        pid_t pid)
 {
+    ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
+          "flags #%x, notificationFrames %u, sessionId %d, transferType %d",
+          streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
+          sessionId, transferType);
+
     switch (transferType) {
     case TRANSFER_DEFAULT:
         if (sharedBuffer != 0) {
@@ -204,15 +236,9 @@
         ALOGE("Invalid transfer type %d", transferType);
         return BAD_VALUE;
     }
+    mSharedBuffer = sharedBuffer;
     mTransfer = transferType;
 
-    // FIXME "int" here is legacy and will be replaced by size_t later
-    if (frameCountInt < 0) {
-        ALOGE("Invalid frame count %d", frameCountInt);
-        return BAD_VALUE;
-    }
-    size_t frameCount = frameCountInt;
-
     ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
             sharedBuffer->size());
 
@@ -226,19 +252,24 @@
         return INVALID_OPERATION;
     }
 
-    mOutput = 0;
-
     // handle default values first.
     if (streamType == AUDIO_STREAM_DEFAULT) {
         streamType = AUDIO_STREAM_MUSIC;
     }
+    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
+        ALOGE("Invalid stream type %d", streamType);
+        return BAD_VALUE;
+    }
+    mStreamType = streamType;
 
+    status_t status;
     if (sampleRate == 0) {
-        uint32_t afSampleRate;
-        if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
-            return NO_INIT;
+        status = AudioSystem::getOutputSamplingRate(&sampleRate, streamType);
+        if (status != NO_ERROR) {
+            ALOGE("Could not get output sample rate for stream type %d; status %d",
+                    streamType, status);
+            return status;
         }
-        sampleRate = afSampleRate;
     }
     mSampleRate = sampleRate;
 
@@ -246,15 +277,21 @@
     if (format == AUDIO_FORMAT_DEFAULT) {
         format = AUDIO_FORMAT_PCM_16_BIT;
     }
-    if (channelMask == 0) {
-        channelMask = AUDIO_CHANNEL_OUT_STEREO;
-    }
 
     // validate parameters
     if (!audio_is_valid_format(format)) {
-        ALOGE("Invalid format %d", format);
+        ALOGE("Invalid format %#x", format);
         return BAD_VALUE;
     }
+    mFormat = format;
+
+    if (!audio_is_output_channel(channelMask)) {
+        ALOGE("Invalid channel mask %#x", channelMask);
+        return BAD_VALUE;
+    }
+    mChannelMask = channelMask;
+    uint32_t channelCount = popcount(channelMask);
+    mChannelCount = channelCount;
 
     // AudioFlinger does not currently support 8-bit data in shared memory
     if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) {
@@ -278,14 +315,6 @@
         flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
     }
 
-    if (!audio_is_output_channel(channelMask)) {
-        ALOGE("Invalid channel mask %#x", channelMask);
-        return BAD_VALUE;
-    }
-    mChannelMask = channelMask;
-    uint32_t channelCount = popcount(channelMask);
-    mChannelCount = channelCount;
-
     if (audio_is_linear_pcm(format)) {
         mFrameSize = channelCount * audio_bytes_per_sample(format);
         mFrameSizeAF = channelCount * sizeof(int16_t);
@@ -294,25 +323,36 @@
         mFrameSizeAF = sizeof(uint8_t);
     }
 
-    audio_io_handle_t output = AudioSystem::getOutput(
-                                    streamType,
-                                    sampleRate, format, channelMask,
-                                    flags,
-                                    offloadInfo);
-
-    if (output == 0) {
-        ALOGE("Could not get audio output for stream type %d", streamType);
-        return BAD_VALUE;
+    // Make copy of input parameter offloadInfo so that in the future:
+    //  (a) createTrack_l doesn't need it as an input parameter
+    //  (b) we can support re-creation of offloaded tracks
+    if (offloadInfo != NULL) {
+        mOffloadInfoCopy = *offloadInfo;
+        mOffloadInfo = &mOffloadInfoCopy;
+    } else {
+        mOffloadInfo = NULL;
     }
 
     mVolume[LEFT] = 1.0f;
     mVolume[RIGHT] = 1.0f;
     mSendLevel = 0.0f;
-    mFrameCount = frameCount;
+    // mFrameCount is initialized in createTrack_l
     mReqFrameCount = frameCount;
     mNotificationFramesReq = notificationFrames;
     mNotificationFramesAct = 0;
     mSessionId = sessionId;
+    int callingpid = IPCThreadState::self()->getCallingPid();
+    int mypid = getpid();
+    if (uid == -1 || (callingpid != mypid)) {
+        mClientUid = IPCThreadState::self()->getCallingUid();
+    } else {
+        mClientUid = uid;
+    }
+    if (pid == -1 || (callingpid != mypid)) {
+        mClientPid = callingpid;
+    } else {
+        mClientPid = pid;
+    }
     mAuxEffectId = 0;
     mFlags = flags;
     mCbf = cbf;
@@ -323,14 +363,7 @@
     }
 
     // create the IAudioTrack
-    status_t status = createTrack_l(streamType,
-                                  sampleRate,
-                                  format,
-                                  frameCount,
-                                  flags,
-                                  sharedBuffer,
-                                  output,
-                                  0 /*epoch*/);
+    status = createTrack_l(0 /*epoch*/);
 
     if (status != NO_ERROR) {
         if (mAudioTrackThread != 0) {
@@ -338,17 +371,20 @@
             mAudioTrackThread->requestExitAndWait();
             mAudioTrackThread.clear();
         }
+        // Use of direct and offloaded output streams is ref counted by audio policy manager.
+#if 0   // FIXME This should no longer be needed
         //Use of direct and offloaded output streams is ref counted by audio policy manager.
         // As getOutput was called above and resulted in an output stream to be opened,
         // we need to release it.
-        AudioSystem::releaseOutput(output);
+        if (mOutput != 0) {
+            AudioSystem::releaseOutput(mOutput);
+            mOutput = 0;
+        }
+#endif
         return status;
     }
 
     mStatus = NO_ERROR;
-    mStreamType = streamType;
-    mFormat = format;
-    mSharedBuffer = sharedBuffer;
     mState = STATE_STOPPED;
     mUserData = user;
     mLoopPeriod = 0;
@@ -356,11 +392,10 @@
     mMarkerReached = false;
     mNewPosition = 0;
     mUpdatePeriod = 0;
-    AudioSystem::acquireAudioSessionId(mSessionId);
+    AudioSystem::acquireAudioSessionId(mSessionId, mClientPid);
     mSequence = 1;
     mObservedSequence = mSequence;
     mInUnderrun = false;
-    mOutput = output;
 
     return NO_ERROR;
 }
@@ -436,12 +471,11 @@
 void AudioTrack::stop()
 {
     AutoMutex lock(mLock);
-    // FIXME pause then stop should not be a nop
-    if (mState != STATE_ACTIVE) {
+    if (mState != STATE_ACTIVE && mState != STATE_PAUSED) {
         return;
     }
 
-    if (isOffloaded()) {
+    if (isOffloaded_l()) {
         mState = STATE_STOPPING;
     } else {
         mState = STATE_STOPPED;
@@ -463,7 +497,7 @@
 
     sp<AudioTrackThread> t = mAudioTrackThread;
     if (t != 0) {
-        if (!isOffloaded()) {
+        if (!isOffloaded_l()) {
             t->pause();
         }
     } else {
@@ -501,7 +535,7 @@
     mRefreshRemaining = true;
 
     mState = STATE_FLUSHED;
-    if (isOffloaded()) {
+    if (isOffloaded_l()) {
         mProxy->interrupt();
     }
     mProxy->flush();
@@ -520,6 +554,16 @@
     }
     mProxy->interrupt();
     mAudioTrack->pause();
+
+    if (isOffloaded()) {
+        if (mOutput != 0) {
+            uint32_t halFrames;
+            // OffloadThread sends HAL pause in its threadLoop.. time saved
+            // here can be slightly off
+            AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition);
+            ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition);
+        }
+    }
 }
 
 status_t AudioTrack::setVolume(float left, float right)
@@ -534,6 +578,9 @@
 
     mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000));
 
+    if (isOffloaded_l()) {
+        mAudioTrack->signal();
+    }
     return NO_ERROR;
 }
 
@@ -591,6 +638,19 @@
     }
 
     AutoMutex lock(mLock);
+
+    // sample rate can be updated during playback by the offloaded decoder so we need to
+    // query the HAL and update if needed.
+// FIXME use Proxy return channel to update the rate from server and avoid polling here
+    if (isOffloaded_l()) {
+        if (mOutput != 0) {
+            uint32_t sampleRate = 0;
+            status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate);
+            if (status == NO_ERROR) {
+                mSampleRate = sampleRate;
+            }
+        }
+    }
     return mSampleRate;
 }
 
@@ -666,6 +726,7 @@
     AutoMutex lock(mLock);
     mNewPosition = mProxy->getPosition() + updatePeriod;
     mUpdatePeriod = updatePeriod;
+
     return NO_ERROR;
 }
 
@@ -719,9 +780,15 @@
     }
 
     AutoMutex lock(mLock);
-    if (isOffloaded()) {
+    if (isOffloaded_l()) {
         uint32_t dspFrames = 0;
 
+        if ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING)) {
+            ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition);
+            *position = mPausedPosition;
+            return NO_ERROR;
+        }
+
         if (mOutput != 0) {
             uint32_t halFrames;
             AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames);
@@ -735,7 +802,7 @@
     return NO_ERROR;
 }
 
-status_t AudioTrack::getBufferPosition(size_t *position)
+status_t AudioTrack::getBufferPosition(uint32_t *position)
 {
     if (mSharedBuffer == 0 || mIsTimed) {
         return INVALID_OPERATION;
@@ -768,23 +835,12 @@
     return NO_ERROR;
 }
 
-audio_io_handle_t AudioTrack::getOutput()
+audio_io_handle_t AudioTrack::getOutput() const
 {
     AutoMutex lock(mLock);
     return mOutput;
 }
 
-// must be called with mLock held
-audio_io_handle_t AudioTrack::getOutput_l()
-{
-    if (mOutput) {
-        return mOutput;
-    } else {
-        return AudioSystem::getOutput(mStreamType,
-                                      mSampleRate, mFormat, mChannelMask, mFlags);
-    }
-}
-
 status_t AudioTrack::attachAuxEffect(int effectId)
 {
     AutoMutex lock(mLock);
@@ -798,15 +854,7 @@
 // -------------------------------------------------------------------------
 
 // must be called with mLock held
-status_t AudioTrack::createTrack_l(
-        audio_stream_type_t streamType,
-        uint32_t sampleRate,
-        audio_format_t format,
-        size_t frameCount,
-        audio_output_flags_t flags,
-        const sp<IMemory>& sharedBuffer,
-        audio_io_handle_t output,
-        size_t epoch)
+status_t AudioTrack::createTrack_l(size_t epoch)
 {
     status_t status;
     const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
@@ -815,87 +863,103 @@
         return NO_INIT;
     }
 
+    audio_io_handle_t output = AudioSystem::getOutput(mStreamType, mSampleRate, mFormat,
+            mChannelMask, mFlags, mOffloadInfo);
+    if (output == 0) {
+        ALOGE("Could not get audio output for stream type %d, sample rate %u, format %#x, "
+              "channel mask %#x, flags %#x",
+              mStreamType, mSampleRate, mFormat, mChannelMask, mFlags);
+        return BAD_VALUE;
+    }
+    {
+    // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
+    // we must release it ourselves if anything goes wrong.
+
     // Not all of these values are needed under all conditions, but it is easier to get them all
 
     uint32_t afLatency;
-    status = AudioSystem::getLatency(output, streamType, &afLatency);
+    status = AudioSystem::getLatency(output, mStreamType, &afLatency);
     if (status != NO_ERROR) {
         ALOGE("getLatency(%d) failed status %d", output, status);
-        return NO_INIT;
+        goto release;
     }
 
     size_t afFrameCount;
-    status = AudioSystem::getFrameCount(output, streamType, &afFrameCount);
+    status = AudioSystem::getFrameCount(output, mStreamType, &afFrameCount);
     if (status != NO_ERROR) {
-        ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status);
-        return NO_INIT;
+        ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, mStreamType, status);
+        goto release;
     }
 
     uint32_t afSampleRate;
-    status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate);
+    status = AudioSystem::getSamplingRate(output, mStreamType, &afSampleRate);
     if (status != NO_ERROR) {
-        ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status);
-        return NO_INIT;
+        ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, mStreamType, status);
+        goto release;
     }
 
     // Client decides whether the track is TIMED (see below), but can only express a preference
     // for FAST.  Server will perform additional tests.
-    if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !(
+    if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !((
             // either of these use cases:
             // use case 1: shared buffer
-            (sharedBuffer != 0) ||
-            // use case 2: callback handler
-            (mCbf != NULL))) {
+            (mSharedBuffer != 0) ||
+            // use case 2: callback transfer mode
+            (mTransfer == TRANSFER_CALLBACK)) &&
+            // matching sample rate
+            (mSampleRate == afSampleRate))) {
         ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client");
         // once denied, do not request again if IAudioTrack is re-created
-        flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
-        mFlags = flags;
+        mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
     }
     ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);
 
     // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where
-    //  n = 1   fast track; nBuffering is ignored
+    //  n = 1   fast track with single buffering; nBuffering is ignored
+    //  n = 2   fast track with double buffering
     //  n = 2   normal track, no sample rate conversion
     //  n = 3   normal track, with sample rate conversion
     //          (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering)
     //  n > 3   very high latency or very small notification interval; nBuffering is ignored
-    const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3;
+    const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3;
 
     mNotificationFramesAct = mNotificationFramesReq;
 
-    if (!audio_is_linear_pcm(format)) {
+    size_t frameCount = mReqFrameCount;
+    if (!audio_is_linear_pcm(mFormat)) {
 
-        if (sharedBuffer != 0) {
+        if (mSharedBuffer != 0) {
             // Same comment as below about ignoring frameCount parameter for set()
-            frameCount = sharedBuffer->size();
+            frameCount = mSharedBuffer->size();
         } else if (frameCount == 0) {
             frameCount = afFrameCount;
         }
         if (mNotificationFramesAct != frameCount) {
             mNotificationFramesAct = frameCount;
         }
-    } else if (sharedBuffer != 0) {
+    } 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 = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
+        size_t alignment = /* mFormat == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
         if (mChannelCount > 1) {
             // More than 2 channels does not require stronger alignment than stereo
             alignment <<= 1;
         }
-        if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) {
+        if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) {
             ALOGE("Invalid buffer alignment: address %p, channel count %u",
-                    sharedBuffer->pointer(), mChannelCount);
-            return BAD_VALUE;
+                    mSharedBuffer->pointer(), mChannelCount);
+            status = BAD_VALUE;
+            goto release;
         }
 
         // When initializing a shared buffer AudioTrack via constructors,
         // there's no frameCount parameter.
         // But when initializing a shared buffer AudioTrack via set(),
         // there _is_ a frameCount parameter.  We silently ignore it.
-        frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t);
+        frameCount = mSharedBuffer->size()/mChannelCount/sizeof(int16_t);
 
-    } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
+    } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) {
 
         // FIXME move these calculations and associated checks to server
 
@@ -907,10 +971,10 @@
             minBufCount = nBuffering;
         }
 
-        size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
+        size_t minFrameCount = (afFrameCount*mSampleRate*minBufCount)/afSampleRate;
         ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
                 ", afLatency=%d",
-                minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
+                minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency);
 
         if (frameCount == 0) {
             frameCount = minFrameCount;
@@ -935,51 +999,65 @@
     }
 
     pid_t tid = -1;
-    if (flags & AUDIO_OUTPUT_FLAG_FAST) {
+    if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
         trackFlags |= IAudioFlinger::TRACK_FAST;
         if (mAudioTrackThread != 0) {
             tid = mAudioTrackThread->getTid();
         }
     }
 
-    if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+    if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
         trackFlags |= IAudioFlinger::TRACK_OFFLOAD;
     }
 
-    sp<IAudioTrack> track = audioFlinger->createTrack(streamType,
-                                                      sampleRate,
+    size_t temp = frameCount;   // temp may be replaced by a revised value of frameCount,
+                                // but we will still need the original value also
+    sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType,
+                                                      mSampleRate,
                                                       // AudioFlinger only sees 16-bit PCM
-                                                      format == AUDIO_FORMAT_PCM_8_BIT ?
-                                                              AUDIO_FORMAT_PCM_16_BIT : format,
+                                                      mFormat == AUDIO_FORMAT_PCM_8_BIT ?
+                                                              AUDIO_FORMAT_PCM_16_BIT : mFormat,
                                                       mChannelMask,
-                                                      frameCount,
+                                                      &temp,
                                                       &trackFlags,
-                                                      sharedBuffer,
+                                                      mSharedBuffer,
                                                       output,
                                                       tid,
                                                       &mSessionId,
                                                       mName,
+                                                      mClientUid,
                                                       &status);
 
-    if (track == 0) {
+    if (status != NO_ERROR) {
         ALOGE("AudioFlinger could not create track, status: %d", status);
-        return status;
+        goto release;
     }
+    ALOG_ASSERT(track != 0);
+
+    // AudioFlinger now owns the reference to the I/O handle,
+    // so we are no longer responsible for releasing it.
+
     sp<IMemory> iMem = track->getCblk();
     if (iMem == 0) {
         ALOGE("Could not get control block");
         return NO_INIT;
     }
+    void *iMemPointer = iMem->pointer();
+    if (iMemPointer == NULL) {
+        ALOGE("Could not get control block pointer");
+        return NO_INIT;
+    }
     // invariant that mAudioTrack != 0 is true only after set() returns successfully
     if (mAudioTrack != 0) {
         mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this);
         mDeathNotifier.clear();
     }
     mAudioTrack = track;
+
     mCblkMemory = iMem;
-    audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer());
+    audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
     mCblk = cblk;
-    size_t temp = cblk->frameCount_;
+    // note that temp is the (possibly revised) value of frameCount
     if (temp < frameCount || (frameCount == 0 && temp == 0)) {
         // In current design, AudioTrack client checks and ensures frame count validity before
         // passing it to AudioFlinger so AudioFlinger should not return a different value except
@@ -987,40 +1065,44 @@
         ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp);
     }
     frameCount = temp;
+
     mAwaitBoost = false;
-    if (flags & AUDIO_OUTPUT_FLAG_FAST) {
+    if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
         if (trackFlags & IAudioFlinger::TRACK_FAST) {
             ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount);
             mAwaitBoost = true;
-            if (sharedBuffer == 0) {
-                // double-buffering is not required for fast tracks, due to tighter scheduling
-                if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount) {
-                    mNotificationFramesAct = frameCount;
+            if (mSharedBuffer == 0) {
+                // Theoretically double-buffering is not required for fast tracks,
+                // due to tighter scheduling.  But in practice, to accommodate kernels with
+                // scheduling jitter, and apps with computation jitter, we use double-buffering.
+                if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {
+                    mNotificationFramesAct = frameCount/nBuffering;
                 }
             }
         } else {
             ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount);
             // once denied, do not request again if IAudioTrack is re-created
-            flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
-            mFlags = flags;
-            if (sharedBuffer == 0) {
+            mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
+            if (mSharedBuffer == 0) {
                 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {
                     mNotificationFramesAct = frameCount/nBuffering;
                 }
             }
         }
     }
-    if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+    if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
         if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) {
             ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful");
         } else {
             ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server");
-            flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
-            mFlags = flags;
-            return NO_INIT;
+            mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+            // FIXME This is a warning, not an error, so don't return error status
+            //return NO_INIT;
         }
     }
 
+    // We retain a copy of the I/O handle, but don't own the reference
+    mOutput = output;
     mRefreshRemaining = true;
 
     // Starting address of buffers in shared memory.  If there is a shared buffer, buffers
@@ -1028,15 +1110,16 @@
     // immediately after the control block.  This address is for the mapping within client
     // address space.  AudioFlinger::TrackBase::mBuffer is for the server address space.
     void* buffers;
-    if (sharedBuffer == 0) {
+    if (mSharedBuffer == 0) {
         buffers = (char*)cblk + sizeof(audio_track_cblk_t);
     } else {
-        buffers = sharedBuffer->pointer();
+        buffers = mSharedBuffer->pointer();
     }
 
     mAudioTrack->attachAuxEffect(mAuxEffectId);
     // FIXME don't believe this lie
-    mLatency = afLatency + (1000*frameCount) / sampleRate;
+    mLatency = afLatency + (1000*frameCount) / mSampleRate;
+
     mFrameCount = frameCount;
     // If IAudioTrack is re-created, don't let the requested frameCount
     // decrease.  This can confuse clients that cache frameCount().
@@ -1045,7 +1128,7 @@
     }
 
     // update proxy
-    if (sharedBuffer == 0) {
+    if (mSharedBuffer == 0) {
         mStaticProxy.clear();
         mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
     } else {
@@ -1063,6 +1146,14 @@
     mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this);
 
     return NO_ERROR;
+    }
+
+release:
+    AudioSystem::releaseOutput(output);
+    if (status == NO_ERROR) {
+        status = NO_INIT;
+    }
+    return status;
 }
 
 status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
@@ -1078,13 +1169,13 @@
     }
 
     const struct timespec *requested;
+    struct timespec timeout;
     if (waitCount == -1) {
         requested = &ClientProxy::kForever;
     } else if (waitCount == 0) {
         requested = &ClientProxy::kNonBlocking;
     } else if (waitCount > 0) {
         long long ms = WAIT_PERIOD_MS * (long long) waitCount;
-        struct timespec timeout;
         timeout.tv_sec = ms / 1000;
         timeout.tv_nsec = (int) (ms % 1000) * 1000000;
         requested = &timeout;
@@ -1209,7 +1300,7 @@
     if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
         // Sanity-check: user is most-likely passing an error code, and it would
         // make the return value ambiguous (actualSize vs error).
-        ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize);
+        ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize);
         return BAD_VALUE;
     }
 
@@ -1315,7 +1406,7 @@
 
 // -------------------------------------------------------------------------
 
-nsecs_t AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
+nsecs_t AudioTrack::processAudioBuffer()
 {
     // Currently the AudioTrack thread is not created if there are no callbacks.
     // Would it ever make sense to run the thread, even without callbacks?
@@ -1353,7 +1444,7 @@
         // for offloaded tracks restoreTrack_l() will just update the sequence and clear
         // AudioSystem cache. We should not exit here but after calling the callback so
         // that the upper layers can recreate the track
-        if (!isOffloaded() || (mSequence == mObservedSequence)) {
+        if (!isOffloaded_l() || (mSequence == mObservedSequence)) {
             status_t status = restoreTrack_l("processAudioBuffer");
             mLock.unlock();
             // Run again immediately, but with a new IAudioTrack
@@ -1408,7 +1499,7 @@
     // Cache other fields that will be needed soon
     uint32_t loopPeriod = mLoopPeriod;
     uint32_t sampleRate = mSampleRate;
-    size_t notificationFrames = mNotificationFramesAct;
+    uint32_t notificationFrames = mNotificationFramesAct;
     if (mRefreshRemaining) {
         mRefreshRemaining = false;
         mRemainingFrames = notificationFrames;
@@ -1572,7 +1663,6 @@
         size_t reqSize = audioBuffer.size;
         mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
         size_t writtenSize = audioBuffer.size;
-        size_t writtenFrames = writtenSize / mFrameSize;
 
         // Sanity check on returned size
         if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
@@ -1638,36 +1728,27 @@
 status_t AudioTrack::restoreTrack_l(const char *from)
 {
     ALOGW("dead IAudioTrack, %s, creating a new one from %s()",
-          isOffloaded() ? "Offloaded" : "PCM", from);
+          isOffloaded_l() ? "Offloaded" : "PCM", from);
     ++mSequence;
     status_t result;
 
     // refresh the audio configuration cache in this process to make sure we get new
-    // output parameters in getOutput_l() and createTrack_l()
+    // output parameters in createTrack_l()
     AudioSystem::clearAudioConfigCache();
 
-    if (isOffloaded()) {
+    if (isOffloaded_l()) {
+        // FIXME re-creation of offloaded tracks is not yet implemented
         return DEAD_OBJECT;
     }
 
-    // force new output query from audio policy manager;
-    mOutput = 0;
-    audio_io_handle_t output = getOutput_l();
-
     // if the new IAudioTrack is created, createTrack_l() will modify the
     // following member variables: mAudioTrack, mCblkMemory and mCblk.
     // It will also delete the strong references on previous IAudioTrack and IMemory
-    size_t position = mProxy->getPosition();
-    mNewPosition = position + mUpdatePeriod;
+
+    // take the frames that will be lost by track recreation into account in saved position
+    size_t position = mProxy->getPosition() + mProxy->getFramesFilled();
     size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0;
-    result = createTrack_l(mStreamType,
-                           mSampleRate,
-                           mFormat,
-                           mReqFrameCount,  // so that frame count never goes down
-                           mFlags,
-                           mSharedBuffer,
-                           output,
-                           position /*epoch*/);
+    result = createTrack_l(position /*epoch*/);
 
     if (result == NO_ERROR) {
         // continue playback from last known position, but
@@ -1695,10 +1776,16 @@
         }
     }
     if (result != NO_ERROR) {
+        // Use of direct and offloaded output streams is ref counted by audio policy manager.
+#if 0   // FIXME This should no longer be needed
         //Use of direct and offloaded output streams is ref counted by audio policy manager.
         // As getOutput was called above and resulted in an output stream to be opened,
         // we need to release it.
-        AudioSystem::releaseOutput(output);
+        if (mOutput != 0) {
+            AudioSystem::releaseOutput(mOutput);
+            mOutput = 0;
+        }
+#endif
         ALOGW("restoreTrack_l() failed status %d", result);
         mState = STATE_STOPPED;
     }
@@ -1731,14 +1818,21 @@
 
 String8 AudioTrack::getParameters(const String8& keys)
 {
-    if (mOutput) {
-        return AudioSystem::getParameters(mOutput, keys);
+    audio_io_handle_t output = getOutput();
+    if (output != 0) {
+        return AudioSystem::getParameters(output, keys);
     } else {
         return String8::empty();
     }
 }
 
-status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
+bool AudioTrack::isOffloaded() const
+{
+    AutoMutex lock(mLock);
+    return isOffloaded_l();
+}
+
+status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
 {
 
     const size_t SIZE = 256;
@@ -1749,7 +1843,7 @@
     snprintf(buffer, 255, "  stream type(%d), left - right volume(%f, %f)\n", mStreamType,
             mVolume[0], mVolume[1]);
     result.append(buffer);
-    snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat,
+    snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%zu)\n", mFormat,
             mChannelCount, mFrameCount);
     result.append(buffer);
     snprintf(buffer, 255, "  sample rate(%u), status(%d)\n", mSampleRate, mStatus);
@@ -1768,7 +1862,7 @@
 
 // =========================================================================
 
-void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who)
+void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
 {
     sp<AudioTrack> audioTrack = mAudioTrack.promote();
     if (audioTrack != 0) {
@@ -1780,7 +1874,8 @@
 // =========================================================================
 
 AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
-    : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL)
+    : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
+      mIgnoreNextPausedInt(false)
 {
 }
 
@@ -1797,6 +1892,10 @@
             // caller will check for exitPending()
             return true;
         }
+        if (mIgnoreNextPausedInt) {
+            mIgnoreNextPausedInt = false;
+            mPausedInt = false;
+        }
         if (mPausedInt) {
             if (mPausedNs > 0) {
                 (void) mMyCond.waitRelative(mMyLock, mPausedNs);
@@ -1807,7 +1906,7 @@
             return true;
         }
     }
-    nsecs_t ns = mReceiver.processAudioBuffer(this);
+    nsecs_t ns = mReceiver.processAudioBuffer();
     switch (ns) {
     case 0:
         return true;
@@ -1831,12 +1930,7 @@
 {
     // must be in this order to avoid a race condition
     Thread::requestExit();
-    AutoMutex _l(mMyLock);
-    if (mPaused || mPausedInt) {
-        mPaused = false;
-        mPausedInt = false;
-        mMyCond.signal();
-    }
+    resume();
 }
 
 void AudioTrack::AudioTrackThread::pause()
@@ -1848,6 +1942,7 @@
 void AudioTrack::AudioTrackThread::resume()
 {
     AutoMutex _l(mMyLock);
+    mIgnoreNextPausedInt = true;
     if (mPaused || mPausedInt) {
         mPaused = false;
         mPausedInt = false;

media/libmedia/AudioTrackShared.cpp

diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index 4fd92b2..fdd1a12 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -26,7 +26,7 @@
 namespace android {
 
 audio_track_cblk_t::audio_track_cblk_t()
-    : mServer(0), frameCount_(0), mFutex(0), mMinimum(0),
+    : mServer(0), mFutex(0), mMinimum(0),
     mVolumeLR(0x10001000), mSampleRate(0), mSendLevel(0), mFlags(0)
 {
     memset(&u, 0, sizeof(u));
@@ -316,6 +316,27 @@
             (mFrameCountP2 - 1);
 }
 
+size_t ClientProxy::getFramesFilled() {
+    audio_track_cblk_t* cblk = mCblk;
+    int32_t front;
+    int32_t rear;
+
+    if (mIsOut) {
+        front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront);
+        rear = cblk->u.mStreaming.mRear;
+    } else {
+        rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
+        front = cblk->u.mStreaming.mFront;
+    }
+    ssize_t filled = rear - front;
+    // pipe should not be overfull
+    if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
+        ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled);
+        return 0;
+    }
+    return (size_t)filled;
+}
+
 // ---------------------------------------------------------------------------
 
 void AudioTrackClientProxy::flush()
@@ -454,9 +475,14 @@
 
 void StaticAudioTrackClientProxy::setLoop(size_t loopStart, size_t loopEnd, int loopCount)
 {
+    // This can only happen on a 64-bit client
+    if (loopStart > UINT32_MAX || loopEnd > UINT32_MAX) {
+        // FIXME Should return an error status
+        return;
+    }
     StaticAudioTrackState newState;
-    newState.mLoopStart = loopStart;
-    newState.mLoopEnd = loopEnd;
+    newState.mLoopStart = (uint32_t) loopStart;
+    newState.mLoopEnd = (uint32_t) loopEnd;
     newState.mLoopCount = loopCount;
     mBufferPosition = loopStart;
     (void) mMutator.push(newState);
@@ -466,7 +492,7 @@
 {
     size_t bufferPosition;
     if (mMutator.ack()) {
-        bufferPosition = mCblk->u.mStatic.mBufferPosition;
+        bufferPosition = (size_t) mCblk->u.mStatic.mBufferPosition;
         if (bufferPosition > mFrameCount) {
             bufferPosition = mFrameCount;
         }
@@ -485,7 +511,7 @@
 {
 }
 
-status_t ServerProxy::obtainBuffer(Buffer* buffer)
+status_t ServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush)
 {
     LOG_ALWAYS_FATAL_IF(buffer == NULL || buffer->mFrameCount == 0);
     if (mIsShutdown) {
@@ -558,7 +584,11 @@
     buffer->mRaw = part1 > 0 ?
             &((char *) mBuffers)[(mIsOut ? front : rear) * mFrameSize] : NULL;
     buffer->mNonContig = availToServer - part1;
-    mUnreleased = part1;
+    // After flush(), allow releaseBuffer() on a previously obtained buffer;
+    // see "Acknowledge any pending flush()" in audioflinger/Tracks.cpp.
+    if (!ackFlush) {
+        mUnreleased = part1;
+    }
     return part1 > 0 ? NO_ERROR : WOULD_BLOCK;
     }
 no_init:
@@ -597,7 +627,7 @@
     if (half == 0) {
         half = 1;
     }
-    size_t minimum = cblk->mMinimum;
+    size_t minimum = (size_t) cblk->mMinimum;
     if (minimum == 0) {
         minimum = mIsOut ? half : 1;
     } else if (minimum > half) {
@@ -735,12 +765,13 @@
             mIsShutdown = true;
             return (ssize_t) NO_INIT;
         }
-        mCblk->u.mStatic.mBufferPosition = position;
+        // This may overflow, but client is not supposed to rely on it
+        mCblk->u.mStatic.mBufferPosition = (uint32_t) position;
     }
     return (ssize_t) position;
 }
 
-status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer)
+status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush __unused)
 {
     if (mIsShutdown) {
         buffer->mFrameCount = 0;
@@ -811,7 +842,8 @@
     mPosition = newPosition;
 
     cblk->mServer += stepCount;
-    cblk->u.mStatic.mBufferPosition = newPosition;
+    // This may overflow, but client is not supposed to rely on it
+    cblk->u.mStatic.mBufferPosition = (uint32_t) newPosition;
     if (setFlags != 0) {
         (void) android_atomic_or(setFlags, &cblk->mFlags);
         // this would be a good place to wake a futex
@@ -822,7 +854,7 @@
     buffer->mNonContig = 0;
 }
 
-void StaticAudioTrackServerProxy::tallyUnderrunFrames(uint32_t frameCount)
+void StaticAudioTrackServerProxy::tallyUnderrunFrames(uint32_t frameCount __unused)
 {
     // Unlike AudioTrackServerProxy::tallyUnderrunFrames() used for streaming tracks,
     // we don't have a location to count underrun frames.  The underrun frame counter

media/libmedia/CharacterEncodingDetector.cpp

diff --git a/media/libmedia/CharacterEncodingDetector.cpp b/media/libmedia/CharacterEncodingDetector.cpp
new file mode 100644
index 0000000..eb091ac
--- /dev/null
+++ b/media/libmedia/CharacterEncodingDetector.cpp
@@ -0,0 +1,364 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "CharacterEncodingDector"
+#include <utils/Log.h>
+
+#include "CharacterEncodingDetector.h"
+#include "CharacterEncodingDetectorTables.h"
+
+#include "utils/Vector.h"
+#include "StringArray.h"
+
+#include "unicode/ucnv.h"
+#include "unicode/ucsdet.h"
+#include "unicode/ustring.h"
+
+namespace android {
+
+CharacterEncodingDetector::CharacterEncodingDetector() {
+
+    UErrorCode status = U_ZERO_ERROR;
+    mUtf8Conv = ucnv_open("UTF-8", &status);
+    if (U_FAILURE(status)) {
+        ALOGE("could not create UConverter for UTF-8");
+        mUtf8Conv = NULL;
+    }
+}
+
+CharacterEncodingDetector::~CharacterEncodingDetector() {
+    ucnv_close(mUtf8Conv);
+}
+
+void CharacterEncodingDetector::addTag(const char *name, const char *value) {
+    mNames.push_back(name);
+    mValues.push_back(value);
+}
+
+size_t CharacterEncodingDetector::size() {
+    return mNames.size();
+}
+
+status_t CharacterEncodingDetector::getTag(int index, const char **name, const char**value) {
+    if (index >= mNames.size()) {
+        return BAD_VALUE;
+    }
+
+    *name = mNames.getEntry(index);
+    *value = mValues.getEntry(index);
+    return OK;
+}
+
+static bool isPrintableAscii(const char *value, size_t len) {
+    for (size_t i = 0; i < len; i++) {
+        if ((value[i] & 0x80) || value[i] < 0x20 || value[i] == 0x7f) {
+            return false;
+        }
+    }
+    return true;
+}
+
+void CharacterEncodingDetector::detectAndConvert() {
+
+    int size = mNames.size();
+    ALOGV("%d tags before conversion", size);
+    for (int i = 0; i < size; i++) {
+        ALOGV("%s: %s", mNames.getEntry(i), mValues.getEntry(i));
+    }
+
+    if (size && mUtf8Conv) {
+
+        UErrorCode status = U_ZERO_ERROR;
+        UCharsetDetector *csd = ucsdet_open(&status);
+        const UCharsetMatch *ucm;
+
+        // try combined detection of artist/album/title etc.
+        char buf[1024];
+        buf[0] = 0;
+        int idx;
+        for (int i = 0; i < size; i++) {
+            const char *name = mNames.getEntry(i);
+            const char *value = mValues.getEntry(i);
+            if (!isPrintableAscii(value, strlen(value)) && (
+                        !strcmp(name, "artist") ||
+                        !strcmp(name, "albumartist") ||
+                        !strcmp(name, "composer") ||
+                        !strcmp(name, "genre") ||
+                        !strcmp(name, "album") ||
+                        !strcmp(name, "title"))) {
+                strlcat(buf, value, sizeof(buf));
+                // separate tags by space so ICU's ngram detector can do its job
+                strlcat(buf, " ", sizeof(buf));
+            }
+        }
+        ucsdet_setText(csd, buf, strlen(buf), &status);
+
+        int32_t matches;
+        const UCharsetMatch** ucma = ucsdet_detectAll(csd, &matches, &status);
+        const char *combinedenc = "???";
+
+        const UCharsetMatch* bestCombinedMatch = getPreferred(buf, strlen(buf), ucma, matches);
+
+        if (bestCombinedMatch != NULL) {
+            combinedenc = ucsdet_getName(bestCombinedMatch, &status);
+        }
+
+        for (int i = 0; i < size; i++) {
+            const char *name = mNames.getEntry(i);
+            uint8_t* src = (uint8_t *)mValues.getEntry(i);
+            int len = strlen((char *)src);
+            uint8_t* dest = src;
+
+            ALOGV("@@@ checking %s", name);
+            const char *s = mValues.getEntry(i);
+            int32_t inputLength = strlen(s);
+            const char *enc;
+
+            if (!strcmp(name, "artist") ||
+                    !strcmp(name, "albumartist") ||
+                    !strcmp(name, "composer") ||
+                    !strcmp(name, "genre") ||
+                    !strcmp(name, "album") ||
+                    !strcmp(name, "title")) {
+                // use encoding determined from the combination of artist/album/title etc.
+                enc = combinedenc;
+            } else {
+                ucsdet_setText(csd, s, inputLength, &status);
+                ucm = ucsdet_detect(csd, &status);
+                if (!ucm) {
+                    mValues.setEntry(i, "???");
+                    continue;
+                }
+                enc = ucsdet_getName(ucm, &status);
+                ALOGV("@@@@ recognized charset: %s for %s confidence %d",
+                        enc, mNames.getEntry(i), ucsdet_getConfidence(ucm, &status));
+            }
+
+            if (strcmp(enc,"UTF-8") != 0) {
+                // only convert if the source encoding isn't already UTF-8
+                ALOGV("@@@ using converter %s for %s", enc, mNames.getEntry(i));
+                UConverter *conv = ucnv_open(enc, &status);
+                if (U_FAILURE(status)) {
+                    ALOGE("could not create UConverter for %s", enc);
+                    continue;
+                }
+
+                // convert from native encoding to UTF-8
+                const char* source = mValues.getEntry(i);
+                int targetLength = len * 3 + 1;
+                char* buffer = new char[targetLength];
+                // don't normally check for NULL, but in this case targetLength may be large
+                if (!buffer)
+                    break;
+                char* target = buffer;
+
+                ucnv_convertEx(mUtf8Conv, conv, &target, target + targetLength,
+                        &source, source + strlen(source),
+                        NULL, NULL, NULL, NULL, TRUE, TRUE, &status);
+
+                if (U_FAILURE(status)) {
+                    ALOGE("ucnv_convertEx failed: %d", status);
+                    mValues.setEntry(i, "???");
+                } else {
+                    // zero terminate
+                    *target = 0;
+                    mValues.setEntry(i, buffer);
+                }
+
+                delete[] buffer;
+
+                ucnv_close(conv);
+            }
+        }
+
+        for (int i = size - 1; i >= 0; --i) {
+            if (strlen(mValues.getEntry(i)) == 0) {
+                ALOGV("erasing %s because entry is empty", mNames.getEntry(i));
+                mNames.erase(i);
+                mValues.erase(i);
+            }
+        }
+
+        ucsdet_close(csd);
+    }
+}
+
+/*
+ * When ICU detects multiple encoding matches, apply additional heuristics to determine
+ * which one is the best match, since ICU can't always be trusted to make the right choice.
+ *
+ * What this method does is:
+ * - decode the input using each of the matches found
+ * - recalculate the starting confidence level for multibyte encodings using a different
+ *   algorithm and larger frequent character lists than ICU
+ * - devalue encoding where the conversion contains unlikely characters (symbols, reserved, etc)
+ * - pick the highest match
+ */
+const UCharsetMatch *CharacterEncodingDetector::getPreferred(
+        const char *input, size_t len, const UCharsetMatch** ucma, size_t nummatches) {
+
+    Vector<const UCharsetMatch*> matches;
+    UErrorCode status = U_ZERO_ERROR;
+
+    ALOGV("%d matches", nummatches);
+    for (size_t i = 0; i < nummatches; i++) {
+        const char *encname = ucsdet_getName(ucma[i], &status);
+        int confidence = ucsdet_getConfidence(ucma[i], &status);
+        ALOGV("%d: %s %d", i, encname, confidence);
+        matches.push_back(ucma[i]);
+    }
+
+    size_t num = matches.size();
+    if (num == 0) {
+        return NULL;
+    }
+    if (num == 1) {
+        return matches[0];
+    }
+
+    ALOGV("considering %d matches", num);
+
+    // keep track of how many "special" characters result when converting the input using each
+    // encoding
+    Vector<int> newconfidence;
+    for (size_t i = 0; i < num; i++) {
+        const uint16_t *freqdata = NULL;
+        float freqcoverage = 0;
+        status = U_ZERO_ERROR;
+        const char *encname = ucsdet_getName(matches[i], &status);
+        int confidence = ucsdet_getConfidence(matches[i], &status);
+        if (!strcmp("GB18030", encname)) {
+            freqdata = frequent_zhCN;
+            freqcoverage = frequent_zhCN_coverage;
+        } else if (!strcmp("Big5", encname)) {
+            freqdata = frequent_zhTW;
+            freqcoverage = frequent_zhTW_coverage;
+        } else if (!strcmp("EUC-KR", encname)) {
+            freqdata = frequent_ko;
+            freqcoverage = frequent_ko_coverage;
+        } else if (!strcmp("EUC-JP", encname)) {
+            freqdata = frequent_ja;
+            freqcoverage = frequent_ja_coverage;
+        } else if (!strcmp("Shift_JIS", encname)) {
+            freqdata = frequent_ja;
+            freqcoverage = frequent_ja_coverage;
+        }
+
+        ALOGV("%d: %s %d", i, encname, confidence);
+        UConverter *conv = ucnv_open(encname, &status);
+        const char *source = input;
+        const char *sourceLimit = input + len;
+        status = U_ZERO_ERROR;
+        int demerit = 0;
+        int frequentchars = 0;
+        int totalchars = 0;
+        while (true) {
+            // demerit the current encoding for each "special" character found after conversion.
+            // The amount of demerit is somewhat arbitrarily chosen.
+            int inchar;
+            if (source != sourceLimit) {
+                inchar = (source[0] << 8) + source[1];
+            }
+            UChar32 c = ucnv_getNextUChar(conv, &source, sourceLimit, &status);
+            if (!U_SUCCESS(status)) {
+                break;
+            }
+            if (c < 0x20 || (c >= 0x7f && c <= 0x009f)) {
+                ALOGV("control character %x", c);
+                demerit += 100;
+            } else if ((c >= 0xa0 && c <= 0xbe)         // symbols, superscripts
+                    || (c == 0xd7) || (c == 0xf7)       // multiplication and division signs
+                    || (c >= 0x2000 && c <= 0x209f)) {  // punctuation, superscripts
+                ALOGV("unlikely character %x", c);
+                demerit += 10;
+            } else if (c >= 0xe000 && c <= 0xf8ff) {
+                ALOGV("private use character %x", c);
+                demerit += 30;
+            } else if (c >= 0x2190 && c <= 0x2bff) {
+                // this range comprises various symbol ranges that are unlikely to appear in
+                // music file metadata.
+                ALOGV("symbol %x", c);
+                demerit += 10;
+            } else if (c == 0xfffd) {
+                ALOGV("replacement character");
+                demerit += 50;
+            } else if (c >= 0xfff0 && c <= 0xfffc) {
+                ALOGV("unicode special %x", c);
+                demerit += 50;
+            } else if (freqdata != NULL) {
+                totalchars++;
+                if (isFrequent(freqdata, c)) {
+                    frequentchars++;
+                }
+            }
+        }
+        if (freqdata != NULL && totalchars != 0) {
+            int myconfidence = 10 + float((100 * frequentchars) / totalchars) / freqcoverage;
+            ALOGV("ICU confidence: %d, my confidence: %d (%d %d)", confidence, myconfidence,
+                    totalchars, frequentchars);
+            if (myconfidence > 100) myconfidence = 100;
+            if (myconfidence < 0) myconfidence = 0;
+            confidence = myconfidence;
+        }
+        ALOGV("%d-%d=%d", confidence, demerit, confidence - demerit);
+        newconfidence.push_back(confidence - demerit);
+        ucnv_close(conv);
+        if (i == 0 && (confidence - demerit) == 100) {
+            // no need to check any further, we'll end up using this match anyway
+            break;
+        }
+    }
+
+    // find match with highest confidence after adjusting for unlikely characters
+    int highest = newconfidence[0];
+    size_t highestidx = 0;
+    num = newconfidence.size();
+    for (size_t i = 1; i < num; i++) {
+        if (newconfidence[i] > highest) {
+            highest = newconfidence[i];
+            highestidx = i;
+        }
+    }
+    status = U_ZERO_ERROR;
+    ALOGV("selecting '%s' w/ %d confidence", ucsdet_getName(matches[highestidx], &status), highest);
+    return matches[highestidx];
+}
+
+
+bool CharacterEncodingDetector::isFrequent(const uint16_t *values, uint32_t c) {
+
+    int start = 0;
+    int end = 511; // All the tables have 512 entries
+    int mid = (start+end)/2;
+
+    while(start <= end) {
+        if(c == values[mid]) {
+            return true;
+        } else if (c > values[mid]) {
+            start = mid + 1;
+        } else {
+            end = mid - 1;
+        }
+
+        mid = (start + end) / 2;
+    }
+
+    return false;
+}
+
+
+}  // namespace android

media/libmedia/CharacterEncodingDetector.h

diff --git a/media/libmedia/CharacterEncodingDetector.h b/media/libmedia/CharacterEncodingDetector.h
new file mode 100644
index 0000000..3655a91
--- /dev/null
+++ b/media/libmedia/CharacterEncodingDetector.h
@@ -0,0 +1,61 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef _CHARACTER_ENCODING_DETECTOR_H
+#define _CHARACTER_ENCODING_DETECTOR_H
+
+#include <media/mediascanner.h>
+
+#include "StringArray.h"
+
+#include "unicode/ucnv.h"
+#include "unicode/ucsdet.h"
+#include "unicode/ustring.h"
+
+namespace android {
+
+class CharacterEncodingDetector {
+
+    public:
+    CharacterEncodingDetector();
+        ~CharacterEncodingDetector();
+
+        void addTag(const char *name, const char *value);
+        size_t size();
+
+        void detectAndConvert();
+        status_t getTag(int index, const char **name, const char**value);
+
+    private:
+        const UCharsetMatch *getPreferred(
+                const char *input, size_t len, const UCharsetMatch** ucma, size_t matches);
+
+        bool isFrequent(const uint16_t *values, uint32_t c);
+
+        // cached name and value strings, for native encoding support.
+        // TODO: replace these with byte blob arrays that don't require the data to be
+        // singlenullbyte-terminated
+        StringArray     mNames;
+        StringArray     mValues;
+
+        UConverter*     mUtf8Conv;
+};
+
+
+
+};  // namespace android
+
+#endif

media/libmedia/CharacterEncodingDetectorTables.h

diff --git a/media/libmedia/CharacterEncodingDetectorTables.h b/media/libmedia/CharacterEncodingDetectorTables.h
new file mode 100644
index 0000000..1fe1137
--- /dev/null
+++ b/media/libmedia/CharacterEncodingDetectorTables.h
@@ -0,0 +1,2092 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// The 512 most frequently occuring characters for the zhCN language in a sample of the Internet.
+// Ordered by codepoint, comment shows character and ranking by frequency
+const uint16_t frequent_zhCN[] = {
+    0x4E00, // 一, #2
+    0x4E07, // 万, #306
+    0x4E09, // 三, #138
+    0x4E0A, // 上, #16
+    0x4E0B, // 下, #25
+    0x4E0D, // 不, #7
+    0x4E0E, // 与, #133
+    0x4E13, // 专, #151
+    0x4E16, // 世, #346
+    0x4E1A, // 业, #39
+    0x4E1C, // 东, #197
+    0x4E24, // 两, #376
+    0x4E2A, // 个, #23
+    0x4E2D, // 中, #4
+    0x4E3A, // 为, #31
+    0x4E3B, // 主, #95
+    0x4E3E, // 举, #418
+    0x4E48, // 么, #93
+    0x4E4B, // 之, #131
+    0x4E50, // 乐, #130
+    0x4E5F, // 也, #145
+    0x4E66, // 书, #283
+    0x4E70, // 买, #483
+    0x4E86, // 了, #13
+    0x4E8B, // 事, #168
+    0x4E8C, // 二, #218
+    0x4E8E, // 于, #64
+    0x4E94, // 五, #430
+    0x4E9A, // 亚, #468
+    0x4E9B, // 些, #366
+    0x4EA4, // 交, #243
+    0x4EA7, // 产, #86
+    0x4EAB, // 享, #345
+    0x4EAC, // 京, #206
+    0x4EBA, // 人, #3
+    0x4EC0, // 什, #287
+    0x4ECB, // 介, #478
+    0x4ECE, // 从, #381
+    0x4ED6, // 他, #129
+    0x4EE3, // 代, #241
+    0x4EE5, // 以, #51
+    0x4EEC, // 们, #83
+    0x4EF6, // 件, #141
+    0x4EF7, // 价, #140
+    0x4EFB, // 任, #383
+    0x4F01, // 企, #439
+    0x4F18, // 优, #374
+    0x4F1A, // 会, #29
+    0x4F20, // 传, #222
+    0x4F46, // 但, #451
+    0x4F4D, // 位, #208
+    0x4F53, // 体, #98
+    0x4F55, // 何, #339
+    0x4F5C, // 作, #44
+    0x4F60, // 你, #76
+    0x4F7F, // 使, #272
+    0x4F9B, // 供, #375
+    0x4FDD, // 保, #180
+    0x4FE1, // 信, #84
+    0x4FEE, // 修, #437
+    0x503C, // 值, #450
+    0x505A, // 做, #368
+    0x5065, // 健, #484
+    0x50CF, // 像, #487
+    0x513F, // 儿, #326
+    0x5143, // 元, #202
+    0x5148, // 先, #485
+    0x5149, // 光, #254
+    0x514B, // 克, #503
+    0x514D, // 免, #349
+    0x5165, // 入, #156
+    0x5168, // 全, #47
+    0x516C, // 公, #35
+    0x5171, // 共, #448
+    0x5173, // 关, #49
+    0x5176, // 其, #195
+    0x5177, // 具, #329
+    0x5185, // 内, #109
+    0x518C, // 册, #225
+    0x519B, // 军, #466
+    0x51FA, // 出, #53
+    0x51FB, // 击, #359
+    0x5206, // 分, #22
+    0x5217, // 列, #410
+    0x521B, // 创, #399
+    0x5229, // 利, #296
+    0x522B, // 别, #372
+    0x5230, // 到, #33
+    0x5236, // 制, #192
+    0x524D, // 前, #117
+    0x529B, // 力, #173
+    0x529E, // 办, #436
+    0x529F, // 功, #455
+    0x52A0, // 加, #97
+    0x52A1, // 务, #100
+    0x52A8, // 动, #46
+    0x52A9, // 助, #365
+    0x5305, // 包, #331
+    0x5316, // 化, #155
+    0x5317, // 北, #194
+    0x533A, // 区, #105
+    0x533B, // 医, #234
+    0x5341, // 十, #294
+    0x534E, // 华, #205
+    0x5355, // 单, #259
+    0x5357, // 南, #182
+    0x535A, // 博, #153
+    0x5361, // 卡, #332
+    0x539F, // 原, #271
+    0x53BB, // 去, #282
+    0x53C2, // 参, #500
+    0x53CA, // 及, #255
+    0x53CB, // 友, #186
+    0x53CD, // 反, #422
+    0x53D1, // 发, #15
+    0x53D7, // 受, #507
+    0x53D8, // 变, #395
+    0x53E3, // 口, #293
+    0x53EA, // 只, #340
+    0x53EF, // 可, #45
+    0x53F0, // 台, #267
+    0x53F7, // 号, #121
+    0x53F8, // 司, #150
+    0x5404, // 各, #491
+    0x5408, // 合, #115
+    0x540C, // 同, #189
+    0x540D, // 名, #127
+    0x540E, // 后, #75
+    0x5411, // 向, #459
+    0x5427, // 吧, #353
+    0x544A, // 告, #318
+    0x5458, // 员, #232
+    0x5468, // 周, #347
+    0x548C, // 和, #43
+    0x54C1, // 品, #36
+    0x5546, // 商, #148
+    0x5668, // 器, #228
+    0x56DB, // 四, #352
+    0x56DE, // 回, #38
+    0x56E0, // 因, #355
+    0x56E2, // 团, #412
+    0x56ED, // 园, #470
+    0x56FD, // 国, #12
+    0x56FE, // 图, #32
+    0x5728, // 在, #10
+    0x5730, // 地, #30
+    0x573A, // 场, #177
+    0x575B, // 坛, #364
+    0x578B, // 型, #274
+    0x57CE, // 城, #172
+    0x57FA, // 基, #315
+    0x58EB, // 士, #434
+    0x58F0, // 声, #397
+    0x5904, // 处, #416
+    0x5907, // 备, #270
+    0x590D, // 复, #122
+    0x5916, // 外, #190
+    0x591A, // 多, #40
+    0x5927, // 大, #8
+    0x5929, // 天, #52
+    0x592A, // 太, #456
+    0x5934, // 头, #258
+    0x5973, // 女, #65
+    0x597D, // 好, #62
+    0x5982, // 如, #135
+    0x5A31, // 娱, #452
+    0x5B50, // 子, #37
+    0x5B57, // 字, #285
+    0x5B66, // 学, #19
+    0x5B89, // 安, #144
+    0x5B8C, // 完, #469
+    0x5B9A, // 定, #179
+    0x5B9D, // 宝, #188
+    0x5B9E, // 实, #154
+    0x5BA2, // 客, #174
+    0x5BB6, // 家, #26
+    0x5BB9, // 容, #307
+    0x5BC6, // 密, #471
+    0x5BF9, // 对, #90
+    0x5BFC, // 导, #348
+    0x5C06, // 将, #265
+    0x5C0F, // 小, #28
+    0x5C11, // 少, #379
+    0x5C14, // 尔, #490
+    0x5C31, // 就, #101
+    0x5C55, // 展, #291
+    0x5C71, // 山, #239
+    0x5DDE, // 州, #227
+    0x5DE5, // 工, #73
+    0x5DF1, // 己, #480
+    0x5DF2, // 已, #310
+    0x5E02, // 市, #78
+    0x5E03, // 布, #350
+    0x5E08, // 师, #277
+    0x5E16, // 帖, #396
+    0x5E26, // 带, #449
+    0x5E2E, // 帮, #461
+    0x5E38, // 常, #319
+    0x5E73, // 平, #217
+    0x5E74, // 年, #20
+    0x5E76, // 并, #440
+    0x5E7F, // 广, #166
+    0x5E93, // 库, #446
+    0x5E94, // 应, #187
+    0x5E97, // 店, #320
+    0x5EA6, // 度, #114
+    0x5EB7, // 康, #499
+    0x5EFA, // 建, #211
+    0x5F00, // 开, #72
+    0x5F0F, // 式, #207
+    0x5F15, // 引, #495
+    0x5F20, // 张, #385
+    0x5F3A, // 强, #404
+    0x5F53, // 当, #233
+    0x5F55, // 录, #146
+    0x5F62, // 形, #494
+    0x5F69, // 彩, #356
+    0x5F71, // 影, #214
+    0x5F88, // 很, #300
+    0x5F97, // 得, #193
+    0x5FAE, // 微, #245
+    0x5FC3, // 心, #70
+    0x5FEB, // 快, #324
+    0x6001, // 态, #508
+    0x600E, // 怎, #370
+    0x6027, // 性, #99
+    0x603B, // 总, #398
+    0x606F, // 息, #176
+    0x60A8, // 您, #251
+    0x60C5, // 情, #87
+    0x60F3, // 想, #290
+    0x610F, // 意, #184
+    0x611F, // 感, #253
+    0x620F, // 戏, #237
+    0x6210, // 成, #71
+    0x6211, // 我, #11
+    0x6216, // 或, #321
+    0x6218, // 战, #369
+    0x6237, // 户, #215
+    0x623F, // 房, #236
+    0x6240, // 所, #147
+    0x624B, // 手, #55
+    0x624D, // 才, #407
+    0x6253, // 打, #281
+    0x6280, // 技, #203
+    0x6295, // 投, #408
+    0x62A4, // 护, #502
+    0x62A5, // 报, #113
+    0x62DB, // 招, #363
+    0x6301, // 持, #403
+    0x6307, // 指, #414
+    0x636E, // 据, #409
+    0x6392, // 排, #377
+    0x63A5, // 接, #266
+    0x63A8, // 推, #244
+    0x63D0, // 提, #181
+    0x641C, // 搜, #301
+    0x64AD, // 播, #401
+    0x652F, // 支, #400
+    0x6536, // 收, #158
+    0x653E, // 放, #317
+    0x653F, // 政, #380
+    0x6548, // 效, #496
+    0x6559, // 教, #170
+    0x6570, // 数, #136
+    0x6587, // 文, #21
+    0x6599, // 料, #295
+    0x65AF, // 斯, #473
+    0x65B0, // 新, #14
+    0x65B9, // 方, #68
+    0x65C5, // 旅, #457
+    0x65E0, // 无, #164
+    0x65E5, // 日, #50
+    0x65F6, // 时, #18
+    0x660E, // 明, #132
+    0x6613, // 易, #428
+    0x661F, // 星, #240
+    0x662F, // 是, #6
+    0x663E, // 显, #486
+    0x66F4, // 更, #103
+    0x6700, // 最, #61
+    0x6708, // 月, #80
+    0x6709, // 有, #5
+    0x670D, // 服, #94
+    0x671F, // 期, #139
+    0x672C, // 本, #56
+    0x672F, // 术, #216
+    0x673A, // 机, #27
+    0x6743, // 权, #250
+    0x6761, // 条, #309
+    0x6765, // 来, #42
+    0x677F, // 板, #505
+    0x6797, // 林, #475
+    0x679C, // 果, #212
+    0x67E5, // 查, #165
+    0x6807, // 标, #269
+    0x6821, // 校, #462
+    0x6837, // 样, #314
+    0x683C, // 格, #238
+    0x6848, // 案, #378
+    0x697C, // 楼, #342
+    0x6A21, // 模, #413
+    0x6B21, // 次, #263
+    0x6B22, // 欢, #443
+    0x6B3E, // 款, #358
+    0x6B63, // 正, #219
+    0x6B64, // 此, #362
+    0x6BD4, // 比, #298
+    0x6C11, // 民, #279
+    0x6C14, // 气, #303
+    0x6C34, // 水, #163
+    0x6C42, // 求, #373
+    0x6C5F, // 江, #336
+    0x6CA1, // 没, #229
+    0x6CBB, // 治, #425
+    0x6CD5, // 法, #85
+    0x6CE8, // 注, #119
+    0x6D3B, // 活, #231
+    0x6D41, // 流, #280
+    0x6D4B, // 测, #460
+    0x6D77, // 海, #124
+    0x6D88, // 消, #415
+    0x6DF1, // 深, #477
+    0x6E05, // 清, #311
+    0x6E38, // 游, #81
+    0x6E90, // 源, #325
+    0x706B, // 火, #498
+    0x70B9, // 点, #58
+    0x70ED, // 热, #183
+    0x7136, // 然, #308
+    0x7167, // 照, #431
+    0x7231, // 爱, #223
+    0x7247, // 片, #128
+    0x7248, // 版, #91
+    0x724C, // 牌, #429
+    0x7269, // 物, #169
+    0x7279, // 特, #224
+    0x738B, // 王, #351
+    0x73A9, // 玩, #476
+    0x73B0, // 现, #125
+    0x7403, // 球, #367
+    0x7406, // 理, #69
+    0x751F, // 生, #24
+    0x7528, // 用, #17
+    0x7531, // 由, #441
+    0x7535, // 电, #34
+    0x7537, // 男, #275
+    0x754C, // 界, #419
+    0x75C5, // 病, #371
+    0x767B, // 登, #204
+    0x767D, // 白, #338
+    0x767E, // 百, #157
+    0x7684, // 的, #1
+    0x76D8, // 盘, #493
+    0x76EE, // 目, #261
+    0x76F4, // 直, #391
+    0x76F8, // 相, #143
+    0x7701, // 省, #464
+    0x770B, // 看, #54
+    0x771F, // 真, #249
+    0x7740, // 着, #302
+    0x77E5, // 知, #142
+    0x7801, // 码, #257
+    0x7814, // 研, #387
+    0x793A, // 示, #334
+    0x793E, // 社, #343
+    0x795E, // 神, #330
+    0x798F, // 福, #509
+    0x79BB, // 离, #454
+    0x79CD, // 种, #278
+    0x79D1, // 科, #126
+    0x79EF, // 积, #390
+    0x7A0B, // 程, #209
+    0x7A76, // 究, #504
+    0x7A7A, // 空, #312
+    0x7ACB, // 立, #393
+    0x7AD9, // 站, #107
+    0x7AE0, // 章, #304
+    0x7B2C, // 第, #96
+    0x7B49, // 等, #210
+    0x7B54, // 答, #256
+    0x7B80, // 简, #474
+    0x7BA1, // 管, #221
+    0x7C7B, // 类, #246
+    0x7CBE, // 精, #226
+    0x7CFB, // 系, #89
+    0x7D22, // 索, #354
+    0x7EA2, // 红, #417
+    0x7EA7, // 级, #178
+    0x7EBF, // 线, #108
+    0x7EC4, // 组, #389
+    0x7EC6, // 细, #442
+    0x7ECF, // 经, #74
+    0x7ED3, // 结, #333
+    0x7ED9, // 给, #384
+    0x7EDC, // 络, #472
+    0x7EDF, // 统, #344
+    0x7F16, // 编, #424
+    0x7F51, // 网, #9
+    0x7F6E, // 置, #411
+    0x7F8E, // 美, #60
+    0x8001, // 老, #292
+    0x8003, // 考, #288
+    0x8005, // 者, #106
+    0x800C, // 而, #297
+    0x8054, // 联, #159
+    0x80B2, // 育, #327
+    0x80FD, // 能, #59
+    0x81EA, // 自, #77
+    0x8272, // 色, #198
+    0x8282, // 节, #361
+    0x82B1, // 花, #299
+    0x82F1, // 英, #316
+    0x8350, // 荐, #402
+    0x836F, // 药, #481
+    0x8425, // 营, #394
+    0x85CF, // 藏, #337
+    0x884C, // 行, #41
+    0x8868, // 表, #104
+    0x88AB, // 被, #289
+    0x88C5, // 装, #161
+    0x897F, // 西, #199
+    0x8981, // 要, #48
+    0x89C1, // 见, #360
+    0x89C2, // 观, #423
+    0x89C4, // 规, #453
+    0x89C6, // 视, #120
+    0x89E3, // 解, #264
+    0x8A00, // 言, #433
+    0x8BA1, // 计, #191
+    0x8BA4, // 认, #482
+    0x8BA9, // 让, #421
+    0x8BAE, // 议, #427
+    0x8BAF, // 讯, #388
+    0x8BB0, // 记, #273
+    0x8BBA, // 论, #66
+    0x8BBE, // 设, #162
+    0x8BC1, // 证, #201
+    0x8BC4, // 评, #111
+    0x8BC6, // 识, #463
+    0x8BD5, // 试, #323
+    0x8BDD, // 话, #247
+    0x8BE2, // 询, #432
+    0x8BE5, // 该, #447
+    0x8BE6, // 详, #497
+    0x8BED, // 语, #268
+    0x8BF4, // 说, #112
+    0x8BF7, // 请, #213
+    0x8BFB, // 读, #341
+    0x8C03, // 调, #438
+    0x8D22, // 财, #488
+    0x8D28, // 质, #386
+    0x8D2D, // 购, #260
+    0x8D34, // 贴, #510
+    0x8D39, // 费, #242
+    0x8D44, // 资, #116
+    0x8D77, // 起, #220
+    0x8D85, // 超, #406
+    0x8DEF, // 路, #235
+    0x8EAB, // 身, #262
+    0x8F66, // 车, #82
+    0x8F6C, // 转, #322
+    0x8F7D, // 载, #175
+    0x8FBE, // 达, #435
+    0x8FC7, // 过, #118
+    0x8FD0, // 运, #357
+    0x8FD1, // 近, #492
+    0x8FD8, // 还, #171
+    0x8FD9, // 这, #57
+    0x8FDB, // 进, #160
+    0x8FDE, // 连, #489
+    0x9009, // 选, #328
+    0x901A, // 通, #137
+    0x901F, // 速, #458
+    0x9020, // 造, #511
+    0x9053, // 道, #79
+    0x90A3, // 那, #305
+    0x90E8, // 部, #102
+    0x90FD, // 都, #167
+    0x914D, // 配, #479
+    0x9152, // 酒, #444
+    0x91CC, // 里, #196
+    0x91CD, // 重, #230
+    0x91CF, // 量, #248
+    0x91D1, // 金, #134
+    0x9500, // 销, #465
+    0x957F, // 长, #152
+    0x95E8, // 门, #185
+    0x95EE, // 问, #92
+    0x95F4, // 间, #88
+    0x95FB, // 闻, #313
+    0x9605, // 阅, #467
+    0x9633, // 阳, #420
+    0x9645, // 际, #501
+    0x9650, // 限, #286
+    0x9662, // 院, #276
+    0x96C6, // 集, #284
+    0x9700, // 需, #405
+    0x9762, // 面, #123
+    0x97F3, // 音, #335
+    0x9875, // 页, #63
+    0x9879, // 项, #506
+    0x9891, // 频, #200
+    0x9898, // 题, #110
+    0x98CE, // 风, #252
+    0x98DF, // 食, #445
+    0x9996, // 首, #149
+    0x9999, // 香, #512
+    0x9A6C, // 马, #392
+    0x9A8C, // 验, #382
+    0x9AD8, // 高, #67
+    0x9F99, // 龙, #426
+};
+// the percentage of the sample covered by the above characters
+static const float frequent_zhCN_coverage=0.718950369339973;
+
+// The 512 most frequently occuring characters for the zhTW language in a sample of the Internet.
+// Ordered by codepoint, comment shows character and ranking by frequency
+const uint16_t frequent_zhTW[] = {
+    0x4E00, // 一, #2
+    0x4E09, // 三, #131
+    0x4E0A, // 上, #12
+    0x4E0B, // 下, #37
+    0x4E0D, // 不, #6
+    0x4E16, // 世, #312
+    0x4E26, // 並, #434
+    0x4E2D, // 中, #9
+    0x4E3B, // 主, #97
+    0x4E4B, // 之, #55
+    0x4E5F, // 也, #95
+    0x4E86, // 了, #19
+    0x4E8B, // 事, #128
+    0x4E8C, // 二, #187
+    0x4E94, // 五, #339
+    0x4E9B, // 些, #435
+    0x4E9E, // 亞, #432
+    0x4EA4, // 交, #264
+    0x4EAB, // 享, #160
+    0x4EBA, // 人, #3
+    0x4EC0, // 什, #483
+    0x4ECA, // 今, #380
+    0x4ECB, // 介, #468
+    0x4ED6, // 他, #65
+    0x4EE3, // 代, #284
+    0x4EE5, // 以, #26
+    0x4EF6, // 件, #234
+    0x4EFB, // 任, #381
+    0x4EFD, // 份, #447
+    0x4F46, // 但, #281
+    0x4F4D, // 位, #202
+    0x4F4F, // 住, #471
+    0x4F55, // 何, #334
+    0x4F5C, // 作, #56
+    0x4F60, // 你, #64
+    0x4F7F, // 使, #236
+    0x4F86, // 來, #38
+    0x4F9B, // 供, #397
+    0x4FBF, // 便, #440
+    0x4FC2, // 係, #506
+    0x4FDD, // 保, #161
+    0x4FE1, // 信, #268
+    0x4FEE, // 修, #473
+    0x500B, // 個, #27
+    0x5011, // 們, #109
+    0x505A, // 做, #383
+    0x5065, // 健, #415
+    0x5099, // 備, #461
+    0x50B3, // 傳, #277
+    0x50CF, // 像, #403
+    0x50F9, // 價, #93
+    0x512A, // 優, #396
+    0x5143, // 元, #158
+    0x5148, // 先, #382
+    0x5149, // 光, #216
+    0x514D, // 免, #321
+    0x5152, // 兒, #374
+    0x5165, // 入, #58
+    0x5167, // 內, #106
+    0x5168, // 全, #67
+    0x5169, // 兩, #322
+    0x516C, // 公, #53
+    0x516D, // 六, #493
+    0x5171, // 共, #456
+    0x5176, // 其, #148
+    0x5177, // 具, #328
+    0x518A, // 冊, #360
+    0x518D, // 再, #311
+    0x51FA, // 出, #44
+    0x5206, // 分, #15
+    0x5217, // 列, #259
+    0x5225, // 別, #361
+    0x5229, // 利, #251
+    0x5230, // 到, #29
+    0x5247, // 則, #511
+    0x524D, // 前, #82
+    0x5275, // 創, #409
+    0x529B, // 力, #176
+    0x529F, // 功, #430
+    0x52A0, // 加, #87
+    0x52A9, // 助, #465
+    0x52D5, // 動, #48
+    0x52D9, // 務, #102
+    0x5305, // 包, #248
+    0x5316, // 化, #223
+    0x5317, // 北, #145
+    0x5340, // 區, #60
+    0x5341, // 十, #242
+    0x5357, // 南, #261
+    0x535A, // 博, #484
+    0x5361, // 卡, #327
+    0x5370, // 印, #498
+    0x5373, // 即, #351
+    0x539F, // 原, #237
+    0x53BB, // 去, #190
+    0x53C3, // 參, #444
+    0x53C8, // 又, #426
+    0x53CA, // 及, #136
+    0x53CB, // 友, #142
+    0x53D6, // 取, #422
+    0x53D7, // 受, #410
+    0x53E3, // 口, #357
+    0x53EA, // 只, #250
+    0x53EF, // 可, #35
+    0x53F0, // 台, #34
+    0x53F8, // 司, #226
+    0x5403, // 吃, #362
+    0x5404, // 各, #454
+    0x5408, // 合, #147
+    0x540C, // 同, #173
+    0x540D, // 名, #108
+    0x544A, // 告, #186
+    0x548C, // 和, #130
+    0x54C1, // 品, #23
+    0x54E1, // 員, #150
+    0x5546, // 商, #75
+    0x554F, // 問, #120
+    0x559C, // 喜, #502
+    0x55AE, // 單, #210
+    0x55CE, // 嗎, #443
+    0x5668, // 器, #305
+    0x56DB, // 四, #318
+    0x56DE, // 回, #59
+    0x56E0, // 因, #253
+    0x570B, // 國, #21
+    0x5712, // 園, #345
+    0x5716, // 圖, #73
+    0x5718, // 團, #338
+    0x5728, // 在, #11
+    0x5730, // 地, #50
+    0x578B, // 型, #270
+    0x57CE, // 城, #466
+    0x57FA, // 基, #349
+    0x5831, // 報, #127
+    0x5834, // 場, #165
+    0x58EB, // 士, #372
+    0x5916, // 外, #152
+    0x591A, // 多, #54
+    0x5927, // 大, #8
+    0x5929, // 天, #43
+    0x592A, // 太, #343
+    0x5947, // 奇, #325
+    0x5973, // 女, #85
+    0x5979, // 她, #420
+    0x597D, // 好, #22
+    0x5982, // 如, #144
+    0x5B50, // 子, #46
+    0x5B57, // 字, #275
+    0x5B78, // 學, #49
+    0x5B89, // 安, #239
+    0x5B8C, // 完, #320
+    0x5B9A, // 定, #159
+    0x5BA2, // 客, #188
+    0x5BB6, // 家, #31
+    0x5BB9, // 容, #244
+    0x5BE6, // 實, #198
+    0x5BF6, // 寶, #367
+    0x5C07, // 將, #232
+    0x5C08, // 專, #133
+    0x5C0B, // 尋, #352
+    0x5C0D, // 對, #126
+    0x5C0E, // 導, #418
+    0x5C0F, // 小, #20
+    0x5C11, // 少, #368
+    0x5C31, // 就, #63
+    0x5C55, // 展, #341
+    0x5C71, // 山, #273
+    0x5DE5, // 工, #121
+    0x5DF1, // 己, #402
+    0x5DF2, // 已, #299
+    0x5E02, // 市, #81
+    0x5E2B, // 師, #262
+    0x5E36, // 帶, #470
+    0x5E38, // 常, #303
+    0x5E73, // 平, #297
+    0x5E74, // 年, #30
+    0x5E97, // 店, #171
+    0x5EA6, // 度, #220
+    0x5EB7, // 康, #441
+    0x5EE3, // 廣, #279
+    0x5EFA, // 建, #254
+    0x5F0F, // 式, #155
+    0x5F15, // 引, #346
+    0x5F35, // 張, #366
+    0x5F37, // 強, #437
+    0x5F71, // 影, #94
+    0x5F88, // 很, #177
+    0x5F8C, // 後, #66
+    0x5F97, // 得, #113
+    0x5F9E, // 從, #436
+    0x5FC3, // 心, #57
+    0x5FEB, // 快, #292
+    0x6027, // 性, #175
+    0x606F, // 息, #378
+    0x60A8, // 您, #252
+    0x60C5, // 情, #123
+    0x60F3, // 想, #178
+    0x610F, // 意, #168
+    0x611B, // 愛, #125
+    0x611F, // 感, #211
+    0x61C9, // 應, #164
+    0x6210, // 成, #86
+    0x6211, // 我, #7
+    0x6216, // 或, #199
+    0x6230, // 戰, #438
+    0x6232, // 戲, #309
+    0x6236, // 戶, #497
+    0x623F, // 房, #274
+    0x6240, // 所, #79
+    0x624B, // 手, #68
+    0x624D, // 才, #400
+    0x6253, // 打, #278
+    0x627E, // 找, #449
+    0x6280, // 技, #332
+    0x6295, // 投, #425
+    0x62C9, // 拉, #500
+    0x62CD, // 拍, #398
+    0x6307, // 指, #407
+    0x6392, // 排, #458
+    0x63A5, // 接, #326
+    0x63A8, // 推, #153
+    0x63D0, // 提, #235
+    0x641C, // 搜, #314
+    0x6469, // 摩, #472
+    0x6536, // 收, #249
+    0x6539, // 改, #508
+    0x653E, // 放, #331
+    0x653F, // 政, #295
+    0x6559, // 教, #184
+    0x6574, // 整, #394
+    0x6578, // 數, #134
+    0x6587, // 文, #16
+    0x6599, // 料, #167
+    0x65AF, // 斯, #476
+    0x65B0, // 新, #10
+    0x65B9, // 方, #96
+    0x65BC, // 於, #70
+    0x65C5, // 旅, #289
+    0x65E5, // 日, #18
+    0x660E, // 明, #118
+    0x6613, // 易, #482
+    0x661F, // 星, #205
+    0x662F, // 是, #5
+    0x6642, // 時, #13
+    0x66F4, // 更, #149
+    0x66F8, // 書, #209
+    0x6700, // 最, #51
+    0x6703, // 會, #14
+    0x6708, // 月, #25
+    0x6709, // 有, #4
+    0x670D, // 服, #99
+    0x671F, // 期, #139
+    0x672A, // 未, #404
+    0x672C, // 本, #45
+    0x6771, // 東, #221
+    0x677F, // 板, #364
+    0x6797, // 林, #330
+    0x679C, // 果, #179
+    0x67E5, // 查, #283
+    0x683C, // 格, #157
+    0x6848, // 案, #392
+    0x689D, // 條, #406
+    0x696D, // 業, #103
+    0x6A02, // 樂, #116
+    0x6A13, // 樓, #411
+    0x6A19, // 標, #384
+    0x6A23, // 樣, #306
+    0x6A5F, // 機, #40
+    0x6AA2, // 檢, #359
+    0x6B0A, // 權, #228
+    0x6B21, // 次, #227
+    0x6B3E, // 款, #276
+    0x6B4C, // 歌, #496
+    0x6B61, // 歡, #427
+    0x6B63, // 正, #206
+    0x6B64, // 此, #247
+    0x6BCF, // 每, #391
+    0x6BD4, // 比, #257
+    0x6C11, // 民, #230
+    0x6C23, // 氣, #200
+    0x6C34, // 水, #140
+    0x6C42, // 求, #501
+    0x6C92, // 沒, #162
+    0x6CD5, // 法, #89
+    0x6D3B, // 活, #124
+    0x6D41, // 流, #315
+    0x6D77, // 海, #258
+    0x6D88, // 消, #342
+    0x6E05, // 清, #329
+    0x6E2F, // 港, #293
+    0x6F14, // 演, #491
+    0x7063, // 灣, #195
+    0x70BA, // 為, #39
+    0x7121, // 無, #107
+    0x7136, // 然, #215
+    0x7167, // 照, #376
+    0x71B1, // 熱, #245
+    0x7247, // 片, #90
+    0x7248, // 版, #112
+    0x724C, // 牌, #467
+    0x7269, // 物, #110
+    0x7279, // 特, #183
+    0x738B, // 王, #287
+    0x73A9, // 玩, #354
+    0x73FE, // 現, #143
+    0x7403, // 球, #350
+    0x7406, // 理, #105
+    0x751F, // 生, #24
+    0x7522, // 產, #201
+    0x7528, // 用, #17
+    0x7531, // 由, #288
+    0x7537, // 男, #298
+    0x754C, // 界, #399
+    0x7559, // 留, #218
+    0x756B, // 畫, #412
+    0x7576, // 當, #185
+    0x767B, // 登, #138
+    0x767C, // 發, #28
+    0x767D, // 白, #377
+    0x767E, // 百, #393
+    0x7684, // 的, #1
+    0x76EE, // 目, #271
+    0x76F4, // 直, #379
+    0x76F8, // 相, #98
+    0x770B, // 看, #52
+    0x771F, // 真, #180
+    0x773C, // 眼, #433
+    0x77E5, // 知, #170
+    0x78BC, // 碼, #481
+    0x793A, // 示, #353
+    0x793E, // 社, #333
+    0x795E, // 神, #304
+    0x7968, // 票, #477
+    0x798F, // 福, #494
+    0x79C1, // 私, #507
+    0x79D1, // 科, #280
+    0x7A0B, // 程, #272
+    0x7A2E, // 種, #337
+    0x7A4D, // 積, #385
+    0x7A7A, // 空, #324
+    0x7ACB, // 立, #286
+    0x7AD9, // 站, #117
+    0x7AE0, // 章, #141
+    0x7B2C, // 第, #135
+    0x7B49, // 等, #240
+    0x7BA1, // 管, #340
+    0x7BC0, // 節, #431
+    0x7BC7, // 篇, #479
+    0x7C21, // 簡, #499
+    0x7CBE, // 精, #213
+    0x7CFB, // 系, #212
+    0x7D04, // 約, #462
+    0x7D05, // 紅, #452
+    0x7D1A, // 級, #267
+    0x7D30, // 細, #486
+    0x7D44, // 組, #335
+    0x7D50, // 結, #243
+    0x7D66, // 給, #355
+    0x7D71, // 統, #375
+    0x7D93, // 經, #111
+    0x7DB2, // 網, #32
+    0x7DDA, // 線, #151
+    0x7E23, // 縣, #439
+    0x7E3D, // 總, #370
+    0x7F8E, // 美, #41
+    0x7FA9, // 義, #504
+    0x8001, // 老, #290
+    0x8003, // 考, #428
+    0x8005, // 者, #92
+    0x800C, // 而, #217
+    0x805E, // 聞, #181
+    0x806F, // 聯, #310
+    0x8072, // 聲, #413
+    0x80A1, // 股, #390
+    0x80B2, // 育, #453
+    0x80FD, // 能, #71
+    0x8166, // 腦, #408
+    0x81EA, // 自, #61
+    0x81F3, // 至, #344
+    0x8207, // 與, #84
+    0x8209, // 舉, #463
+    0x8272, // 色, #192
+    0x82B1, // 花, #255
+    0x82F1, // 英, #348
+    0x83EF, // 華, #196
+    0x842C, // 萬, #316
+    0x843D, // 落, #308
+    0x8457, // 著, #233
+    0x85A6, // 薦, #401
+    0x85CF, // 藏, #503
+    0x85DD, // 藝, #488
+    0x8655, // 處, #419
+    0x865F, // 號, #191
+    0x884C, // 行, #47
+    0x8853, // 術, #395
+    0x8868, // 表, #77
+    0x88AB, // 被, #291
+    0x88DD, // 裝, #256
+    0x88E1, // 裡, #369
+    0x88FD, // 製, #510
+    0x897F, // 西, #300
+    0x8981, // 要, #36
+    0x898B, // 見, #307
+    0x8996, // 視, #204
+    0x89BA, // 覺, #450
+    0x89BD, // 覽, #387
+    0x89C0, // 觀, #365
+    0x89E3, // 解, #323
+    0x8A00, // 言, #169
+    0x8A02, // 訂, #423
+    0x8A08, // 計, #225
+    0x8A0A, // 訊, #156
+    0x8A0E, // 討, #373
+    0x8A18, // 記, #222
+    0x8A2D, // 設, #174
+    0x8A3B, // 註, #356
+    0x8A55, // 評, #246
+    0x8A66, // 試, #448
+    0x8A71, // 話, #229
+    0x8A72, // 該, #446
+    0x8A8D, // 認, #464
+    0x8A9E, // 語, #371
+    0x8AAA, // 說, #91
+    0x8ABF, // 調, #509
+    0x8ACB, // 請, #119
+    0x8AD6, // 論, #114
+    0x8B1D, // 謝, #389
+    0x8B49, // 證, #429
+    0x8B58, // 識, #416
+    0x8B70, // 議, #485
+    0x8B77, // 護, #475
+    0x8B80, // 讀, #386
+    0x8B8A, // 變, #388
+    0x8B93, // 讓, #336
+    0x8CA8, // 貨, #313
+    0x8CB7, // 買, #260
+    0x8CBB, // 費, #203
+    0x8CC7, // 資, #62
+    0x8CE3, // 賣, #294
+    0x8CEA, // 質, #457
+    0x8CFC, // 購, #189
+    0x8D77, // 起, #214
+    0x8D85, // 超, #296
+    0x8DDF, // 跟, #489
+    0x8DEF, // 路, #137
+    0x8EAB, // 身, #197
+    0x8ECA, // 車, #76
+    0x8F09, // 載, #301
+    0x8F49, // 轉, #282
+    0x8FD1, // 近, #414
+    0x9001, // 送, #363
+    0x9019, // 這, #42
+    0x901A, // 通, #207
+    0x901F, // 速, #495
+    0x9020, // 造, #455
+    0x9023, // 連, #285
+    0x9032, // 進, #231
+    0x904A, // 遊, #132
+    0x904B, // 運, #219
+    0x904E, // 過, #101
+    0x9053, // 道, #146
+    0x9054, // 達, #417
+    0x9078, // 選, #182
+    0x9084, // 還, #154
+    0x908A, // 邊, #487
+    0x90A3, // 那, #269
+    0x90E8, // 部, #78
+    0x90FD, // 都, #104
+    0x914D, // 配, #421
+    0x9152, // 酒, #512
+    0x91AB, // 醫, #358
+    0x91CD, // 重, #224
+    0x91CF, // 量, #319
+    0x91D1, // 金, #115
+    0x9304, // 錄, #302
+    0x9577, // 長, #172
+    0x9580, // 門, #193
+    0x958B, // 開, #72
+    0x9593, // 間, #80
+    0x95B1, // 閱, #405
+    0x95DC, // 關, #74
+    0x963F, // 阿, #460
+    0x9650, // 限, #265
+    0x9662, // 院, #474
+    0x9664, // 除, #478
+    0x969B, // 際, #459
+    0x96C6, // 集, #347
+    0x96E2, // 離, #442
+    0x96FB, // 電, #33
+    0x9700, // 需, #445
+    0x975E, // 非, #451
+    0x9762, // 面, #129
+    0x97F3, // 音, #194
+    0x9801, // 頁, #83
+    0x982D, // 頭, #238
+    0x984C, // 題, #122
+    0x985E, // 類, #163
+    0x98A8, // 風, #266
+    0x98DF, // 食, #208
+    0x9910, // 餐, #469
+    0x9928, // 館, #424
+    0x9996, // 首, #166
+    0x9999, // 香, #263
+    0x99AC, // 馬, #317
+    0x9A57, // 驗, #492
+    0x9AD4, // 體, #100
+    0x9AD8, // 高, #88
+    0x9EBC, // 麼, #241
+    0x9EC3, // 黃, #480
+    0x9ED1, // 黑, #490
+    0x9EDE, // 點, #69
+    0x9F8D, // 龍, #505
+};
+// the percentage of the sample covered by the above characters
+static const float frequent_zhTW_coverage=0.704841200026877;
+
+// The 512 most frequently occuring characters for the ja language in a sample of the Internet.
+// Ordered by codepoint, comment shows character and ranking by frequency
+const uint16_t frequent_ja[] = {
+    0x3005, // 々, #352
+    0x3041, // ぁ, #486
+    0x3042, // あ, #50
+    0x3044, // い, #2
+    0x3046, // う, #33
+    0x3048, // え, #83
+    0x304A, // お, #37
+    0x304B, // か, #21
+    0x304C, // が, #17
+    0x304D, // き, #51
+    0x304E, // ぎ, #324
+    0x304F, // く, #38
+    0x3050, // ぐ, #334
+    0x3051, // け, #60
+    0x3052, // げ, #296
+    0x3053, // こ, #34
+    0x3054, // ご, #100
+    0x3055, // さ, #31
+    0x3056, // ざ, #378
+    0x3057, // し, #4
+    0x3058, // じ, #121
+    0x3059, // す, #12
+    0x305A, // ず, #215
+    0x305B, // せ, #86
+    0x305D, // そ, #68
+    0x305F, // た, #11
+    0x3060, // だ, #42
+    0x3061, // ち, #67
+    0x3063, // っ, #23
+    0x3064, // つ, #73
+    0x3066, // て, #7
+    0x3067, // で, #6
+    0x3068, // と, #14
+    0x3069, // ど, #75
+    0x306A, // な, #8
+    0x306B, // に, #5
+    0x306D, // ね, #123
+    0x306E, // の, #1
+    0x306F, // は, #16
+    0x3070, // ば, #150
+    0x3071, // ぱ, #259
+    0x3072, // ひ, #364
+    0x3073, // び, #266
+    0x3075, // ふ, #484
+    0x3076, // ぶ, #330
+    0x3078, // へ, #146
+    0x3079, // べ, #207
+    0x307B, // ほ, #254
+    0x307E, // ま, #18
+    0x307F, // み, #74
+    0x3080, // む, #285
+    0x3081, // め, #78
+    0x3082, // も, #32
+    0x3083, // ゃ, #111
+    0x3084, // や, #85
+    0x3086, // ゆ, #392
+    0x3087, // ょ, #224
+    0x3088, // よ, #63
+    0x3089, // ら, #29
+    0x308A, // り, #28
+    0x308B, // る, #9
+    0x308C, // れ, #35
+    0x308D, // ろ, #127
+    0x308F, // わ, #88
+    0x3092, // を, #19
+    0x3093, // ん, #22
+    0x30A1, // ァ, #193
+    0x30A2, // ア, #27
+    0x30A3, // ィ, #70
+    0x30A4, // イ, #15
+    0x30A6, // ウ, #89
+    0x30A7, // ェ, #134
+    0x30A8, // エ, #81
+    0x30A9, // ォ, #225
+    0x30AA, // オ, #76
+    0x30AB, // カ, #52
+    0x30AC, // ガ, #147
+    0x30AD, // キ, #66
+    0x30AE, // ギ, #246
+    0x30AF, // ク, #25
+    0x30B0, // グ, #39
+    0x30B1, // ケ, #137
+    0x30B2, // ゲ, #200
+    0x30B3, // コ, #46
+    0x30B4, // ゴ, #183
+    0x30B5, // サ, #64
+    0x30B6, // ザ, #221
+    0x30B7, // シ, #48
+    0x30B8, // ジ, #55
+    0x30B9, // ス, #13
+    0x30BA, // ズ, #103
+    0x30BB, // セ, #109
+    0x30BC, // ゼ, #499
+    0x30BD, // ソ, #175
+    0x30BF, // タ, #45
+    0x30C0, // ダ, #104
+    0x30C1, // チ, #71
+    0x30C3, // ッ, #20
+    0x30C4, // ツ, #119
+    0x30C6, // テ, #59
+    0x30C7, // デ, #82
+    0x30C8, // ト, #10
+    0x30C9, // ド, #44
+    0x30CA, // ナ, #102
+    0x30CB, // ニ, #72
+    0x30CD, // ネ, #117
+    0x30CE, // ノ, #192
+    0x30CF, // ハ, #164
+    0x30D0, // バ, #62
+    0x30D1, // パ, #90
+    0x30D2, // ヒ, #398
+    0x30D3, // ビ, #77
+    0x30D4, // ピ, #135
+    0x30D5, // フ, #47
+    0x30D6, // ブ, #56
+    0x30D7, // プ, #43
+    0x30D8, // ヘ, #268
+    0x30D9, // ベ, #157
+    0x30DA, // ペ, #125
+    0x30DB, // ホ, #155
+    0x30DC, // ボ, #168
+    0x30DD, // ポ, #114
+    0x30DE, // マ, #57
+    0x30DF, // ミ, #97
+    0x30E0, // ム, #69
+    0x30E1, // メ, #53
+    0x30E2, // モ, #142
+    0x30E3, // ャ, #93
+    0x30E4, // ヤ, #258
+    0x30E5, // ュ, #79
+    0x30E6, // ユ, #405
+    0x30E7, // ョ, #98
+    0x30E9, // ラ, #26
+    0x30EA, // リ, #30
+    0x30EB, // ル, #24
+    0x30EC, // レ, #41
+    0x30ED, // ロ, #40
+    0x30EF, // ワ, #144
+    0x30F3, // ン, #3
+    0x30F4, // ヴ, #483
+    0x30FD, // ヽ, #501
+    0x4E00, // 一, #84
+    0x4E07, // 万, #337
+    0x4E09, // 三, #323
+    0x4E0A, // 上, #133
+    0x4E0B, // 下, #180
+    0x4E0D, // 不, #277
+    0x4E16, // 世, #385
+    0x4E2D, // 中, #87
+    0x4E3B, // 主, #432
+    0x4E88, // 予, #326
+    0x4E8B, // 事, #95
+    0x4E8C, // 二, #394
+    0x4E95, // 井, #468
+    0x4EA4, // 交, #410
+    0x4EAC, // 京, #260
+    0x4EBA, // 人, #61
+    0x4ECA, // 今, #184
+    0x4ECB, // 介, #358
+    0x4ED5, // 仕, #391
+    0x4ED6, // 他, #256
+    0x4ED8, // 付, #243
+    0x4EE3, // 代, #280
+    0x4EE5, // 以, #216
+    0x4EF6, // 件, #190
+    0x4F1A, // 会, #105
+    0x4F4D, // 位, #177
+    0x4F4F, // 住, #376
+    0x4F53, // 体, #223
+    0x4F55, // 何, #294
+    0x4F5C, // 作, #154
+    0x4F7F, // 使, #233
+    0x4F9B, // 供, #503
+    0x4FA1, // 価, #217
+    0x4FBF, // 便, #511
+    0x4FDD, // 保, #279
+    0x4FE1, // 信, #271
+    0x500B, // 個, #415
+    0x50CF, // 像, #178
+    0x512A, // 優, #403
+    0x5143, // 元, #384
+    0x5148, // 先, #311
+    0x5149, // 光, #488
+    0x5165, // 入, #115
+    0x5168, // 全, #173
+    0x516C, // 公, #287
+    0x5177, // 具, #447
+    0x5185, // 内, #169
+    0x5186, // 円, #131
+    0x5199, // 写, #275
+    0x51FA, // 出, #110
+    0x5206, // 分, #130
+    0x5207, // 切, #401
+    0x521D, // 初, #319
+    0x5225, // 別, #290
+    0x5229, // 利, #226
+    0x5236, // 制, #507
+    0x524D, // 前, #124
+    0x529B, // 力, #272
+    0x52A0, // 加, #249
+    0x52D5, // 動, #120
+    0x52D9, // 務, #421
+    0x52DF, // 募, #476
+    0x5316, // 化, #308
+    0x5317, // 北, #341
+    0x533A, // 区, #348
+    0x539F, // 原, #321
+    0x53C2, // 参, #452
+    0x53CB, // 友, #451
+    0x53D6, // 取, #237
+    0x53D7, // 受, #354
+    0x53E3, // 口, #289
+    0x53E4, // 古, #339
+    0x53EF, // 可, #298
+    0x53F0, // 台, #439
+    0x53F7, // 号, #361
+    0x5408, // 合, #118
+    0x540C, // 同, #263
+    0x540D, // 名, #65
+    0x5411, // 向, #434
+    0x544A, // 告, #386
+    0x5468, // 周, #393
+    0x5473, // 味, #299
+    0x548C, // 和, #350
+    0x54C1, // 品, #96
+    0x54E1, // 員, #293
+    0x5546, // 商, #198
+    0x554F, // 問, #158
+    0x55B6, // 営, #438
+    0x5668, // 器, #366
+    0x56DE, // 回, #143
+    0x56F3, // 図, #444
+    0x56FD, // 国, #153
+    0x5712, // 園, #435
+    0x571F, // 土, #239
+    0x5728, // 在, #351
+    0x5730, // 地, #163
+    0x578B, // 型, #430
+    0x5831, // 報, #112
+    0x5834, // 場, #139
+    0x58F2, // 売, #232
+    0x5909, // 変, #306
+    0x5916, // 外, #222
+    0x591A, // 多, #336
+    0x5927, // 大, #80
+    0x5929, // 天, #278
+    0x5973, // 女, #161
+    0x597D, // 好, #349
+    0x5A5A, // 婚, #479
+    0x5B50, // 子, #113
+    0x5B57, // 字, #492
+    0x5B66, // 学, #132
+    0x5B89, // 安, #295
+    0x5B9A, // 定, #145
+    0x5B9F, // 実, #220
+    0x5BA4, // 室, #482
+    0x5BAE, // 宮, #487
+    0x5BB6, // 家, #211
+    0x5BB9, // 容, #333
+    0x5BFE, // 対, #252
+    0x5C02, // 専, #474
+    0x5C0F, // 小, #212
+    0x5C11, // 少, #377
+    0x5C4B, // 屋, #284
+    0x5C71, // 山, #206
+    0x5CA1, // 岡, #429
+    0x5CF6, // 島, #297
+    0x5DDD, // 川, #253
+    0x5DE5, // 工, #374
+    0x5E02, // 市, #159
+    0x5E2F, // 帯, #416
+    0x5E38, // 常, #437
+    0x5E73, // 平, #390
+    0x5E74, // 年, #54
+    0x5E83, // 広, #367
+    0x5E97, // 店, #149
+    0x5EA6, // 度, #269
+    0x5EAB, // 庫, #380
+    0x5F0F, // 式, #265
+    0x5F15, // 引, #345
+    0x5F37, // 強, #446
+    0x5F53, // 当, #240
+    0x5F62, // 形, #502
+    0x5F8C, // 後, #230
+    0x5F97, // 得, #490
+    0x5FC3, // 心, #307
+    0x5FC5, // 必, #422
+    0x5FDC, // 応, #356
+    0x601D, // 思, #189
+    0x6027, // 性, #201
+    0x6075, // 恵, #400
+    0x60C5, // 情, #140
+    0x60F3, // 想, #477
+    0x610F, // 意, #305
+    0x611B, // 愛, #273
+    0x611F, // 感, #257
+    0x6210, // 成, #262
+    0x6226, // 戦, #365
+    0x6240, // 所, #236
+    0x624B, // 手, #160
+    0x6295, // 投, #129
+    0x6301, // 持, #355
+    0x6307, // 指, #425
+    0x63A2, // 探, #369
+    0x63B2, // 掲, #399
+    0x643A, // 携, #459
+    0x652F, // 支, #512
+    0x653E, // 放, #469
+    0x6559, // 教, #270
+    0x6570, // 数, #181
+    0x6587, // 文, #202
+    0x6599, // 料, #106
+    0x65B0, // 新, #99
+    0x65B9, // 方, #126
+    0x65C5, // 旅, #445
+    0x65E5, // 日, #36
+    0x660E, // 明, #300
+    0x6620, // 映, #418
+    0x6642, // 時, #107
+    0x66F4, // 更, #359
+    0x66F8, // 書, #174
+    0x6700, // 最, #152
+    0x6708, // 月, #49
+    0x6709, // 有, #302
+    0x671F, // 期, #332
+    0x6728, // 木, #203
+    0x672C, // 本, #92
+    0x6750, // 材, #489
+    0x6751, // 村, #466
+    0x6765, // 来, #267
+    0x6771, // 東, #191
+    0x677F, // 板, #411
+    0x679C, // 果, #441
+    0x6821, // 校, #327
+    0x682A, // 株, #412
+    0x683C, // 格, #228
+    0x691C, // 検, #179
+    0x696D, // 業, #166
+    0x697D, // 楽, #172
+    0x69D8, // 様, #255
+    0x6A5F, // 機, #235
+    0x6B21, // 次, #318
+    0x6B62, // 止, #475
+    0x6B63, // 正, #312
+    0x6C17, // 気, #116
+    0x6C34, // 水, #165
+    0x6C42, // 求, #465
+    0x6C7A, // 決, #370
+    0x6CBB, // 治, #505
+    0x6CC1, // 況, #462
+    0x6CD5, // 法, #227
+    0x6CE8, // 注, #372
+    0x6D3B, // 活, #303
+    0x6D41, // 流, #480
+    0x6D77, // 海, #274
+    0x6E08, // 済, #417
+    0x6F14, // 演, #504
+    0x706B, // 火, #264
+    0x70B9, // 点, #331
+    0x7121, // 無, #58
+    0x7248, // 版, #409
+    0x7269, // 物, #170
+    0x7279, // 特, #242
+    0x72B6, // 状, #458
+    0x73FE, // 現, #322
+    0x7406, // 理, #162
+    0x751F, // 生, #122
+    0x7523, // 産, #320
+    0x7528, // 用, #94
+    0x7530, // 田, #195
+    0x7537, // 男, #373
+    0x753A, // 町, #314
+    0x753B, // 画, #91
+    0x754C, // 界, #436
+    0x756A, // 番, #261
+    0x75C5, // 病, #428
+    0x767A, // 発, #194
+    0x767B, // 登, #231
+    0x767D, // 白, #419
+    0x7684, // 的, #251
+    0x76EE, // 目, #197
+    0x76F4, // 直, #497
+    0x76F8, // 相, #286
+    0x770C, // 県, #199
+    0x771F, // 真, #219
+    0x7740, // 着, #283
+    0x77E5, // 知, #185
+    0x77F3, // 石, #500
+    0x78BA, // 確, #383
+    0x793A, // 示, #241
+    0x793E, // 社, #167
+    0x795E, // 神, #315
+    0x798F, // 福, #423
+    0x79C1, // 私, #347
+    0x79D1, // 科, #420
+    0x7A0E, // 税, #368
+    0x7A2E, // 種, #455
+    0x7A3F, // 稿, #148
+    0x7A7A, // 空, #427
+    0x7ACB, // 立, #309
+    0x7B11, // 笑, #454
+    0x7B2C, // 第, #317
+    0x7B49, // 等, #457
+    0x7B54, // 答, #426
+    0x7BA1, // 管, #481
+    0x7CFB, // 系, #408
+    0x7D04, // 約, #276
+    0x7D20, // 素, #407
+    0x7D22, // 索, #214
+    0x7D30, // 細, #381
+    0x7D39, // 紹, #471
+    0x7D42, // 終, #456
+    0x7D44, // 組, #424
+    0x7D4C, // 経, #360
+    0x7D50, // 結, #291
+    0x7D9A, // 続, #357
+    0x7DCF, // 総, #467
+    0x7DDA, // 線, #338
+    0x7DE8, // 編, #453
+    0x7F8E, // 美, #204
+    0x8003, // 考, #387
+    0x8005, // 者, #151
+    0x805E, // 聞, #463
+    0x8077, // 職, #363
+    0x80B2, // 育, #433
+    0x80FD, // 能, #250
+    0x8179, // 腹, #396
+    0x81EA, // 自, #156
+    0x826F, // 良, #329
+    0x8272, // 色, #402
+    0x82B1, // 花, #440
+    0x82B8, // 芸, #413
+    0x82F1, // 英, #485
+    0x8449, // 葉, #472
+    0x884C, // 行, #128
+    0x8853, // 術, #460
+    0x8868, // 表, #209
+    0x88FD, // 製, #431
+    0x897F, // 西, #406
+    0x8981, // 要, #313
+    0x898B, // 見, #101
+    0x898F, // 規, #375
+    0x89A7, // 覧, #171
+    0x89E3, // 解, #388
+    0x8A00, // 言, #210
+    0x8A08, // 計, #343
+    0x8A18, // 記, #136
+    0x8A2D, // 設, #292
+    0x8A71, // 話, #213
+    0x8A73, // 詳, #371
+    0x8A8D, // 認, #404
+    0x8A9E, // 語, #234
+    0x8AAC, // 説, #494
+    0x8AAD, // 読, #301
+    0x8ABF, // 調, #443
+    0x8AC7, // 談, #448
+    0x8B77, // 護, #509
+    0x8C37, // 谷, #506
+    0x8CA9, // 販, #362
+    0x8CB7, // 買, #346
+    0x8CC7, // 資, #473
+    0x8CEA, // 質, #281
+    0x8CFC, // 購, #495
+    0x8EAB, // 身, #470
+    0x8ECA, // 車, #205
+    0x8EE2, // 転, #335
+    0x8F09, // 載, #342
+    0x8FBC, // 込, #229
+    0x8FD1, // 近, #304
+    0x8FD4, // 返, #461
+    0x8FFD, // 追, #379
+    0x9001, // 送, #186
+    0x901A, // 通, #182
+    0x901F, // 速, #340
+    0x9023, // 連, #244
+    0x904B, // 運, #382
+    0x904E, // 過, #498
+    0x9053, // 道, #282
+    0x9054, // 達, #450
+    0x9055, // 違, #414
+    0x9078, // 選, #288
+    0x90E8, // 部, #208
+    0x90FD, // 都, #344
+    0x914D, // 配, #389
+    0x91CD, // 重, #478
+    0x91CE, // 野, #245
+    0x91D1, // 金, #138
+    0x9332, // 録, #238
+    0x9577, // 長, #247
+    0x9580, // 門, #508
+    0x958B, // 開, #248
+    0x9593, // 間, #141
+    0x95A2, // 関, #188
+    0x962A, // 阪, #496
+    0x9650, // 限, #395
+    0x9662, // 院, #449
+    0x9664, // 除, #510
+    0x969B, // 際, #493
+    0x96C6, // 集, #196
+    0x96D1, // 雑, #442
+    0x96FB, // 電, #187
+    0x9762, // 面, #328
+    0x97F3, // 音, #325
+    0x984C, // 題, #310
+    0x985E, // 類, #491
+    0x98A8, // 風, #353
+    0x98DF, // 食, #218
+    0x9928, // 館, #464
+    0x99C5, // 駅, #316
+    0x9A13, // 験, #397
+    0x9AD8, // 高, #176
+    0xFF57, // ｗ, #108
+};
+// the percentage of the sample covered by the above characters
+static const float frequent_ja_coverage=0.880569589120162;
+
+// The 512 most frequently occuring characters for the ko language in a sample of the Internet.
+// Ordered by codepoint, comment shows character and ranking by frequency
+const uint16_t frequent_ko[] = {
+    0x314B, // ㅋ, #148
+    0x314E, // ㅎ, #390
+    0x3160, // ㅠ, #354
+    0x318D, // ㆍ, #439
+    0xAC00, // 가, #6
+    0xAC01, // 각, #231
+    0xAC04, // 간, #106
+    0xAC08, // 갈, #362
+    0xAC10, // 감, #122
+    0xAC11, // 갑, #493
+    0xAC15, // 강, #155
+    0xAC19, // 같, #264
+    0xAC1C, // 개, #87
+    0xAC1D, // 객, #198
+    0xAC24, // 갤, #457
+    0xAC70, // 거, #91
+    0xAC74, // 건, #161
+    0xAC78, // 걸, #338
+    0xAC80, // 검, #184
+    0xAC83, // 것, #116
+    0xAC8C, // 게, #36
+    0xACA0, // 겠, #233
+    0xACA8, // 겨, #341
+    0xACA9, // 격, #245
+    0xACAC, // 견, #413
+    0xACB0, // 결, #202
+    0xACBD, // 경, #62
+    0xACC4, // 계, #142
+    0xACE0, // 고, #12
+    0xACE1, // 곡, #444
+    0xACE8, // 골, #379
+    0xACF3, // 곳, #388
+    0xACF5, // 공, #59
+    0xACFC, // 과, #69
+    0xAD00, // 관, #95
+    0xAD11, // 광, #235
+    0xAD50, // 교, #128
+    0xAD6C, // 구, #52
+    0xAD6D, // 국, #85
+    0xAD70, // 군, #293
+    0xAD74, // 굴, #487
+    0xAD81, // 궁, #441
+    0xAD8C, // 권, #192
+    0xADC0, // 귀, #386
+    0xADDC, // 규, #367
+    0xADF8, // 그, #30
+    0xADF9, // 극, #424
+    0xADFC, // 근, #241
+    0xAE00, // 글, #61
+    0xAE08, // 금, #138
+    0xAE09, // 급, #269
+    0xAE30, // 기, #3
+    0xAE34, // 긴, #465
+    0xAE38, // 길, #297
+    0xAE40, // 김, #205
+    0xAE4C, // 까, #171
+    0xAED8, // 께, #273
+    0xAF43, // 꽃, #475
+    0xB05D, // 끝, #505
+    0xB07C, // 끼, #490
+    0xB098, // 나, #39
+    0xB09C, // 난, #274
+    0xB0A0, // 날, #292
+    0xB0A8, // 남, #139
+    0xB0B4, // 내, #56
+    0xB108, // 너, #272
+    0xB110, // 널, #476
+    0xB118, // 넘, #492
+    0xB124, // 네, #100
+    0xB137, // 넷, #329
+    0xB140, // 녀, #288
+    0xB144, // 년, #151
+    0xB178, // 노, #149
+    0xB17C, // 논, #491
+    0xB180, // 놀, #464
+    0xB18D, // 농, #442
+    0xB204, // 누, #319
+    0xB208, // 눈, #383
+    0xB274, // 뉴, #173
+    0xB290, // 느, #368
+    0xB294, // 는, #5
+    0xB298, // 늘, #322
+    0xB2A5, // 능, #190
+    0xB2C8, // 니, #16
+    0xB2D8, // 님, #153
+    0xB2E4, // 다, #2
+    0xB2E8, // 단, #134
+    0xB2EB, // 닫, #195
+    0xB2EC, // 달, #243
+    0xB2F4, // 담, #254
+    0xB2F5, // 답, #287
+    0xB2F9, // 당, #159
+    0xB300, // 대, #33
+    0xB313, // 댓, #303
+    0xB354, // 더, #140
+    0xB358, // 던, #252
+    0xB367, // 덧, #463
+    0xB370, // 데, #104
+    0xB378, // 델, #429
+    0xB3C4, // 도, #25
+    0xB3C5, // 독, #301
+    0xB3CC, // 돌, #309
+    0xB3D9, // 동, #58
+    0xB418, // 되, #82
+    0xB41C, // 된, #189
+    0xB420, // 될, #408
+    0xB429, // 됩, #332
+    0xB450, // 두, #199
+    0xB4A4, // 뒤, #496
+    0xB4DC, // 드, #40
+    0xB4E0, // 든, #283
+    0xB4E4, // 들, #54
+    0xB4EF, // 듯, #478
+    0xB4F1, // 등, #90
+    0xB514, // 디, #133
+    0xB529, // 딩, #462
+    0xB530, // 따, #333
+    0xB54C, // 때, #240
+    0xB610, // 또, #313
+    0xB77C, // 라, #42
+    0xB77D, // 락, #355
+    0xB780, // 란, #290
+    0xB78C, // 람, #246
+    0xB78D, // 랍, #420
+    0xB791, // 랑, #270
+    0xB798, // 래, #174
+    0xB799, // 랙, #381
+    0xB79C, // 랜, #357
+    0xB7A8, // 램, #359
+    0xB7A9, // 랩, #402
+    0xB7C9, // 량, #346
+    0xB7EC, // 러, #130
+    0xB7F0, // 런, #312
+    0xB7FC, // 럼, #327
+    0xB7FD, // 럽, #447
+    0xB807, // 렇, #412
+    0xB808, // 레, #114
+    0xB80C, // 렌, #395
+    0xB824, // 려, #158
+    0xB825, // 력, #194
+    0xB828, // 련, #326
+    0xB839, // 령, #389
+    0xB85C, // 로, #4
+    0xB85D, // 록, #84
+    0xB860, // 론, #366
+    0xB8CC, // 료, #154
+    0xB8E8, // 루, #236
+    0xB958, // 류, #265
+    0xB974, // 르, #212
+    0xB978, // 른, #250
+    0xB97C, // 를, #35
+    0xB984, // 름, #276
+    0xB9AC, // 리, #19
+    0xB9AD, // 릭, #394
+    0xB9B0, // 린, #259
+    0xB9B4, // 릴, #485
+    0xB9BC, // 림, #305
+    0xB9BD, // 립, #217
+    0xB9C1, // 링, #351
+    0xB9C8, // 마, #67
+    0xB9C9, // 막, #310
+    0xB9CC, // 만, #65
+    0xB9CE, // 많, #257
+    0xB9D0, // 말, #188
+    0xB9DB, // 맛, #397
+    0xB9DD, // 망, #370
+    0xB9DE, // 맞, #399
+    0xB9E4, // 매, #125
+    0xB9E8, // 맨, #422
+    0xBA38, // 머, #311
+    0xBA39, // 먹, #377
+    0xBA3C, // 먼, #469
+    0xBA54, // 메, #147
+    0xBA70, // 며, #191
+    0xBA74, // 면, #72
+    0xBA85, // 명, #131
+    0xBAA8, // 모, #73
+    0xBAA9, // 목, #157
+    0xBAB0, // 몰, #401
+    0xBAB8, // 몸, #437
+    0xBABB, // 못, #336
+    0xBB34, // 무, #80
+    0xBB38, // 문, #57
+    0xBB3C, // 물, #94
+    0xBBA4, // 뮤, #431
+    0xBBF8, // 미, #76
+    0xBBFC, // 민, #200
+    0xBC00, // 밀, #308
+    0xBC0F, // 및, #249
+    0xBC14, // 바, #89
+    0xBC15, // 박, #226
+    0xBC18, // 반, #175
+    0xBC1B, // 받, #248
+    0xBC1C, // 발, #164
+    0xBC29, // 방, #92
+    0xBC30, // 배, #162
+    0xBC31, // 백, #256
+    0xBC84, // 버, #111
+    0xBC88, // 번, #167
+    0xBC8C, // 벌, #423
+    0xBC94, // 범, #427
+    0xBC95, // 법, #207
+    0xBCA0, // 베, #281
+    0xBCA4, // 벤, #378
+    0xBCA8, // 벨, #387
+    0xBCC0, // 변, #253
+    0xBCC4, // 별, #262
+    0xBCD1, // 병, #340
+    0xBCF4, // 보, #20
+    0xBCF5, // 복, #204
+    0xBCF8, // 본, #182
+    0xBCFC, // 볼, #385
+    0xBD09, // 봉, #405
+    0xBD80, // 부, #46
+    0xBD81, // 북, #261
+    0xBD84, // 분, #105
+    0xBD88, // 불, #225
+    0xBDF0, // 뷰, #350
+    0xBE0C, // 브, #214
+    0xBE14, // 블, #99
+    0xBE44, // 비, #55
+    0xBE4C, // 빌, #510
+    0xBE60, // 빠, #398
+    0xC0AC, // 사, #14
+    0xC0AD, // 삭, #342
+    0xC0B0, // 산, #121
+    0xC0B4, // 살, #279
+    0xC0BC, // 삼, #348
+    0xC0C1, // 상, #41
+    0xC0C8, // 새, #282
+    0xC0C9, // 색, #181
+    0xC0DD, // 생, #109
+    0xC11C, // 서, #21
+    0xC11D, // 석, #234
+    0xC120, // 선, #107
+    0xC124, // 설, #170
+    0xC131, // 성, #50
+    0xC138, // 세, #60
+    0xC139, // 섹, #456
+    0xC13C, // 센, #267
+    0xC154, // 셔, #455
+    0xC158, // 션, #237
+    0xC15C, // 셜, #448
+    0xC168, // 셨, #421
+    0xC18C, // 소, #51
+    0xC18D, // 속, #219
+    0xC190, // 손, #323
+    0xC1A1, // 송, #203
+    0xC1C4, // 쇄, #501
+    0xC1FC, // 쇼, #364
+    0xC218, // 수, #27
+    0xC219, // 숙, #467
+    0xC21C, // 순, #258
+    0xC220, // 술, #302
+    0xC26C, // 쉬, #511
+    0xC288, // 슈, #384
+    0xC2A4, // 스, #11
+    0xC2AC, // 슬, #438
+    0xC2B4, // 슴, #504
+    0xC2B5, // 습, #77
+    0xC2B9, // 승, #299
+    0xC2DC, // 시, #13
+    0xC2DD, // 식, #137
+    0xC2E0, // 신, #47
+    0xC2E4, // 실, #132
+    0xC2EC, // 심, #196
+    0xC2ED, // 십, #482
+    0xC2F6, // 싶, #352
+    0xC2F8, // 싸, #419
+    0xC4F0, // 쓰, #278
+    0xC528, // 씨, #360
+    0xC544, // 아, #23
+    0xC545, // 악, #296
+    0xC548, // 안, #71
+    0xC54A, // 않, #209
+    0xC54C, // 알, #222
+    0xC554, // 암, #460
+    0xC558, // 았, #349
+    0xC559, // 앙, #473
+    0xC55E, // 앞, #434
+    0xC560, // 애, #271
+    0xC561, // 액, #415
+    0xC571, // 앱, #477
+    0xC57C, // 야, #124
+    0xC57D, // 약, #229
+    0xC591, // 양, #177
+    0xC5B4, // 어, #24
+    0xC5B5, // 억, #407
+    0xC5B8, // 언, #294
+    0xC5BC, // 얼, #356
+    0xC5C4, // 엄, #426
+    0xC5C5, // 업, #118
+    0xC5C6, // 없, #178
+    0xC5C8, // 었, #165
+    0xC5D0, // 에, #9
+    0xC5D4, // 엔, #375
+    0xC5D8, // 엘, #506
+    0xC5EC, // 여, #66
+    0xC5ED, // 역, #186
+    0xC5EE, // 엮, #488
+    0xC5F0, // 연, #96
+    0xC5F4, // 열, #266
+    0xC5FC, // 염, #449
+    0xC600, // 였, #374
+    0xC601, // 영, #83
+    0xC608, // 예, #168
+    0xC624, // 오, #75
+    0xC628, // 온, #300
+    0xC62C, // 올, #306
+    0xC640, // 와, #119
+    0xC644, // 완, #361
+    0xC654, // 왔, #489
+    0xC655, // 왕, #418
+    0xC678, // 외, #218
+    0xC694, // 요, #43
+    0xC695, // 욕, #479
+    0xC6A9, // 용, #48
+    0xC6B0, // 우, #64
+    0xC6B1, // 욱, #503
+    0xC6B4, // 운, #108
+    0xC6B8, // 울, #223
+    0xC6C0, // 움, #317
+    0xC6C3, // 웃, #404
+    0xC6CC, // 워, #280
+    0xC6D0, // 원, #45
+    0xC6D4, // 월, #150
+    0xC6E8, // 웨, #446
+    0xC6F9, // 웹, #500
+    0xC704, // 위, #78
+    0xC720, // 유, #81
+    0xC721, // 육, #321
+    0xC724, // 윤, #416
+    0xC73C, // 으, #49
+    0xC740, // 은, #31
+    0xC744, // 을, #17
+    0xC74C, // 음, #112
+    0xC751, // 응, #461
+    0xC758, // 의, #8
+    0xC774, // 이, #1
+    0xC775, // 익, #403
+    0xC778, // 인, #18
+    0xC77C, // 일, #28
+    0xC784, // 임, #160
+    0xC785, // 입, #93
+    0xC788, // 있, #44
+    0xC790, // 자, #22
+    0xC791, // 작, #88
+    0xC798, // 잘, #347
+    0xC7A1, // 잡, #372
+    0xC7A5, // 장, #53
+    0xC7AC, // 재, #120
+    0xC7C1, // 쟁, #483
+    0xC800, // 저, #98
+    0xC801, // 적, #97
+    0xC804, // 전, #34
+    0xC808, // 절, #320
+    0xC810, // 점, #201
+    0xC811, // 접, #331
+    0xC815, // 정, #26
+    0xC81C, // 제, #29
+    0xC838, // 져, #414
+    0xC870, // 조, #86
+    0xC871, // 족, #373
+    0xC874, // 존, #432
+    0xC880, // 좀, #470
+    0xC885, // 종, #208
+    0xC88B, // 좋, #239
+    0xC8E0, // 죠, #451
+    0xC8FC, // 주, #38
+    0xC8FD, // 죽, #471
+    0xC900, // 준, #286
+    0xC904, // 줄, #392
+    0xC911, // 중, #103
+    0xC988, // 즈, #255
+    0xC98C, // 즌, #507
+    0xC990, // 즐, #371
+    0xC99D, // 증, #260
+    0xC9C0, // 지, #10
+    0xC9C1, // 직, #216
+    0xC9C4, // 진, #79
+    0xC9C8, // 질, #238
+    0xC9D1, // 집, #206
+    0xC9DC, // 짜, #411
+    0xC9F8, // 째, #494
+    0xCABD, // 쪽, #435
+    0xCC28, // 차, #146
+    0xCC29, // 착, #443
+    0xCC2C, // 찬, #481
+    0xCC30, // 찰, #440
+    0xCC38, // 참, #343
+    0xCC3D, // 창, #304
+    0xCC3E, // 찾, #335
+    0xCC44, // 채, #284
+    0xCC45, // 책, #298
+    0xCC98, // 처, #242
+    0xCC9C, // 천, #143
+    0xCCA0, // 철, #380
+    0xCCA8, // 첨, #452
+    0xCCAB, // 첫, #484
+    0xCCAD, // 청, #197
+    0xCCB4, // 체, #126
+    0xCCD0, // 쳐, #472
+    0xCD08, // 초, #220
+    0xCD1D, // 총, #406
+    0xCD5C, // 최, #179
+    0xCD94, // 추, #136
+    0xCD95, // 축, #337
+    0xCD9C, // 출, #166
+    0xCDA9, // 충, #369
+    0xCDE8, // 취, #210
+    0xCE20, // 츠, #215
+    0xCE21, // 측, #468
+    0xCE35, // 층, #512
+    0xCE58, // 치, #102
+    0xCE5C, // 친, #325
+    0xCE68, // 침, #263
+    0xCE74, // 카, #115
+    0xCE7C, // 칼, #466
+    0xCE90, // 캐, #454
+    0xCEE4, // 커, #285
+    0xCEE8, // 컨, #328
+    0xCEF4, // 컴, #417
+    0xCF00, // 케, #339
+    0xCF13, // 켓, #509
+    0xCF1C, // 켜, #508
+    0xCF54, // 코, #193
+    0xCF58, // 콘, #391
+    0xCFE0, // 쿠, #393
+    0xD035, // 퀵, #453
+    0xD06C, // 크, #101
+    0xD070, // 큰, #495
+    0xD074, // 클, #289
+    0xD0A4, // 키, #230
+    0xD0C0, // 타, #127
+    0xD0C1, // 탁, #314
+    0xD0C4, // 탄, #450
+    0xD0C8, // 탈, #436
+    0xD0DC, // 태, #221
+    0xD0DD, // 택, #275
+    0xD130, // 터, #70
+    0xD14C, // 테, #213
+    0xD150, // 텐, #324
+    0xD154, // 텔, #430
+    0xD15C, // 템, #382
+    0xD1A0, // 토, #145
+    0xD1B5, // 통, #156
+    0xD22C, // 투, #227
+    0xD2B8, // 트, #37
+    0xD2B9, // 특, #247
+    0xD2F0, // 티, #187
+    0xD305, // 팅, #410
+    0xD30C, // 파, #141
+    0xD310, // 판, #163
+    0xD314, // 팔, #499
+    0xD328, // 패, #307
+    0xD32C, // 팬, #459
+    0xD338, // 팸, #433
+    0xD37C, // 퍼, #344
+    0xD398, // 페, #172
+    0xD3B8, // 편, #251
+    0xD3C9, // 평, #291
+    0xD3EC, // 포, #68
+    0xD3ED, // 폭, #445
+    0xD3F0, // 폰, #318
+    0xD45C, // 표, #232
+    0xD480, // 풀, #497
+    0xD488, // 품, #113
+    0xD48D, // 풍, #425
+    0xD504, // 프, #110
+    0xD508, // 픈, #498
+    0xD50C, // 플, #211
+    0xD53C, // 피, #169
+    0xD544, // 필, #295
+    0xD551, // 핑, #376
+    0xD558, // 하, #7
+    0xD559, // 학, #129
+    0xD55C, // 한, #15
+    0xD560, // 할, #144
+    0xD568, // 함, #152
+    0xD569, // 합, #123
+    0xD56D, // 항, #268
+    0xD574, // 해, #32
+    0xD588, // 했, #180
+    0xD589, // 행, #135
+    0xD5A5, // 향, #345
+    0xD5C8, // 허, #396
+    0xD5D8, // 험, #316
+    0xD5E4, // 헤, #474
+    0xD604, // 현, #185
+    0xD611, // 협, #315
+    0xD615, // 형, #244
+    0xD61C, // 혜, #428
+    0xD638, // 호, #117
+    0xD63C, // 혼, #358
+    0xD648, // 홈, #330
+    0xD64D, // 홍, #363
+    0xD654, // 화, #63
+    0xD655, // 확, #183
+    0xD658, // 환, #224
+    0xD65C, // 활, #277
+    0xD669, // 황, #353
+    0xD68C, // 회, #74
+    0xD68D, // 획, #458
+    0xD69F, // 횟, #409
+    0xD6A8, // 효, #400
+    0xD6C4, // 후, #176
+    0xD6C8, // 훈, #486
+    0xD734, // 휴, #365
+    0xD754, // 흔, #480
+    0xD76C, // 희, #334
+    0xD788, // 히, #228
+    0xD798, // 힘, #502
+};
+// the percentage of the sample covered by the above characters
+static const float frequent_ko_coverage=0.948157021464184;
+

media/libmedia/IAudioFlinger.cpp

diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 5d81e5d..a9a9f1a 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -58,7 +58,7 @@
     RESTORE_OUTPUT,
     OPEN_INPUT,
     CLOSE_INPUT,
-    SET_STREAM_OUTPUT,
+    INVALIDATE_STREAM,
     SET_VOICE_VOLUME,
     GET_RENDER_POSITION,
     GET_INPUT_FRAMES_LOST,
@@ -89,13 +89,14 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                size_t frameCount,
+                                size_t *pFrameCount,
                                 track_flags_t *flags,
                                 const sp<IMemory>& sharedBuffer,
                                 audio_io_handle_t output,
                                 pid_t tid,
                                 int *sessionId,
                                 String8& name,
+                                int clientUid,
                                 status_t *status)
     {
         Parcel data, reply;
@@ -105,6 +106,7 @@
         data.writeInt32(sampleRate);
         data.writeInt32(format);
         data.writeInt32(channelMask);
+        size_t frameCount = pFrameCount != NULL ? *pFrameCount : 0;
         data.writeInt32(frameCount);
         track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT;
         data.writeInt32(lFlags);
@@ -116,15 +118,20 @@
         }
         data.writeInt32((int32_t) output);
         data.writeInt32((int32_t) tid);
-        int lSessionId = 0;
+        int lSessionId = AUDIO_SESSION_ALLOCATE;
         if (sessionId != NULL) {
             lSessionId = *sessionId;
         }
         data.writeInt32(lSessionId);
+        data.writeInt32(clientUid);
         status_t lStatus = remote()->transact(CREATE_TRACK, data, &reply);
         if (lStatus != NO_ERROR) {
             ALOGE("createTrack error: %s", strerror(-lStatus));
         } else {
+            frameCount = reply.readInt32();
+            if (pFrameCount != NULL) {
+                *pFrameCount = frameCount;
+            }
             lFlags = reply.readInt32();
             if (flags != NULL) {
                 *flags = lFlags;
@@ -136,6 +143,17 @@
             name = reply.readString8();
             lStatus = reply.readInt32();
             track = interface_cast<IAudioTrack>(reply.readStrongBinder());
+            if (lStatus == NO_ERROR) {
+                if (track == 0) {
+                    ALOGE("createTrack should have returned an IAudioTrack");
+                    lStatus = UNKNOWN_ERROR;
+                }
+            } else {
+                if (track != 0) {
+                    ALOGE("createTrack returned an IAudioTrack but with status %d", lStatus);
+                    track.clear();
+                }
+            }
         }
         if (status != NULL) {
             *status = lStatus;
@@ -148,7 +166,7 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                size_t frameCount,
+                                size_t *pFrameCount,
                                 track_flags_t *flags,
                                 pid_t tid,
                                 int *sessionId,
@@ -161,11 +179,12 @@
         data.writeInt32(sampleRate);
         data.writeInt32(format);
         data.writeInt32(channelMask);
+        size_t frameCount = pFrameCount != NULL ? *pFrameCount : 0;
         data.writeInt32(frameCount);
         track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT;
         data.writeInt32(lFlags);
         data.writeInt32((int32_t) tid);
-        int lSessionId = 0;
+        int lSessionId = AUDIO_SESSION_ALLOCATE;
         if (sessionId != NULL) {
             lSessionId = *sessionId;
         }
@@ -174,6 +193,10 @@
         if (lStatus != NO_ERROR) {
             ALOGE("openRecord error: %s", strerror(-lStatus));
         } else {
+            frameCount = reply.readInt32();
+            if (pFrameCount != NULL) {
+                *pFrameCount = frameCount;
+            }
             lFlags = reply.readInt32();
             if (flags != NULL) {
                 *flags = lFlags;
@@ -184,6 +207,17 @@
             }
             lStatus = reply.readInt32();
             record = interface_cast<IAudioRecord>(reply.readStrongBinder());
+            if (lStatus == NO_ERROR) {
+                if (record == 0) {
+                    ALOGE("openRecord should have returned an IAudioRecord");
+                    lStatus = UNKNOWN_ERROR;
+                }
+            } else {
+                if (record != 0) {
+                    ALOGE("openRecord returned an IAudioRecord but with status %d", lStatus);
+                    record.clear();
+                }
+            }
         }
         if (status != NULL) {
             *status = lStatus;
@@ -511,13 +545,12 @@
         return reply.readInt32();
     }
 
-    virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
+    virtual status_t invalidateStream(audio_stream_type_t stream)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
         data.writeInt32((int32_t) stream);
-        data.writeInt32((int32_t) output);
-        remote()->transact(SET_STREAM_OUTPUT, data, &reply);
+        remote()->transact(INVALIDATE_STREAM, data, &reply);
         return reply.readInt32();
     }
 
@@ -530,7 +563,7 @@
         return reply.readInt32();
     }
 
-    virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
+    virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
             audio_io_handle_t output) const
     {
         Parcel data, reply;
@@ -551,13 +584,16 @@
         return status;
     }
 
-    virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const
+    virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
         data.writeInt32((int32_t) ioHandle);
-        remote()->transact(GET_INPUT_FRAMES_LOST, data, &reply);
-        return reply.readInt32();
+        status_t status = remote()->transact(GET_INPUT_FRAMES_LOST, data, &reply);
+        if (status != NO_ERROR) {
+            return 0;
+        }
+        return (uint32_t) reply.readInt32();
     }
 
     virtual int newAudioSessionId()
@@ -572,19 +608,21 @@
         return id;
     }
 
-    virtual void acquireAudioSessionId(int audioSession)
+    virtual void acquireAudioSessionId(int audioSession, int pid)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
         data.writeInt32(audioSession);
+        data.writeInt32(pid);
         remote()->transact(ACQUIRE_AUDIO_SESSION_ID, data, &reply);
     }
 
-    virtual void releaseAudioSessionId(int audioSession)
+    virtual void releaseAudioSessionId(int audioSession, int pid)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
         data.writeInt32(audioSession);
+        data.writeInt32(pid);
         remote()->transact(RELEASE_AUDIO_SESSION_ID, data, &reply);
     }
 
@@ -769,6 +807,7 @@
             audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
             pid_t tid = (pid_t) data.readInt32();
             int sessionId = data.readInt32();
+            int clientUid = data.readInt32();
             String8 name;
             status_t status;
             sp<IAudioTrack> track;
@@ -779,9 +818,11 @@
             } else {
                 track = createTrack(
                         (audio_stream_type_t) streamType, sampleRate, format,
-                        channelMask, frameCount, &flags, buffer, output, tid,
-                        &sessionId, name, &status);
+                        channelMask, &frameCount, &flags, buffer, output, tid,
+                        &sessionId, name, clientUid, &status);
+                LOG_ALWAYS_FATAL_IF((track != 0) != (status == NO_ERROR));
             }
+            reply->writeInt32(frameCount);
             reply->writeInt32(flags);
             reply->writeInt32(sessionId);
             reply->writeString8(name);
@@ -801,7 +842,9 @@
             int sessionId = data.readInt32();
             status_t status;
             sp<IAudioRecord> record = openRecord(input,
-                    sampleRate, format, channelMask, frameCount, &flags, tid, &sessionId, &status);
+                    sampleRate, format, channelMask, &frameCount, &flags, tid, &sessionId, &status);
+            LOG_ALWAYS_FATAL_IF((record != 0) != (status == NO_ERROR));
+            reply->writeInt32(frameCount);
             reply->writeInt32(flags);
             reply->writeInt32(sessionId);
             reply->writeInt32(status);
@@ -944,7 +987,7 @@
                                                  &latency,
                                                  flags,
                                                  hasOffloadInfo ? &offloadInfo : NULL);
-            ALOGV("OPEN_OUTPUT output, %p", output);
+            ALOGV("OPEN_OUTPUT output, %d", output);
             reply->writeInt32((int32_t) output);
             reply->writeInt32(devices);
             reply->writeInt32(samplingRate);
@@ -1000,11 +1043,10 @@
             reply->writeInt32(closeInput((audio_io_handle_t) data.readInt32()));
             return NO_ERROR;
         } break;
-        case SET_STREAM_OUTPUT: {
+        case INVALIDATE_STREAM: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            uint32_t stream = data.readInt32();
-            audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
-            reply->writeInt32(setStreamOutput((audio_stream_type_t) stream, output));
+            audio_stream_type_t stream = (audio_stream_type_t) data.readInt32();
+            reply->writeInt32(invalidateStream(stream));
             return NO_ERROR;
         } break;
         case SET_VOICE_VOLUME: {
@@ -1016,8 +1058,8 @@
         case GET_RENDER_POSITION: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
-            size_t halFrames;
-            size_t dspFrames;
+            uint32_t halFrames;
+            uint32_t dspFrames;
             status_t status = getRenderPosition(&halFrames, &dspFrames, output);
             reply->writeInt32(status);
             if (status == NO_ERROR) {
@@ -1029,7 +1071,7 @@
         case GET_INPUT_FRAMES_LOST: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32();
-            reply->writeInt32(getInputFramesLost(ioHandle));
+            reply->writeInt32((int32_t) getInputFramesLost(ioHandle));
             return NO_ERROR;
         } break;
         case NEW_AUDIO_SESSION_ID: {
@@ -1040,13 +1082,15 @@
         case ACQUIRE_AUDIO_SESSION_ID: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             int audioSession = data.readInt32();
-            acquireAudioSessionId(audioSession);
+            int pid = data.readInt32();
+            acquireAudioSessionId(audioSession, pid);
             return NO_ERROR;
         } break;
         case RELEASE_AUDIO_SESSION_ID: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             int audioSession = data.readInt32();
-            releaseAudioSessionId(audioSession);
+            int pid = data.readInt32();
+            releaseAudioSessionId(audioSession, pid);
             return NO_ERROR;
         } break;
         case QUERY_NUM_EFFECTS: {

media/libmedia/IAudioPolicyService.cpp

diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index 4be3c09..1a027a6 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -476,10 +476,11 @@
         case START_OUTPUT: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
             audio_io_handle_t output = static_cast <audio_io_handle_t>(data.readInt32());
-            uint32_t stream = data.readInt32();
+            audio_stream_type_t stream =
+                                static_cast <audio_stream_type_t>(data.readInt32());
             int session = data.readInt32();
             reply->writeInt32(static_cast <uint32_t>(startOutput(output,
-                                                                 (audio_stream_type_t)stream,
+                                                                 stream,
                                                                  session)));
             return NO_ERROR;
         } break;
@@ -487,10 +488,11 @@
         case STOP_OUTPUT: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
             audio_io_handle_t output = static_cast <audio_io_handle_t>(data.readInt32());
-            uint32_t stream = data.readInt32();
+            audio_stream_type_t stream =
+                                static_cast <audio_stream_type_t>(data.readInt32());
             int session = data.readInt32();
             reply->writeInt32(static_cast <uint32_t>(stopOutput(output,
-                                                                (audio_stream_type_t)stream,
+                                                                stream,
                                                                 session)));
             return NO_ERROR;
         } break;
@@ -633,7 +635,7 @@
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
             audio_stream_type_t stream = (audio_stream_type_t) data.readInt32();
             uint32_t inPastMs = (uint32_t)data.readInt32();
-            reply->writeInt32( isStreamActive((audio_stream_type_t) stream, inPastMs) );
+            reply->writeInt32( isStreamActive(stream, inPastMs) );
             return NO_ERROR;
         } break;
 
@@ -641,7 +643,7 @@
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
             audio_stream_type_t stream = (audio_stream_type_t) data.readInt32();
             uint32_t inPastMs = (uint32_t)data.readInt32();
-            reply->writeInt32( isStreamActiveRemotely((audio_stream_type_t) stream, inPastMs) );
+            reply->writeInt32( isStreamActiveRemotely(stream, inPastMs) );
             return NO_ERROR;
         } break;
 

media/libmedia/IAudioRecord.cpp

diff --git a/media/libmedia/IAudioRecord.cpp b/media/libmedia/IAudioRecord.cpp
index 4a7de65..9866d70 100644
--- a/media/libmedia/IAudioRecord.cpp
+++ b/media/libmedia/IAudioRecord.cpp
@@ -50,6 +50,9 @@
         status_t status = remote()->transact(GET_CBLK, data, &reply);
         if (status == NO_ERROR) {
             cblk = interface_cast<IMemory>(reply.readStrongBinder());
+            if (cblk != 0 && cblk->pointer() == NULL) {
+                cblk.clear();
+            }
         }
         return cblk;
     }

media/libmedia/IAudioTrack.cpp

diff --git a/media/libmedia/IAudioTrack.cpp b/media/libmedia/IAudioTrack.cpp
index f0d75ba..ffc21fc 100644
--- a/media/libmedia/IAudioTrack.cpp
+++ b/media/libmedia/IAudioTrack.cpp
@@ -41,6 +41,7 @@
     SET_MEDIA_TIME_TRANSFORM,
     SET_PARAMETERS,
     GET_TIMESTAMP,
+    SIGNAL,
 };
 
 class BpAudioTrack : public BpInterface<IAudioTrack>
@@ -59,6 +60,9 @@
         status_t status = remote()->transact(GET_CBLK, data, &reply);
         if (status == NO_ERROR) {
             cblk = interface_cast<IMemory>(reply.readStrongBinder());
+            if (cblk != 0 && cblk->pointer() == NULL) {
+                cblk.clear();
+            }
         }
         return cblk;
     }
@@ -121,6 +125,9 @@
             status = reply.readInt32();
             if (status == NO_ERROR) {
                 *buffer = interface_cast<IMemory>(reply.readStrongBinder());
+                if (*buffer != 0 && (*buffer)->pointer() == NULL) {
+                    (*buffer).clear();
+                }
             }
         }
         return status;
@@ -182,6 +189,12 @@
         }
         return status;
     }
+
+    virtual void signal() {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+        remote()->transact(SIGNAL, data, &reply);
+    }
 };
 
 IMPLEMENT_META_INTERFACE(AudioTrack, "android.media.IAudioTrack");
@@ -269,6 +282,11 @@
             }
             return NO_ERROR;
         } break;
+        case SIGNAL: {
+            CHECK_INTERFACE(IAudioTrack, data, reply);
+            signal();
+            return NO_ERROR;
+        } break;
         default:
             return BBinder::onTransact(code, data, reply, flags);
     }

media/libmedia/IEffect.cpp

diff --git a/media/libmedia/IEffect.cpp b/media/libmedia/IEffect.cpp
index a303a8f..b94012a 100644
--- a/media/libmedia/IEffect.cpp
+++ b/media/libmedia/IEffect.cpp
@@ -117,6 +117,9 @@
         status_t status = remote()->transact(GET_CBLK, data, &reply);
         if (status == NO_ERROR) {
             cblk = interface_cast<IMemory>(reply.readStrongBinder());
+            if (cblk != 0 && cblk->pointer() == NULL) {
+                cblk.clear();
+            }
         }
         return cblk;
     }

media/libmedia/IMediaDeathNotifier.cpp

diff --git a/media/libmedia/IMediaDeathNotifier.cpp b/media/libmedia/IMediaDeathNotifier.cpp
index 9db5b1b..10b4934 100644
--- a/media/libmedia/IMediaDeathNotifier.cpp
+++ b/media/libmedia/IMediaDeathNotifier.cpp
@@ -75,7 +75,7 @@
 }
 
 void
-IMediaDeathNotifier::DeathNotifier::binderDied(const wp<IBinder>& who) {
+IMediaDeathNotifier::DeathNotifier::binderDied(const wp<IBinder>& who __unused) {
     ALOGW("media server died");
 
     // Need to do this with the lock held

media/libmedia/IMediaHTTPConnection.cpp

diff --git a/media/libmedia/IMediaHTTPConnection.cpp b/media/libmedia/IMediaHTTPConnection.cpp
new file mode 100644
index 0000000..7e26ee6
--- /dev/null
+++ b/media/libmedia/IMediaHTTPConnection.cpp
@@ -0,0 +1,182 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "IMediaHTTPConnection"
+#include <utils/Log.h>
+
+#include <media/IMediaHTTPConnection.h>
+
+#include <binder/IMemory.h>
+#include <binder/Parcel.h>
+#include <utils/String8.h>
+#include <media/stagefright/foundation/ADebug.h>
+
+namespace android {
+
+enum {
+    CONNECT = IBinder::FIRST_CALL_TRANSACTION,
+    DISCONNECT,
+    READ_AT,
+    GET_SIZE,
+    GET_MIME_TYPE,
+    GET_URI
+};
+
+struct BpMediaHTTPConnection : public BpInterface<IMediaHTTPConnection> {
+    BpMediaHTTPConnection(const sp<IBinder> &impl)
+        : BpInterface<IMediaHTTPConnection>(impl) {
+    }
+
+    virtual bool connect(
+            const char *uri, const KeyedVector<String8, String8> *headers) {
+        Parcel data, reply;
+        data.writeInterfaceToken(
+                IMediaHTTPConnection::getInterfaceDescriptor());
+
+        String16 tmp(uri);
+        data.writeString16(tmp);
+
+        tmp = String16("");
+        if (headers != NULL) {
+            for (size_t i = 0; i < headers->size(); ++i) {
+                String16 key(headers->keyAt(i).string());
+                String16 val(headers->valueAt(i).string());
+
+                tmp.append(key);
+                tmp.append(String16(": "));
+                tmp.append(val);
+                tmp.append(String16("\r\n"));
+            }
+        }
+        data.writeString16(tmp);
+
+        remote()->transact(CONNECT, data, &reply);
+
+        int32_t exceptionCode = reply.readExceptionCode();
+
+        if (exceptionCode) {
+            return UNKNOWN_ERROR;
+        }
+
+        sp<IBinder> binder = reply.readStrongBinder();
+        mMemory = interface_cast<IMemory>(binder);
+
+        return mMemory != NULL;
+    }
+
+    virtual void disconnect() {
+        Parcel data, reply;
+        data.writeInterfaceToken(
+                IMediaHTTPConnection::getInterfaceDescriptor());
+
+        remote()->transact(DISCONNECT, data, &reply);
+    }
+
+    virtual ssize_t readAt(off64_t offset, void *buffer, size_t size) {
+        Parcel data, reply;
+        data.writeInterfaceToken(
+                IMediaHTTPConnection::getInterfaceDescriptor());
+
+        data.writeInt64(offset);
+        data.writeInt32(size);
+
+        status_t err = remote()->transact(READ_AT, data, &reply);
+        if (err != OK) {
+            ALOGE("remote readAt failed");
+            return UNKNOWN_ERROR;
+        }
+
+        int32_t exceptionCode = reply.readExceptionCode();
+
+        if (exceptionCode) {
+            return UNKNOWN_ERROR;
+        }
+
+        int32_t len = reply.readInt32();
+
+        if (len > 0) {
+            memcpy(buffer, mMemory->pointer(), len);
+        }
+
+        return len;
+    }
+
+    virtual off64_t getSize() {
+        Parcel data, reply;
+        data.writeInterfaceToken(
+                IMediaHTTPConnection::getInterfaceDescriptor());
+
+        remote()->transact(GET_SIZE, data, &reply);
+
+        int32_t exceptionCode = reply.readExceptionCode();
+
+        if (exceptionCode) {
+            return UNKNOWN_ERROR;
+        }
+
+        return reply.readInt64();
+    }
+
+    virtual status_t getMIMEType(String8 *mimeType) {
+        *mimeType = String8("");
+
+        Parcel data, reply;
+        data.writeInterfaceToken(
+                IMediaHTTPConnection::getInterfaceDescriptor());
+
+        remote()->transact(GET_MIME_TYPE, data, &reply);
+
+        int32_t exceptionCode = reply.readExceptionCode();
+
+        if (exceptionCode) {
+            return UNKNOWN_ERROR;
+        }
+
+        *mimeType = String8(reply.readString16());
+
+        return OK;
+    }
+
+    virtual status_t getUri(String8 *uri) {
+        *uri = String8("");
+
+        Parcel data, reply;
+        data.writeInterfaceToken(
+                IMediaHTTPConnection::getInterfaceDescriptor());
+
+        remote()->transact(GET_URI, data, &reply);
+
+        int32_t exceptionCode = reply.readExceptionCode();
+
+        if (exceptionCode) {
+            return UNKNOWN_ERROR;
+        }
+
+        *uri = String8(reply.readString16());
+
+        return OK;
+    }
+
+private:
+    sp<IMemory> mMemory;
+};
+
+IMPLEMENT_META_INTERFACE(
+        MediaHTTPConnection, "android.media.IMediaHTTPConnection");
+
+}  // namespace android
+

media/libmedia/IMediaHTTPService.cpp

diff --git a/media/libmedia/IMediaHTTPService.cpp b/media/libmedia/IMediaHTTPService.cpp
new file mode 100644
index 0000000..1260582
--- /dev/null
+++ b/media/libmedia/IMediaHTTPService.cpp
@@ -0,0 +1,58 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "IMediaHTTPService"
+#include <utils/Log.h>
+
+#include <media/IMediaHTTPService.h>
+
+#include <binder/Parcel.h>
+#include <media/IMediaHTTPConnection.h>
+
+namespace android {
+
+enum {
+    MAKE_HTTP = IBinder::FIRST_CALL_TRANSACTION,
+};
+
+struct BpMediaHTTPService : public BpInterface<IMediaHTTPService> {
+    BpMediaHTTPService(const sp<IBinder> &impl)
+        : BpInterface<IMediaHTTPService>(impl) {
+    }
+
+    virtual sp<IMediaHTTPConnection> makeHTTPConnection() {
+        Parcel data, reply;
+        data.writeInterfaceToken(
+                IMediaHTTPService::getInterfaceDescriptor());
+
+        remote()->transact(MAKE_HTTP, data, &reply);
+
+        status_t err = reply.readInt32();
+
+        if (err != OK) {
+            return NULL;
+        }
+
+        return interface_cast<IMediaHTTPConnection>(reply.readStrongBinder());
+    }
+};
+
+IMPLEMENT_META_INTERFACE(
+        MediaHTTPService, "android.media.IMediaHTTPService");
+
+}  // namespace android
+

media/libmedia/IMediaMetadataRetriever.cpp

diff --git a/media/libmedia/IMediaMetadataRetriever.cpp b/media/libmedia/IMediaMetadataRetriever.cpp
index bb066a0..c7d9d51 100644
--- a/media/libmedia/IMediaMetadataRetriever.cpp
+++ b/media/libmedia/IMediaMetadataRetriever.cpp
@@ -18,6 +18,7 @@
 #include <stdint.h>
 #include <sys/types.h>
 #include <binder/Parcel.h>
+#include <media/IMediaHTTPService.h>
 #include <media/IMediaMetadataRetriever.h>
 #include <utils/String8.h>
 #include <utils/KeyedVector.h>
@@ -84,10 +85,16 @@
     }
 
     status_t setDataSource(
-            const char *srcUrl, const KeyedVector<String8, String8> *headers)
+            const sp<IMediaHTTPService> &httpService,
+            const char *srcUrl,
+            const KeyedVector<String8, String8> *headers)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IMediaMetadataRetriever::getInterfaceDescriptor());
+        data.writeInt32(httpService != NULL);
+        if (httpService != NULL) {
+            data.writeStrongBinder(httpService->asBinder());
+        }
         data.writeCString(srcUrl);
 
         if (headers == NULL) {
@@ -195,6 +202,13 @@
         } break;
         case SET_DATA_SOURCE_URL: {
             CHECK_INTERFACE(IMediaMetadataRetriever, data, reply);
+
+            sp<IMediaHTTPService> httpService;
+            if (data.readInt32()) {
+                httpService =
+                    interface_cast<IMediaHTTPService>(data.readStrongBinder());
+            }
+
             const char* srcUrl = data.readCString();
 
             KeyedVector<String8, String8> headers;
@@ -206,7 +220,8 @@
             }
 
             reply->writeInt32(
-                    setDataSource(srcUrl, numHeaders > 0 ? &headers : NULL));
+                    setDataSource(
+                        httpService, srcUrl, numHeaders > 0 ? &headers : NULL));
 
             return NO_ERROR;
         } break;

media/libmedia/IMediaPlayer.cpp

diff --git a/media/libmedia/IMediaPlayer.cpp b/media/libmedia/IMediaPlayer.cpp
index e79bcd2..d778d05 100644
--- a/media/libmedia/IMediaPlayer.cpp
+++ b/media/libmedia/IMediaPlayer.cpp
@@ -21,6 +21,7 @@
 
 #include <binder/Parcel.h>
 
+#include <media/IMediaHTTPService.h>
 #include <media/IMediaPlayer.h>
 #include <media/IStreamSource.h>
 
@@ -75,11 +76,17 @@
         remote()->transact(DISCONNECT, data, &reply);
     }
 
-    status_t setDataSource(const char* url,
+    status_t setDataSource(
+            const sp<IMediaHTTPService> &httpService,
+            const char* url,
             const KeyedVector<String8, String8>* headers)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IMediaPlayer::getInterfaceDescriptor());
+        data.writeInt32(httpService != NULL);
+        if (httpService != NULL) {
+            data.writeStrongBinder(httpService->asBinder());
+        }
         data.writeCString(url);
         if (headers == NULL) {
             data.writeInt32(0);
@@ -355,6 +362,13 @@
         } break;
         case SET_DATA_SOURCE_URL: {
             CHECK_INTERFACE(IMediaPlayer, data, reply);
+
+            sp<IMediaHTTPService> httpService;
+            if (data.readInt32()) {
+                httpService =
+                    interface_cast<IMediaHTTPService>(data.readStrongBinder());
+            }
+
             const char* url = data.readCString();
             KeyedVector<String8, String8> headers;
             int32_t numHeaders = data.readInt32();
@@ -363,7 +377,8 @@
                 String8 value = data.readString8();
                 headers.add(key, value);
             }
-            reply->writeInt32(setDataSource(url, numHeaders > 0 ? &headers : NULL));
+            reply->writeInt32(setDataSource(
+                        httpService, url, numHeaders > 0 ? &headers : NULL));
             return NO_ERROR;
         } break;
         case SET_DATA_SOURCE_FD: {

media/libmedia/IMediaPlayerService.cpp

diff --git a/media/libmedia/IMediaPlayerService.cpp b/media/libmedia/IMediaPlayerService.cpp
index 3c22b4c..d116b14 100644
--- a/media/libmedia/IMediaPlayerService.cpp
+++ b/media/libmedia/IMediaPlayerService.cpp
@@ -23,6 +23,7 @@
 #include <media/ICrypto.h>
 #include <media/IDrm.h>
 #include <media/IHDCP.h>
+#include <media/IMediaHTTPService.h>
 #include <media/IMediaPlayerService.h>
 #include <media/IMediaRecorder.h>
 #include <media/IOMX.h>
@@ -48,7 +49,6 @@
     ADD_BATTERY_DATA,
     PULL_BATTERY_DATA,
     LISTEN_FOR_REMOTE_DISPLAY,
-    UPDATE_PROXY_CONFIG,
 };
 
 class BpMediaPlayerService: public BpInterface<IMediaPlayerService>
@@ -86,12 +86,21 @@
         return interface_cast<IMediaRecorder>(reply.readStrongBinder());
     }
 
-    virtual status_t decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
-                               audio_format_t* pFormat,
-                               const sp<IMemoryHeap>& heap, size_t *pSize)
+    virtual status_t decode(
+            const sp<IMediaHTTPService> &httpService,
+            const char* url,
+            uint32_t *pSampleRate,
+            int* pNumChannels,
+            audio_format_t* pFormat,
+            const sp<IMemoryHeap>& heap,
+            size_t *pSize)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
+        data.writeInt32(httpService != NULL);
+        if (httpService != NULL) {
+            data.writeStrongBinder(httpService->asBinder());
+        }
         data.writeCString(url);
         data.writeStrongBinder(heap->asBinder());
         status_t status = remote()->transact(DECODE_URL, data, &reply);
@@ -182,25 +191,6 @@
         remote()->transact(LISTEN_FOR_REMOTE_DISPLAY, data, &reply);
         return interface_cast<IRemoteDisplay>(reply.readStrongBinder());
     }
-
-    virtual status_t updateProxyConfig(
-            const char *host, int32_t port, const char *exclusionList) {
-        Parcel data, reply;
-
-        data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
-        if (host == NULL) {
-            data.writeInt32(0);
-        } else {
-            data.writeInt32(1);
-            data.writeCString(host);
-            data.writeInt32(port);
-            data.writeCString(exclusionList);
-        }
-
-        remote()->transact(UPDATE_PROXY_CONFIG, data, &reply);
-
-        return reply.readInt32();
-    }
 };
 
 IMPLEMENT_META_INTERFACE(MediaPlayerService, "android.media.IMediaPlayerService");
@@ -222,13 +212,25 @@
         } break;
         case DECODE_URL: {
             CHECK_INTERFACE(IMediaPlayerService, data, reply);
+            sp<IMediaHTTPService> httpService;
+            if (data.readInt32()) {
+                httpService =
+                    interface_cast<IMediaHTTPService>(data.readStrongBinder());
+            }
             const char* url = data.readCString();
             sp<IMemoryHeap> heap = interface_cast<IMemoryHeap>(data.readStrongBinder());
             uint32_t sampleRate;
             int numChannels;
             audio_format_t format;
             size_t size;
-            status_t status = decode(url, &sampleRate, &numChannels, &format, heap, &size);
+            status_t status =
+                decode(httpService,
+                       url,
+                       &sampleRate,
+                       &numChannels,
+                       &format,
+                       heap,
+                       &size);
             reply->writeInt32(status);
             if (status == NO_ERROR) {
                 reply->writeInt32(sampleRate);
@@ -316,24 +318,6 @@
             reply->writeStrongBinder(display->asBinder());
             return NO_ERROR;
         } break;
-        case UPDATE_PROXY_CONFIG:
-        {
-            CHECK_INTERFACE(IMediaPlayerService, data, reply);
-
-            const char *host = NULL;
-            int32_t port = 0;
-            const char *exclusionList = NULL;
-
-            if (data.readInt32()) {
-                host = data.readCString();
-                port = data.readInt32();
-                exclusionList = data.readCString();
-            }
-
-            reply->writeInt32(updateProxyConfig(host, port, exclusionList));
-
-            return OK;
-        }
         default:
             return BBinder::onTransact(code, data, reply, flags);
     }

media/libmedia/IOMX.cpp

diff --git a/media/libmedia/IOMX.cpp b/media/libmedia/IOMX.cpp
index ef99f4f..71ce320 100644
--- a/media/libmedia/IOMX.cpp
+++ b/media/libmedia/IOMX.cpp
@@ -43,6 +43,7 @@
     CREATE_INPUT_SURFACE,
     SIGNAL_END_OF_INPUT_STREAM,
     STORE_META_DATA_IN_BUFFERS,
+    PREPARE_FOR_ADAPTIVE_PLAYBACK,
     ALLOC_BUFFER,
     ALLOC_BUFFER_WITH_BACKUP,
     FREE_BUFFER,
@@ -351,6 +352,22 @@
         return err;
     }
 
+    virtual status_t prepareForAdaptivePlayback(
+            node_id node, OMX_U32 port_index, OMX_BOOL enable,
+            OMX_U32 max_width, OMX_U32 max_height) {
+        Parcel data, reply;
+        data.writeInterfaceToken(IOMX::getInterfaceDescriptor());
+        data.writeIntPtr((intptr_t)node);
+        data.writeInt32(port_index);
+        data.writeInt32((int32_t)enable);
+        data.writeInt32(max_width);
+        data.writeInt32(max_height);
+        remote()->transact(PREPARE_FOR_ADAPTIVE_PLAYBACK, data, &reply);
+
+        status_t err = reply.readInt32();
+        return err;
+    }
+
     virtual status_t allocateBuffer(
             node_id node, OMX_U32 port_index, size_t size,
             buffer_id *buffer, void **buffer_data) {
@@ -770,6 +787,23 @@
             return NO_ERROR;
         }
 
+        case PREPARE_FOR_ADAPTIVE_PLAYBACK:
+        {
+            CHECK_OMX_INTERFACE(IOMX, data, reply);
+
+            node_id node = (void*)data.readIntPtr();
+            OMX_U32 port_index = data.readInt32();
+            OMX_BOOL enable = (OMX_BOOL)data.readInt32();
+            OMX_U32 max_width = data.readInt32();
+            OMX_U32 max_height = data.readInt32();
+
+            status_t err = prepareForAdaptivePlayback(
+                    node, port_index, enable, max_width, max_height);
+            reply->writeInt32(err);
+
+            return NO_ERROR;
+        }
+
         case ALLOC_BUFFER:
         {
             CHECK_OMX_INTERFACE(IOMX, data, reply);

media/libmedia/JetPlayer.cpp

diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp
index e914b34..f0f1832 100644
--- a/media/libmedia/JetPlayer.cpp
+++ b/media/libmedia/JetPlayer.cpp
@@ -90,7 +90,7 @@
             pLibConfig->sampleRate,
             AUDIO_FORMAT_PCM_16_BIT,
             audio_channel_out_mask_from_count(pLibConfig->numChannels),
-            mTrackBufferSize,
+            (size_t) mTrackBufferSize,
             AUDIO_OUTPUT_FLAG_NONE);
 
     // create render and playback thread

media/libmedia/MediaProfiles.cpp

diff --git a/media/libmedia/MediaProfiles.cpp b/media/libmedia/MediaProfiles.cpp
index 8319cd7..1074da9 100644
--- a/media/libmedia/MediaProfiles.cpp
+++ b/media/libmedia/MediaProfiles.cpp
@@ -81,8 +81,14 @@
     {"timelapseqvga", CAMCORDER_QUALITY_TIME_LAPSE_QVGA},
 };
 
+#if LOG_NDEBUG
+#define UNUSED __unused
+#else
+#define UNUSED
+#endif
+
 /*static*/ void
-MediaProfiles::logVideoCodec(const MediaProfiles::VideoCodec& codec)
+MediaProfiles::logVideoCodec(const MediaProfiles::VideoCodec& codec UNUSED)
 {
     ALOGV("video codec:");
     ALOGV("codec = %d", codec.mCodec);
@@ -93,7 +99,7 @@
 }
 
 /*static*/ void
-MediaProfiles::logAudioCodec(const MediaProfiles::AudioCodec& codec)
+MediaProfiles::logAudioCodec(const MediaProfiles::AudioCodec& codec UNUSED)
 {
     ALOGV("audio codec:");
     ALOGV("codec = %d", codec.mCodec);
@@ -103,7 +109,7 @@
 }
 
 /*static*/ void
-MediaProfiles::logVideoEncoderCap(const MediaProfiles::VideoEncoderCap& cap)
+MediaProfiles::logVideoEncoderCap(const MediaProfiles::VideoEncoderCap& cap UNUSED)
 {
     ALOGV("video encoder cap:");
     ALOGV("codec = %d", cap.mCodec);
@@ -114,7 +120,7 @@
 }
 
 /*static*/ void
-MediaProfiles::logAudioEncoderCap(const MediaProfiles::AudioEncoderCap& cap)
+MediaProfiles::logAudioEncoderCap(const MediaProfiles::AudioEncoderCap& cap UNUSED)
 {
     ALOGV("audio encoder cap:");
     ALOGV("codec = %d", cap.mCodec);
@@ -124,21 +130,21 @@
 }
 
 /*static*/ void
-MediaProfiles::logVideoDecoderCap(const MediaProfiles::VideoDecoderCap& cap)
+MediaProfiles::logVideoDecoderCap(const MediaProfiles::VideoDecoderCap& cap UNUSED)
 {
     ALOGV("video decoder cap:");
     ALOGV("codec = %d", cap.mCodec);
 }
 
 /*static*/ void
-MediaProfiles::logAudioDecoderCap(const MediaProfiles::AudioDecoderCap& cap)
+MediaProfiles::logAudioDecoderCap(const MediaProfiles::AudioDecoderCap& cap UNUSED)
 {
     ALOGV("audio codec cap:");
     ALOGV("codec = %d", cap.mCodec);
 }
 
 /*static*/ void
-MediaProfiles::logVideoEditorCap(const MediaProfiles::VideoEditorCap& cap)
+MediaProfiles::logVideoEditorCap(const MediaProfiles::VideoEditorCap& cap UNUSED)
 {
     ALOGV("videoeditor cap:");
     ALOGV("mMaxInputFrameWidth = %d", cap.mMaxInputFrameWidth);

media/libmedia/MediaScannerClient.cpp

diff --git a/media/libmedia/MediaScannerClient.cpp b/media/libmedia/MediaScannerClient.cpp
index 93a4a4c..1661f04 100644
--- a/media/libmedia/MediaScannerClient.cpp
+++ b/media/libmedia/MediaScannerClient.cpp
@@ -14,217 +14,57 @@
  * limitations under the License.
  */
 
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MediaScannerClient"
+#include <utils/Log.h>
+
 #include <media/mediascanner.h>
 
+#include "CharacterEncodingDetector.h"
 #include "StringArray.h"
 
-#include "autodetect.h"
-#include "unicode/ucnv.h"
-#include "unicode/ustring.h"
-
 namespace android {
 
 MediaScannerClient::MediaScannerClient()
-    :   mNames(NULL),
-        mValues(NULL),
-        mLocaleEncoding(kEncodingNone)
+    :   mEncodingDetector(NULL)
 {
 }
 
 MediaScannerClient::~MediaScannerClient()
 {
-    delete mNames;
-    delete mValues;
+    delete mEncodingDetector;
 }
 
 void MediaScannerClient::setLocale(const char* locale)
 {
-    if (!locale) return;
-
-    if (!strncmp(locale, "ja", 2))
-        mLocaleEncoding = kEncodingShiftJIS;
-    else if (!strncmp(locale, "ko", 2))
-        mLocaleEncoding = kEncodingEUCKR;
-    else if (!strncmp(locale, "zh", 2)) {
-        if (!strcmp(locale, "zh_CN")) {
-            // simplified chinese for mainland China
-            mLocaleEncoding = kEncodingGBK;
-        } else {
-            // assume traditional for non-mainland Chinese locales (Taiwan, Hong Kong, Singapore)
-            mLocaleEncoding = kEncodingBig5;
-        }
-    }
+    mLocale = locale; // not currently used
 }
 
 void MediaScannerClient::beginFile()
 {
-    mNames = new StringArray;
-    mValues = new StringArray;
+    delete mEncodingDetector;
+    mEncodingDetector = new CharacterEncodingDetector();
 }
 
 status_t MediaScannerClient::addStringTag(const char* name, const char* value)
 {
-    if (mLocaleEncoding != kEncodingNone) {
-        // don't bother caching strings that are all ASCII.
-        // call handleStringTag directly instead.
-        // check to see if value (which should be utf8) has any non-ASCII characters
-        bool nonAscii = false;
-        const char* chp = value;
-        char ch;
-        while ((ch = *chp++)) {
-            if (ch & 0x80) {
-                nonAscii = true;
-                break;
-            }
-        }
-
-        if (nonAscii) {
-            // save the strings for later so they can be used for native encoding detection
-            mNames->push_back(name);
-            mValues->push_back(value);
-            return OK;
-        }
-        // else fall through
-    }
-
-    // autodetection is not necessary, so no need to cache the values
-    // pass directly to the client instead
-    return handleStringTag(name, value);
-}
-
-static uint32_t possibleEncodings(const char* s)
-{
-    uint32_t result = kEncodingAll;
-    // if s contains a native encoding, then it was mistakenly encoded in utf8 as if it were latin-1
-    // so we need to reverse the latin-1 -> utf8 conversion to get the native chars back
-    uint8_t ch1, ch2;
-    uint8_t* chp = (uint8_t *)s;
-
-    while ((ch1 = *chp++)) {
-        if (ch1 & 0x80) {
-            ch2 = *chp++;
-            ch1 = ((ch1 << 6) & 0xC0) | (ch2 & 0x3F);
-            // ch1 is now the first byte of the potential native char
-
-            ch2 = *chp++;
-            if (ch2 & 0x80)
-                ch2 = ((ch2 << 6) & 0xC0) | (*chp++ & 0x3F);
-            // ch2 is now the second byte of the potential native char
-            int ch = (int)ch1 << 8 | (int)ch2;
-            result &= findPossibleEncodings(ch);
-        }
-        // else ASCII character, which could be anything
-    }
-
-    return result;
-}
-
-void MediaScannerClient::convertValues(uint32_t encoding)
-{
-    const char* enc = NULL;
-    switch (encoding) {
-        case kEncodingShiftJIS:
-            enc = "shift-jis";
-            break;
-        case kEncodingGBK:
-            enc = "gbk";
-            break;
-        case kEncodingBig5:
-            enc = "Big5";
-            break;
-        case kEncodingEUCKR:
-            enc = "EUC-KR";
-            break;
-    }
-
-    if (enc) {
-        UErrorCode status = U_ZERO_ERROR;
-
-        UConverter *conv = ucnv_open(enc, &status);
-        if (U_FAILURE(status)) {
-            ALOGE("could not create UConverter for %s", enc);
-            return;
-        }
-        UConverter *utf8Conv = ucnv_open("UTF-8", &status);
-        if (U_FAILURE(status)) {
-            ALOGE("could not create UConverter for UTF-8");
-            ucnv_close(conv);
-            return;
-        }
-
-        // for each value string, convert from native encoding to UTF-8
-        for (int i = 0; i < mNames->size(); i++) {
-            // first we need to untangle the utf8 and convert it back to the original bytes
-            // since we are reducing the length of the string, we can do this in place
-            uint8_t* src = (uint8_t *)mValues->getEntry(i);
-            int len = strlen((char *)src);
-            uint8_t* dest = src;
-
-            uint8_t uch;
-            while ((uch = *src++)) {
-                if (uch & 0x80)
-                    *dest++ = ((uch << 6) & 0xC0) | (*src++ & 0x3F);
-                else
-                    *dest++ = uch;
-            }
-            *dest = 0;
-
-            // now convert from native encoding to UTF-8
-            const char* source = mValues->getEntry(i);
-            int targetLength = len * 3 + 1;
-            char* buffer = new char[targetLength];
-            // don't normally check for NULL, but in this case targetLength may be large
-            if (!buffer)
-                break;
-            char* target = buffer;
-
-            ucnv_convertEx(utf8Conv, conv, &target, target + targetLength,
-                    &source, (const char *)dest, NULL, NULL, NULL, NULL, TRUE, TRUE, &status);
-            if (U_FAILURE(status)) {
-                ALOGE("ucnv_convertEx failed: %d", status);
-                mValues->setEntry(i, "???");
-            } else {
-                // zero terminate
-                *target = 0;
-                mValues->setEntry(i, buffer);
-            }
-
-            delete[] buffer;
-        }
-
-        ucnv_close(conv);
-        ucnv_close(utf8Conv);
-    }
+    mEncodingDetector->addTag(name, value);
+    return OK;
 }
 
 void MediaScannerClient::endFile()
 {
-    if (mLocaleEncoding != kEncodingNone) {
-        int size = mNames->size();
-        uint32_t encoding = kEncodingAll;
+    mEncodingDetector->detectAndConvert();
 
-        // compute a bit mask containing all possible encodings
-        for (int i = 0; i < mNames->size(); i++)
-            encoding &= possibleEncodings(mValues->getEntry(i));
-
-        // if the locale encoding matches, then assume we have a native encoding.
-        if (encoding & mLocaleEncoding)
-            convertValues(mLocaleEncoding);
-
-        // finally, push all name/value pairs to the client
-        for (int i = 0; i < mNames->size(); i++) {
-            status_t status = handleStringTag(mNames->getEntry(i), mValues->getEntry(i));
-            if (status) {
-                break;
-            }
+    int size = mEncodingDetector->size();
+    if (size) {
+        for (int i = 0; i < size; i++) {
+            const char *name;
+            const char *value;
+            mEncodingDetector->getTag(i, &name, &value);
+            handleStringTag(name, value);
         }
     }
-    // else addStringTag() has done all the work so we have nothing to do
-
-    delete mNames;
-    delete mValues;
-    mNames = NULL;
-    mValues = NULL;
 }
 
 }  // namespace android

media/libmedia/MemoryLeakTrackUtil.cpp

diff --git a/media/libmedia/MemoryLeakTrackUtil.cpp b/media/libmedia/MemoryLeakTrackUtil.cpp
index 6a108ae..f004ca4 100644
--- a/media/libmedia/MemoryLeakTrackUtil.cpp
+++ b/media/libmedia/MemoryLeakTrackUtil.cpp
@@ -49,7 +49,7 @@
     }
 
     void append(const char *s) {
-        strcat(mPtr, s);
+        strncat(mPtr, s, MAX_SIZE - size() - 1);
     }
 
     const char *string() const {
@@ -60,6 +60,10 @@
         return strlen(mPtr);
     }
 
+    void clear() {
+        *mPtr = '\0';
+    }
+
 private:
     char *mPtr;
 
@@ -139,6 +143,9 @@
             }
         } while (moved);
 
+        write(fd, result.string(), result.size());
+        result.clear();
+
         for (size_t i = 0; i < count; i++) {
             AllocEntry *e = &entries[i];
 
@@ -152,13 +159,14 @@
                 result.append(buffer);
             }
             result.append("\n");
+
+            write(fd, result.string(), result.size());
+            result.clear();
         }
 
         delete[] entries;
         free_malloc_leak_info(info);
     }
-
-    write(fd, result.string(), result.size());
 }
 
 #else

media/libmedia/SoundPool.cpp

diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp
index 8434d43..a55e09c 100644
--- a/media/libmedia/SoundPool.cpp
+++ b/media/libmedia/SoundPool.cpp
@@ -21,6 +21,7 @@
 #define USE_SHARED_MEM_BUFFER
 
 #include <media/AudioTrack.h>
+#include <media/IMediaHTTPService.h>
 #include <media/mediaplayer.h>
 #include <media/SoundPool.h>
 #include "SoundPoolThread.h"
@@ -199,7 +200,7 @@
     return NULL;
 }
 
-int SoundPool::load(const char* path, int priority)
+int SoundPool::load(const char* path, int priority __unused)
 {
     ALOGV("load: path=%s, priority=%d", path, priority);
     Mutex::Autolock lock(&mLock);
@@ -209,7 +210,7 @@
     return sample->sampleID();
 }
 
-int SoundPool::load(int fd, int64_t offset, int64_t length, int priority)
+int SoundPool::load(int fd, int64_t offset, int64_t length, int priority __unused)
 {
     ALOGV("load: fd=%d, offset=%lld, length=%lld, priority=%d",
             fd, offset, length, priority);
@@ -496,7 +497,14 @@
 
     ALOGV("Start decode");
     if (mUrl) {
-        status = MediaPlayer::decode(mUrl, &sampleRate, &numChannels, &format, mHeap, &mSize);
+        status = MediaPlayer::decode(
+                NULL /* httpService */,
+                mUrl,
+                &sampleRate,
+                &numChannels,
+                &format,
+                mHeap,
+                &mSize);
     } else {
         status = MediaPlayer::decode(mFd, mOffset, mLength, &sampleRate, &numChannels, &format,
                                      mHeap, &mSize);
@@ -579,7 +587,7 @@
         uint32_t sampleRate = uint32_t(float(sample->sampleRate()) * rate + 0.5);
         uint32_t totalFrames = (kDefaultBufferCount * afFrameCount * sampleRate) / afSampleRate;
         uint32_t bufferFrames = (totalFrames + (kDefaultBufferCount - 1)) / kDefaultBufferCount;
-        uint32_t frameCount = 0;
+        size_t frameCount = 0;
 
         if (loop) {
             frameCount = sample->size()/numChannels/
@@ -600,16 +608,15 @@
         // wrong audio audio buffer size  (mAudioBufferSize)
         unsigned long toggle = mToggle ^ 1;
         void *userData = (void *)((unsigned long)this | toggle);
-        uint32_t channels = (numChannels == 2) ?
-                AUDIO_CHANNEL_OUT_STEREO : AUDIO_CHANNEL_OUT_MONO;
+        audio_channel_mask_t channelMask = audio_channel_out_mask_from_count(numChannels);
 
         // do not create a new audio track if current track is compatible with sample parameters
 #ifdef USE_SHARED_MEM_BUFFER
         newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
-                channels, sample->getIMemory(), AUDIO_OUTPUT_FLAG_FAST, callback, userData);
+                channelMask, sample->getIMemory(), AUDIO_OUTPUT_FLAG_FAST, callback, userData);
 #else
         newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
-                channels, frameCount, AUDIO_OUTPUT_FLAG_FAST, callback, userData,
+                channelMask, frameCount, AUDIO_OUTPUT_FLAG_FAST, callback, userData,
                 bufferFrames);
 #endif
         oldTrack = mAudioTrack;
@@ -730,7 +737,8 @@
                     count = b->size;
                 }
                 memcpy(q, p, count);
-//              ALOGV("fill: q=%p, p=%p, mPos=%u, b->size=%u, count=%d", q, p, mPos, b->size, count);
+//              ALOGV("fill: q=%p, p=%p, mPos=%u, b->size=%u, count=%d", q, p, mPos, b->size,
+//                      count);
             } else if (mPos < mAudioBufferSize) {
                 count = mAudioBufferSize - mPos;
                 if (count > b->size) {
@@ -744,11 +752,16 @@
             b->size = count;
             //ALOGV("buffer=%p, [0]=%d", b->i16, b->i16[0]);
         }
-    } else if (event == AudioTrack::EVENT_UNDERRUN || event == AudioTrack::EVENT_BUFFER_END) {
-        ALOGV("process %p channel %d EVENT_UNDERRUN or EVENT_BUFFER_END", this, mChannelID);
+    } else if (event == AudioTrack::EVENT_UNDERRUN || event == AudioTrack::EVENT_BUFFER_END ||
+            event == AudioTrack::EVENT_NEW_IAUDIOTRACK) {
+        ALOGV("process %p channel %d event %s",
+              this, mChannelID, (event == AudioTrack::EVENT_UNDERRUN) ? "UNDERRUN" :
+                      (event == AudioTrack::EVENT_BUFFER_END) ? "BUFFER_END" : "NEW_IAUDIOTRACK");
         mSoundPool->addToStopList(this);
     } else if (event == AudioTrack::EVENT_LOOP_END) {
         ALOGV("End loop %p channel %d", this, mChannelID);
+    } else {
+        ALOGW("SoundChannel::process unexpected event %d", event);
     }
 }
 

media/libmedia/autodetect.cpp

diff --git a/media/libmedia/autodetect.cpp b/media/libmedia/autodetect.cpp
deleted file mode 100644
index be5c3b2..0000000
--- a/media/libmedia/autodetect.cpp
+++ /dev/null
@@ -1,885 +0,0 @@
-/*
- * Copyright (C) 2008 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * 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 "autodetect.h"
-
-struct CharRange {
-    uint16_t first;
-    uint16_t last;
-};
-
-#define ARRAY_SIZE(x)   (sizeof(x) / sizeof(*x))
-
-// generated from http://unicode.org/Public/MAPPINGS/VENDORS/MICSFT/WINDOWS/CP932.TXT
-static const CharRange kShiftJISRanges[] = {
-    { 0x8140, 0x817E },
-    { 0x8180, 0x81AC },
-    { 0x81B8, 0x81BF },
-    { 0x81C8, 0x81CE },
-    { 0x81DA, 0x81E8 },
-    { 0x81F0, 0x81F7 },
-    { 0x81FC, 0x81FC },
-    { 0x824F, 0x8258 },
-    { 0x8260, 0x8279 },
-    { 0x8281, 0x829A },
-    { 0x829F, 0x82F1 },
-    { 0x8340, 0x837E },
-    { 0x8380, 0x8396 },
-    { 0x839F, 0x83B6 },
-    { 0x83BF, 0x83D6 },
-    { 0x8440, 0x8460 },
-    { 0x8470, 0x847E },
-    { 0x8480, 0x8491 },
-    { 0x849F, 0x84BE },
-    { 0x8740, 0x875D },
-    { 0x875F, 0x8775 },
-    { 0x877E, 0x877E },
-    { 0x8780, 0x879C },
-    { 0x889F, 0x88FC },
-    { 0x8940, 0x897E },
-    { 0x8980, 0x89FC },
-    { 0x8A40, 0x8A7E },
-    { 0x8A80, 0x8AFC },
-    { 0x8B40, 0x8B7E },
-    { 0x8B80, 0x8BFC },
-    { 0x8C40, 0x8C7E },
-    { 0x8C80, 0x8CFC },
-    { 0x8D40, 0x8D7E },
-    { 0x8D80, 0x8DFC },
-    { 0x8E40, 0x8E7E },
-    { 0x8E80, 0x8EFC },
-    { 0x8F40, 0x8F7E },
-    { 0x8F80, 0x8FFC },
-    { 0x9040, 0x907E },
-    { 0x9080, 0x90FC },
-    { 0x9140, 0x917E },
-    { 0x9180, 0x91FC },
-    { 0x9240, 0x927E },
-    { 0x9280, 0x92FC },
-    { 0x9340, 0x937E },
-    { 0x9380, 0x93FC },
-    { 0x9440, 0x947E },
-    { 0x9480, 0x94FC },
-    { 0x9540, 0x957E },
-    { 0x9580, 0x95FC },
-    { 0x9640, 0x967E },
-    { 0x9680, 0x96FC },
-    { 0x9740, 0x977E },
-    { 0x9780, 0x97FC },
-    { 0x9840, 0x9872 },
-    { 0x989F, 0x98FC },
-    { 0x9940, 0x997E },
-    { 0x9980, 0x99FC },
-    { 0x9A40, 0x9A7E },
-    { 0x9A80, 0x9AFC },
-    { 0x9B40, 0x9B7E },
-    { 0x9B80, 0x9BFC },
-    { 0x9C40, 0x9C7E },
-    { 0x9C80, 0x9CFC },
-    { 0x9D40, 0x9D7E },
-    { 0x9D80, 0x9DFC },
-    { 0x9E40, 0x9E7E },
-    { 0x9E80, 0x9EFC },
-    { 0x9F40, 0x9F7E },
-    { 0x9F80, 0x9FFC },
-    { 0xE040, 0xE07E },
-    { 0xE080, 0xE0FC },
-    { 0xE140, 0xE17E },
-    { 0xE180, 0xE1FC },
-    { 0xE240, 0xE27E },
-    { 0xE280, 0xE2FC },
-    { 0xE340, 0xE37E },
-    { 0xE380, 0xE3FC },
-    { 0xE440, 0xE47E },
-    { 0xE480, 0xE4FC },
-    { 0xE540, 0xE57E },
-    { 0xE580, 0xE5FC },
-    { 0xE640, 0xE67E },
-    { 0xE680, 0xE6FC },
-    { 0xE740, 0xE77E },
-    { 0xE780, 0xE7FC },
-    { 0xE840, 0xE87E },
-    { 0xE880, 0xE8FC },
-    { 0xE940, 0xE97E },
-    { 0xE980, 0xE9FC },
-    { 0xEA40, 0xEA7E },
-    { 0xEA80, 0xEAA4 },
-    { 0xED40, 0xED7E },
-    { 0xED80, 0xEDFC },
-    { 0xEE40, 0xEE7E },
-    { 0xEE80, 0xEEEC },
-    { 0xEEEF, 0xEEFC },
-    { 0xFA40, 0xFA7E },
-    { 0xFA80, 0xFAFC },
-    { 0xFB40, 0xFB7E },
-    { 0xFB80, 0xFBFC },
-    { 0xFC40, 0xFC4B },
-};
-
-// generated from http://unicode.org/Public/MAPPINGS/VENDORS/MICSFT/WINDOWS/CP936.TXT
-static const CharRange kGBKRanges[] = {
-    { 0x8140, 0x817E },
-    { 0x8180, 0x81FE },
-    { 0x8240, 0x827E },
-    { 0x8280, 0x82FE },
-    { 0x8340, 0x837E },
-    { 0x8380, 0x83FE },
-    { 0x8440, 0x847E },
-    { 0x8480, 0x84FE },
-    { 0x8540, 0x857E },
-    { 0x8580, 0x85FE },
-    { 0x8640, 0x867E },
-    { 0x8680, 0x86FE },
-    { 0x8740, 0x877E },
-    { 0x8780, 0x87FE },
-    { 0x8840, 0x887E },
-    { 0x8880, 0x88FE },
-    { 0x8940, 0x897E },
-    { 0x8980, 0x89FE },
-    { 0x8A40, 0x8A7E },
-    { 0x8A80, 0x8AFE },
-    { 0x8B40, 0x8B7E },
-    { 0x8B80, 0x8BFE },
-    { 0x8C40, 0x8C7E },
-    { 0x8C80, 0x8CFE },
-    { 0x8D40, 0x8D7E },
-    { 0x8D80, 0x8DFE },
-    { 0x8E40, 0x8E7E },
-    { 0x8E80, 0x8EFE },
-    { 0x8F40, 0x8F7E },
-    { 0x8F80, 0x8FFE },
-    { 0x9040, 0x907E },
-    { 0x9080, 0x90FE },
-    { 0x9140, 0x917E },
-    { 0x9180, 0x91FE },
-    { 0x9240, 0x927E },
-    { 0x9280, 0x92FE },
-    { 0x9340, 0x937E },
-    { 0x9380, 0x93FE },
-    { 0x9440, 0x947E },
-    { 0x9480, 0x94FE },
-    { 0x9540, 0x957E },
-    { 0x9580, 0x95FE },
-    { 0x9640, 0x967E },
-    { 0x9680, 0x96FE },
-    { 0x9740, 0x977E },
-    { 0x9780, 0x97FE },
-    { 0x9840, 0x987E },
-    { 0x9880, 0x98FE },
-    { 0x9940, 0x997E },
-    { 0x9980, 0x99FE },
-    { 0x9A40, 0x9A7E },
-    { 0x9A80, 0x9AFE },
-    { 0x9B40, 0x9B7E },
-    { 0x9B80, 0x9BFE },
-    { 0x9C40, 0x9C7E },
-    { 0x9C80, 0x9CFE },
-    { 0x9D40, 0x9D7E },
-    { 0x9D80, 0x9DFE },
-    { 0x9E40, 0x9E7E },
-    { 0x9E80, 0x9EFE },
-    { 0x9F40, 0x9F7E },
-    { 0x9F80, 0x9FFE },
-    { 0xA040, 0xA07E },
-    { 0xA080, 0xA0FE },
-    { 0xA1A1, 0xA1FE },
-    { 0xA2A1, 0xA2AA },
-    { 0xA2B1, 0xA2E2 },
-    { 0xA2E5, 0xA2EE },
-    { 0xA2F1, 0xA2FC },
-    { 0xA3A1, 0xA3FE },
-    { 0xA4A1, 0xA4F3 },
-    { 0xA5A1, 0xA5F6 },
-    { 0xA6A1, 0xA6B8 },
-    { 0xA6C1, 0xA6D8 },
-    { 0xA6E0, 0xA6EB },
-    { 0xA6EE, 0xA6F2 },
-    { 0xA6F4, 0xA6F5 },
-    { 0xA7A1, 0xA7C1 },
-    { 0xA7D1, 0xA7F1 },
-    { 0xA840, 0xA87E },
-    { 0xA880, 0xA895 },
-    { 0xA8A1, 0xA8BB },
-    { 0xA8BD, 0xA8BE },
-    { 0xA8C0, 0xA8C0 },
-    { 0xA8C5, 0xA8E9 },
-    { 0xA940, 0xA957 },
-    { 0xA959, 0xA95A },
-    { 0xA95C, 0xA95C },
-    { 0xA960, 0xA97E },
-    { 0xA980, 0xA988 },
-    { 0xA996, 0xA996 },
-    { 0xA9A4, 0xA9EF },
-    { 0xAA40, 0xAA7E },
-    { 0xAA80, 0xAAA0 },
-    { 0xAB40, 0xAB7E },
-    { 0xAB80, 0xABA0 },
-    { 0xAC40, 0xAC7E },
-    { 0xAC80, 0xACA0 },
-    { 0xAD40, 0xAD7E },
-    { 0xAD80, 0xADA0 },
-    { 0xAE40, 0xAE7E },
-    { 0xAE80, 0xAEA0 },
-    { 0xAF40, 0xAF7E },
-    { 0xAF80, 0xAFA0 },
-    { 0xB040, 0xB07E },
-    { 0xB080, 0xB0FE },
-    { 0xB140, 0xB17E },
-    { 0xB180, 0xB1FE },
-    { 0xB240, 0xB27E },
-    { 0xB280, 0xB2FE },
-    { 0xB340, 0xB37E },
-    { 0xB380, 0xB3FE },
-    { 0xB440, 0xB47E },
-    { 0xB480, 0xB4FE },
-    { 0xB540, 0xB57E },
-    { 0xB580, 0xB5FE },
-    { 0xB640, 0xB67E },
-    { 0xB680, 0xB6FE },
-    { 0xB740, 0xB77E },
-    { 0xB780, 0xB7FE },
-    { 0xB840, 0xB87E },
-    { 0xB880, 0xB8FE },
-    { 0xB940, 0xB97E },
-    { 0xB980, 0xB9FE },
-    { 0xBA40, 0xBA7E },
-    { 0xBA80, 0xBAFE },
-    { 0xBB40, 0xBB7E },
-    { 0xBB80, 0xBBFE },
-    { 0xBC40, 0xBC7E },
-    { 0xBC80, 0xBCFE },
-    { 0xBD40, 0xBD7E },
-    { 0xBD80, 0xBDFE },
-    { 0xBE40, 0xBE7E },
-    { 0xBE80, 0xBEFE },
-    { 0xBF40, 0xBF7E },
-    { 0xBF80, 0xBFFE },
-    { 0xC040, 0xC07E },
-    { 0xC080, 0xC0FE },
-    { 0xC140, 0xC17E },
-    { 0xC180, 0xC1FE },
-    { 0xC240, 0xC27E },
-    { 0xC280, 0xC2FE },
-    { 0xC340, 0xC37E },
-    { 0xC380, 0xC3FE },
-    { 0xC440, 0xC47E },
-    { 0xC480, 0xC4FE },
-    { 0xC540, 0xC57E },
-    { 0xC580, 0xC5FE },
-    { 0xC640, 0xC67E },
-    { 0xC680, 0xC6FE },
-    { 0xC740, 0xC77E },
-    { 0xC780, 0xC7FE },
-    { 0xC840, 0xC87E },
-    { 0xC880, 0xC8FE },
-    { 0xC940, 0xC97E },
-    { 0xC980, 0xC9FE },
-    { 0xCA40, 0xCA7E },
-    { 0xCA80, 0xCAFE },
-    { 0xCB40, 0xCB7E },
-    { 0xCB80, 0xCBFE },
-    { 0xCC40, 0xCC7E },
-    { 0xCC80, 0xCCFE },
-    { 0xCD40, 0xCD7E },
-    { 0xCD80, 0xCDFE },
-    { 0xCE40, 0xCE7E },
-    { 0xCE80, 0xCEFE },
-    { 0xCF40, 0xCF7E },
-    { 0xCF80, 0xCFFE },
-    { 0xD040, 0xD07E },
-    { 0xD080, 0xD0FE },
-    { 0xD140, 0xD17E },
-    { 0xD180, 0xD1FE },
-    { 0xD240, 0xD27E },
-    { 0xD280, 0xD2FE },
-    { 0xD340, 0xD37E },
-    { 0xD380, 0xD3FE },
-    { 0xD440, 0xD47E },
-    { 0xD480, 0xD4FE },
-    { 0xD540, 0xD57E },
-    { 0xD580, 0xD5FE },
-    { 0xD640, 0xD67E },
-    { 0xD680, 0xD6FE },
-    { 0xD740, 0xD77E },
-    { 0xD780, 0xD7F9 },
-    { 0xD840, 0xD87E },
-    { 0xD880, 0xD8FE },
-    { 0xD940, 0xD97E },
-    { 0xD980, 0xD9FE },
-    { 0xDA40, 0xDA7E },
-    { 0xDA80, 0xDAFE },
-    { 0xDB40, 0xDB7E },
-    { 0xDB80, 0xDBFE },
-    { 0xDC40, 0xDC7E },
-    { 0xDC80, 0xDCFE },
-    { 0xDD40, 0xDD7E },
-    { 0xDD80, 0xDDFE },
-    { 0xDE40, 0xDE7E },
-    { 0xDE80, 0xDEFE },
-    { 0xDF40, 0xDF7E },
-    { 0xDF80, 0xDFFE },
-    { 0xE040, 0xE07E },
-    { 0xE080, 0xE0FE },
-    { 0xE140, 0xE17E },
-    { 0xE180, 0xE1FE },
-    { 0xE240, 0xE27E },
-    { 0xE280, 0xE2FE },
-    { 0xE340, 0xE37E },
-    { 0xE380, 0xE3FE },
-    { 0xE440, 0xE47E },
-    { 0xE480, 0xE4FE },
-    { 0xE540, 0xE57E },
-    { 0xE580, 0xE5FE },
-    { 0xE640, 0xE67E },
-    { 0xE680, 0xE6FE },
-    { 0xE740, 0xE77E },
-    { 0xE780, 0xE7FE },
-    { 0xE840, 0xE87E },
-    { 0xE880, 0xE8FE },
-    { 0xE940, 0xE97E },
-    { 0xE980, 0xE9FE },
-    { 0xEA40, 0xEA7E },
-    { 0xEA80, 0xEAFE },
-    { 0xEB40, 0xEB7E },
-    { 0xEB80, 0xEBFE },
-    { 0xEC40, 0xEC7E },
-    { 0xEC80, 0xECFE },
-    { 0xED40, 0xED7E },
-    { 0xED80, 0xEDFE },
-    { 0xEE40, 0xEE7E },
-    { 0xEE80, 0xEEFE },
-    { 0xEF40, 0xEF7E },
-    { 0xEF80, 0xEFFE },
-    { 0xF040, 0xF07E },
-    { 0xF080, 0xF0FE },
-    { 0xF140, 0xF17E },
-    { 0xF180, 0xF1FE },
-    { 0xF240, 0xF27E },
-    { 0xF280, 0xF2FE },
-    { 0xF340, 0xF37E },
-    { 0xF380, 0xF3FE },
-    { 0xF440, 0xF47E },
-    { 0xF480, 0xF4FE },
-    { 0xF540, 0xF57E },
-    { 0xF580, 0xF5FE },
-    { 0xF640, 0xF67E },
-    { 0xF680, 0xF6FE },
-    { 0xF740, 0xF77E },
-    { 0xF780, 0xF7FE },
-    { 0xF840, 0xF87E },
-    { 0xF880, 0xF8A0 },
-    { 0xF940, 0xF97E },
-    { 0xF980, 0xF9A0 },
-    { 0xFA40, 0xFA7E },
-    { 0xFA80, 0xFAA0 },
-    { 0xFB40, 0xFB7E },
-    { 0xFB80, 0xFBA0 },
-    { 0xFC40, 0xFC7E },
-    { 0xFC80, 0xFCA0 },
-    { 0xFD40, 0xFD7E },
-    { 0xFD80, 0xFDA0 },
-    { 0xFE40, 0xFE4F },
-};
-
-// generated from http://unicode.org/Public/MAPPINGS/VENDORS/MICSFT/WINDOWS/CP949.TXT
-static const CharRange kEUCKRRanges[] = {
-    { 0x8141, 0x815A },
-    { 0x8161, 0x817A },
-    { 0x8181, 0x81FE },
-    { 0x8241, 0x825A },
-    { 0x8261, 0x827A },
-    { 0x8281, 0x82FE },
-    { 0x8341, 0x835A },
-    { 0x8361, 0x837A },
-    { 0x8381, 0x83FE },
-    { 0x8441, 0x845A },
-    { 0x8461, 0x847A },
-    { 0x8481, 0x84FE },
-    { 0x8541, 0x855A },
-    { 0x8561, 0x857A },
-    { 0x8581, 0x85FE },
-    { 0x8641, 0x865A },
-    { 0x8661, 0x867A },
-    { 0x8681, 0x86FE },
-    { 0x8741, 0x875A },
-    { 0x8761, 0x877A },
-    { 0x8781, 0x87FE },
-    { 0x8841, 0x885A },
-    { 0x8861, 0x887A },
-    { 0x8881, 0x88FE },
-    { 0x8941, 0x895A },
-    { 0x8961, 0x897A },
-    { 0x8981, 0x89FE },
-    { 0x8A41, 0x8A5A },
-    { 0x8A61, 0x8A7A },
-    { 0x8A81, 0x8AFE },
-    { 0x8B41, 0x8B5A },
-    { 0x8B61, 0x8B7A },
-    { 0x8B81, 0x8BFE },
-    { 0x8C41, 0x8C5A },
-    { 0x8C61, 0x8C7A },
-    { 0x8C81, 0x8CFE },
-    { 0x8D41, 0x8D5A },
-    { 0x8D61, 0x8D7A },
-    { 0x8D81, 0x8DFE },
-    { 0x8E41, 0x8E5A },
-    { 0x8E61, 0x8E7A },
-    { 0x8E81, 0x8EFE },
-    { 0x8F41, 0x8F5A },
-    { 0x8F61, 0x8F7A },
-    { 0x8F81, 0x8FFE },
-    { 0x9041, 0x905A },
-    { 0x9061, 0x907A },
-    { 0x9081, 0x90FE },
-    { 0x9141, 0x915A },
-    { 0x9161, 0x917A },
-    { 0x9181, 0x91FE },
-    { 0x9241, 0x925A },
-    { 0x9261, 0x927A },
-    { 0x9281, 0x92FE },
-    { 0x9341, 0x935A },
-    { 0x9361, 0x937A },
-    { 0x9381, 0x93FE },
-    { 0x9441, 0x945A },
-    { 0x9461, 0x947A },
-    { 0x9481, 0x94FE },
-    { 0x9541, 0x955A },
-    { 0x9561, 0x957A },
-    { 0x9581, 0x95FE },
-    { 0x9641, 0x965A },
-    { 0x9661, 0x967A },
-    { 0x9681, 0x96FE },
-    { 0x9741, 0x975A },
-    { 0x9761, 0x977A },
-    { 0x9781, 0x97FE },
-    { 0x9841, 0x985A },
-    { 0x9861, 0x987A },
-    { 0x9881, 0x98FE },
-    { 0x9941, 0x995A },
-    { 0x9961, 0x997A },
-    { 0x9981, 0x99FE },
-    { 0x9A41, 0x9A5A },
-    { 0x9A61, 0x9A7A },
-    { 0x9A81, 0x9AFE },
-    { 0x9B41, 0x9B5A },
-    { 0x9B61, 0x9B7A },
-    { 0x9B81, 0x9BFE },
-    { 0x9C41, 0x9C5A },
-    { 0x9C61, 0x9C7A },
-    { 0x9C81, 0x9CFE },
-    { 0x9D41, 0x9D5A },
-    { 0x9D61, 0x9D7A },
-    { 0x9D81, 0x9DFE },
-    { 0x9E41, 0x9E5A },
-    { 0x9E61, 0x9E7A },
-    { 0x9E81, 0x9EFE },
-    { 0x9F41, 0x9F5A },
-    { 0x9F61, 0x9F7A },
-    { 0x9F81, 0x9FFE },
-    { 0xA041, 0xA05A },
-    { 0xA061, 0xA07A },
-    { 0xA081, 0xA0FE },
-    { 0xA141, 0xA15A },
-    { 0xA161, 0xA17A },
-    { 0xA181, 0xA1FE },
-    { 0xA241, 0xA25A },
-    { 0xA261, 0xA27A },
-    { 0xA281, 0xA2E7 },
-    { 0xA341, 0xA35A },
-    { 0xA361, 0xA37A },
-    { 0xA381, 0xA3FE },
-    { 0xA441, 0xA45A },
-    { 0xA461, 0xA47A },
-    { 0xA481, 0xA4FE },
-    { 0xA541, 0xA55A },
-    { 0xA561, 0xA57A },
-    { 0xA581, 0xA5AA },
-    { 0xA5B0, 0xA5B9 },
-    { 0xA5C1, 0xA5D8 },
-    { 0xA5E1, 0xA5F8 },
-    { 0xA641, 0xA65A },
-    { 0xA661, 0xA67A },
-    { 0xA681, 0xA6E4 },
-    { 0xA741, 0xA75A },
-    { 0xA761, 0xA77A },
-    { 0xA781, 0xA7EF },
-    { 0xA841, 0xA85A },
-    { 0xA861, 0xA87A },
-    { 0xA881, 0xA8A4 },
-    { 0xA8A6, 0xA8A6 },
-    { 0xA8A8, 0xA8AF },
-    { 0xA8B1, 0xA8FE },
-    { 0xA941, 0xA95A },
-    { 0xA961, 0xA97A },
-    { 0xA981, 0xA9FE },
-    { 0xAA41, 0xAA5A },
-    { 0xAA61, 0xAA7A },
-    { 0xAA81, 0xAAF3 },
-    { 0xAB41, 0xAB5A },
-    { 0xAB61, 0xAB7A },
-    { 0xAB81, 0xABF6 },
-    { 0xAC41, 0xAC5A },
-    { 0xAC61, 0xAC7A },
-    { 0xAC81, 0xACC1 },
-    { 0xACD1, 0xACF1 },
-    { 0xAD41, 0xAD5A },
-    { 0xAD61, 0xAD7A },
-    { 0xAD81, 0xADA0 },
-    { 0xAE41, 0xAE5A },
-    { 0xAE61, 0xAE7A },
-    { 0xAE81, 0xAEA0 },
-    { 0xAF41, 0xAF5A },
-    { 0xAF61, 0xAF7A },
-    { 0xAF81, 0xAFA0 },
-    { 0xB041, 0xB05A },
-    { 0xB061, 0xB07A },
-    { 0xB081, 0xB0FE },
-    { 0xB141, 0xB15A },
-    { 0xB161, 0xB17A },
-    { 0xB181, 0xB1FE },
-    { 0xB241, 0xB25A },
-    { 0xB261, 0xB27A },
-    { 0xB281, 0xB2FE },
-    { 0xB341, 0xB35A },
-    { 0xB361, 0xB37A },
-    { 0xB381, 0xB3FE },
-    { 0xB441, 0xB45A },
-    { 0xB461, 0xB47A },
-    { 0xB481, 0xB4FE },
-    { 0xB541, 0xB55A },
-    { 0xB561, 0xB57A },
-    { 0xB581, 0xB5FE },
-    { 0xB641, 0xB65A },
-    { 0xB661, 0xB67A },
-    { 0xB681, 0xB6FE },
-    { 0xB741, 0xB75A },
-    { 0xB761, 0xB77A },
-    { 0xB781, 0xB7FE },
-    { 0xB841, 0xB85A },
-    { 0xB861, 0xB87A },
-    { 0xB881, 0xB8FE },
-    { 0xB941, 0xB95A },
-    { 0xB961, 0xB97A },
-    { 0xB981, 0xB9FE },
-    { 0xBA41, 0xBA5A },
-    { 0xBA61, 0xBA7A },
-    { 0xBA81, 0xBAFE },
-    { 0xBB41, 0xBB5A },
-    { 0xBB61, 0xBB7A },
-    { 0xBB81, 0xBBFE },
-    { 0xBC41, 0xBC5A },
-    { 0xBC61, 0xBC7A },
-    { 0xBC81, 0xBCFE },
-    { 0xBD41, 0xBD5A },
-    { 0xBD61, 0xBD7A },
-    { 0xBD81, 0xBDFE },
-    { 0xBE41, 0xBE5A },
-    { 0xBE61, 0xBE7A },
-    { 0xBE81, 0xBEFE },
-    { 0xBF41, 0xBF5A },
-    { 0xBF61, 0xBF7A },
-    { 0xBF81, 0xBFFE },
-    { 0xC041, 0xC05A },
-    { 0xC061, 0xC07A },
-    { 0xC081, 0xC0FE },
-    { 0xC141, 0xC15A },
-    { 0xC161, 0xC17A },
-    { 0xC181, 0xC1FE },
-    { 0xC241, 0xC25A },
-    { 0xC261, 0xC27A },
-    { 0xC281, 0xC2FE },
-    { 0xC341, 0xC35A },
-    { 0xC361, 0xC37A },
-    { 0xC381, 0xC3FE },
-    { 0xC441, 0xC45A },
-    { 0xC461, 0xC47A },
-    { 0xC481, 0xC4FE },
-    { 0xC541, 0xC55A },
-    { 0xC561, 0xC57A },
-    { 0xC581, 0xC5FE },
-    { 0xC641, 0xC652 },
-    { 0xC6A1, 0xC6FE },
-    { 0xC7A1, 0xC7FE },
-    { 0xC8A1, 0xC8FE },
-    { 0xCAA1, 0xCAFE },
-    { 0xCBA1, 0xCBFE },
-    { 0xCCA1, 0xCCFE },
-    { 0xCDA1, 0xCDFE },
-    { 0xCEA1, 0xCEFE },
-    { 0xCFA1, 0xCFFE },
-    { 0xD0A1, 0xD0FE },
-    { 0xD1A1, 0xD1FE },
-    { 0xD2A1, 0xD2FE },
-    { 0xD3A1, 0xD3FE },
-    { 0xD4A1, 0xD4FE },
-    { 0xD5A1, 0xD5FE },
-    { 0xD6A1, 0xD6FE },
-    { 0xD7A1, 0xD7FE },
-    { 0xD8A1, 0xD8FE },
-    { 0xD9A1, 0xD9FE },
-    { 0xDAA1, 0xDAFE },
-    { 0xDBA1, 0xDBFE },
-    { 0xDCA1, 0xDCFE },
-    { 0xDDA1, 0xDDFE },
-    { 0xDEA1, 0xDEFE },
-    { 0xDFA1, 0xDFFE },
-    { 0xE0A1, 0xE0FE },
-    { 0xE1A1, 0xE1FE },
-    { 0xE2A1, 0xE2FE },
-    { 0xE3A1, 0xE3FE },
-    { 0xE4A1, 0xE4FE },
-    { 0xE5A1, 0xE5FE },
-    { 0xE6A1, 0xE6FE },
-    { 0xE7A1, 0xE7FE },
-    { 0xE8A1, 0xE8FE },
-    { 0xE9A1, 0xE9FE },
-    { 0xEAA1, 0xEAFE },
-    { 0xEBA1, 0xEBFE },
-    { 0xECA1, 0xECFE },
-    { 0xEDA1, 0xEDFE },
-    { 0xEEA1, 0xEEFE },
-    { 0xEFA1, 0xEFFE },
-    { 0xF0A1, 0xF0FE },
-    { 0xF1A1, 0xF1FE },
-    { 0xF2A1, 0xF2FE },
-    { 0xF3A1, 0xF3FE },
-    { 0xF4A1, 0xF4FE },
-    { 0xF5A1, 0xF5FE },
-    { 0xF6A1, 0xF6FE },
-    { 0xF7A1, 0xF7FE },
-    { 0xF8A1, 0xF8FE },
-    { 0xF9A1, 0xF9FE },
-    { 0xFAA1, 0xFAFE },
-    { 0xFBA1, 0xFBFE },
-    { 0xFCA1, 0xFCFE },
-    { 0xFDA1, 0xFDFE },
-};
-
-// generated from http://unicode.org/Public/MAPPINGS/VENDORS/MICSFT/WINDOWS/CP950.TXT
-static const CharRange kBig5Ranges[] = {
-    { 0xA140, 0xA17E },
-    { 0xA1A1, 0xA1FE },
-    { 0xA240, 0xA27E },
-    { 0xA2A1, 0xA2FE },
-    { 0xA340, 0xA37E },
-    { 0xA3A1, 0xA3BF },
-    { 0xA3E1, 0xA3E1 },
-    { 0xA440, 0xA47E },
-    { 0xA4A1, 0xA4FE },
-    { 0xA540, 0xA57E },
-    { 0xA5A1, 0xA5FE },
-    { 0xA640, 0xA67E },
-    { 0xA6A1, 0xA6FE },
-    { 0xA740, 0xA77E },
-    { 0xA7A1, 0xA7FE },
-    { 0xA840, 0xA87E },
-    { 0xA8A1, 0xA8FE },
-    { 0xA940, 0xA97E },
-    { 0xA9A1, 0xA9FE },
-    { 0xAA40, 0xAA7E },
-    { 0xAAA1, 0xAAFE },
-    { 0xAB40, 0xAB7E },
-    { 0xABA1, 0xABFE },
-    { 0xAC40, 0xAC7E },
-    { 0xACA1, 0xACFE },
-    { 0xAD40, 0xAD7E },
-    { 0xADA1, 0xADFE },
-    { 0xAE40, 0xAE7E },
-    { 0xAEA1, 0xAEFE },
-    { 0xAF40, 0xAF7E },
-    { 0xAFA1, 0xAFFE },
-    { 0xB040, 0xB07E },
-    { 0xB0A1, 0xB0FE },
-    { 0xB140, 0xB17E },
-    { 0xB1A1, 0xB1FE },
-    { 0xB240, 0xB27E },
-    { 0xB2A1, 0xB2FE },
-    { 0xB340, 0xB37E },
-    { 0xB3A1, 0xB3FE },
-    { 0xB440, 0xB47E },
-    { 0xB4A1, 0xB4FE },
-    { 0xB540, 0xB57E },
-    { 0xB5A1, 0xB5FE },
-    { 0xB640, 0xB67E },
-    { 0xB6A1, 0xB6FE },
-    { 0xB740, 0xB77E },
-    { 0xB7A1, 0xB7FE },
-    { 0xB840, 0xB87E },
-    { 0xB8A1, 0xB8FE },
-    { 0xB940, 0xB97E },
-    { 0xB9A1, 0xB9FE },
-    { 0xBA40, 0xBA7E },
-    { 0xBAA1, 0xBAFE },
-    { 0xBB40, 0xBB7E },
-    { 0xBBA1, 0xBBFE },
-    { 0xBC40, 0xBC7E },
-    { 0xBCA1, 0xBCFE },
-    { 0xBD40, 0xBD7E },
-    { 0xBDA1, 0xBDFE },
-    { 0xBE40, 0xBE7E },
-    { 0xBEA1, 0xBEFE },
-    { 0xBF40, 0xBF7E },
-    { 0xBFA1, 0xBFFE },
-    { 0xC040, 0xC07E },
-    { 0xC0A1, 0xC0FE },
-    { 0xC140, 0xC17E },
-    { 0xC1A1, 0xC1FE },
-    { 0xC240, 0xC27E },
-    { 0xC2A1, 0xC2FE },
-    { 0xC340, 0xC37E },
-    { 0xC3A1, 0xC3FE },
-    { 0xC440, 0xC47E },
-    { 0xC4A1, 0xC4FE },
-    { 0xC540, 0xC57E },
-    { 0xC5A1, 0xC5FE },
-    { 0xC640, 0xC67E },
-    { 0xC940, 0xC97E },
-    { 0xC9A1, 0xC9FE },
-    { 0xCA40, 0xCA7E },
-    { 0xCAA1, 0xCAFE },
-    { 0xCB40, 0xCB7E },
-    { 0xCBA1, 0xCBFE },
-    { 0xCC40, 0xCC7E },
-    { 0xCCA1, 0xCCFE },
-    { 0xCD40, 0xCD7E },
-    { 0xCDA1, 0xCDFE },
-    { 0xCE40, 0xCE7E },
-    { 0xCEA1, 0xCEFE },
-    { 0xCF40, 0xCF7E },
-    { 0xCFA1, 0xCFFE },
-    { 0xD040, 0xD07E },
-    { 0xD0A1, 0xD0FE },
-    { 0xD140, 0xD17E },
-    { 0xD1A1, 0xD1FE },
-    { 0xD240, 0xD27E },
-    { 0xD2A1, 0xD2FE },
-    { 0xD340, 0xD37E },
-    { 0xD3A1, 0xD3FE },
-    { 0xD440, 0xD47E },
-    { 0xD4A1, 0xD4FE },
-    { 0xD540, 0xD57E },
-    { 0xD5A1, 0xD5FE },
-    { 0xD640, 0xD67E },
-    { 0xD6A1, 0xD6FE },
-    { 0xD740, 0xD77E },
-    { 0xD7A1, 0xD7FE },
-    { 0xD840, 0xD87E },
-    { 0xD8A1, 0xD8FE },
-    { 0xD940, 0xD97E },
-    { 0xD9A1, 0xD9FE },
-    { 0xDA40, 0xDA7E },
-    { 0xDAA1, 0xDAFE },
-    { 0xDB40, 0xDB7E },
-    { 0xDBA1, 0xDBFE },
-    { 0xDC40, 0xDC7E },
-    { 0xDCA1, 0xDCFE },
-    { 0xDD40, 0xDD7E },
-    { 0xDDA1, 0xDDFE },
-    { 0xDE40, 0xDE7E },
-    { 0xDEA1, 0xDEFE },
-    { 0xDF40, 0xDF7E },
-    { 0xDFA1, 0xDFFE },
-    { 0xE040, 0xE07E },
-    { 0xE0A1, 0xE0FE },
-    { 0xE140, 0xE17E },
-    { 0xE1A1, 0xE1FE },
-    { 0xE240, 0xE27E },
-    { 0xE2A1, 0xE2FE },
-    { 0xE340, 0xE37E },
-    { 0xE3A1, 0xE3FE },
-    { 0xE440, 0xE47E },
-    { 0xE4A1, 0xE4FE },
-    { 0xE540, 0xE57E },
-    { 0xE5A1, 0xE5FE },
-    { 0xE640, 0xE67E },
-    { 0xE6A1, 0xE6FE },
-    { 0xE740, 0xE77E },
-    { 0xE7A1, 0xE7FE },
-    { 0xE840, 0xE87E },
-    { 0xE8A1, 0xE8FE },
-    { 0xE940, 0xE97E },
-    { 0xE9A1, 0xE9FE },
-    { 0xEA40, 0xEA7E },
-    { 0xEAA1, 0xEAFE },
-    { 0xEB40, 0xEB7E },
-    { 0xEBA1, 0xEBFE },
-    { 0xEC40, 0xEC7E },
-    { 0xECA1, 0xECFE },
-    { 0xED40, 0xED7E },
-    { 0xEDA1, 0xEDFE },
-    { 0xEE40, 0xEE7E },
-    { 0xEEA1, 0xEEFE },
-    { 0xEF40, 0xEF7E },
-    { 0xEFA1, 0xEFFE },
-    { 0xF040, 0xF07E },
-    { 0xF0A1, 0xF0FE },
-    { 0xF140, 0xF17E },
-    { 0xF1A1, 0xF1FE },
-    { 0xF240, 0xF27E },
-    { 0xF2A1, 0xF2FE },
-    { 0xF340, 0xF37E },
-    { 0xF3A1, 0xF3FE },
-    { 0xF440, 0xF47E },
-    { 0xF4A1, 0xF4FE },
-    { 0xF540, 0xF57E },
-    { 0xF5A1, 0xF5FE },
-    { 0xF640, 0xF67E },
-    { 0xF6A1, 0xF6FE },
-    { 0xF740, 0xF77E },
-    { 0xF7A1, 0xF7FE },
-    { 0xF840, 0xF87E },
-    { 0xF8A1, 0xF8FE },
-    { 0xF940, 0xF97E },
-    { 0xF9A1, 0xF9FE },
-};
-
-static bool charMatchesEncoding(int ch, const CharRange* encodingRanges, int rangeCount) {
-    // Use binary search to see if the character is contained in the encoding
-    int low = 0;
-    int high = rangeCount;
-
-    while (low < high) {
-        int i = (low + high) / 2;
-        const CharRange* range = &encodingRanges[i];
-        if (ch >= range->first && ch <= range->last)
-            return true;
-        if (ch > range->last)
-            low = i + 1;
-        else
-            high = i;
-    }
-
-    return false;
-}
-
-extern uint32_t findPossibleEncodings(int ch)
-{
-    // ASCII matches everything
-    if (ch < 256) return kEncodingAll;
-
-    int result = kEncodingNone;
-
-    if (charMatchesEncoding(ch, kShiftJISRanges, ARRAY_SIZE(kShiftJISRanges)))
-        result |= kEncodingShiftJIS;
-    if (charMatchesEncoding(ch, kGBKRanges, ARRAY_SIZE(kGBKRanges)))
-        result |= kEncodingGBK;
-    if (charMatchesEncoding(ch, kBig5Ranges, ARRAY_SIZE(kBig5Ranges)))
-        result |= kEncodingBig5;
-    if (charMatchesEncoding(ch, kEUCKRRanges, ARRAY_SIZE(kEUCKRRanges)))
-        result |= kEncodingEUCKR;
-
-    return result;
-}

media/libmedia/autodetect.h

diff --git a/media/libmedia/autodetect.h b/media/libmedia/autodetect.h
deleted file mode 100644
index 9675db3..0000000
--- a/media/libmedia/autodetect.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (C) 2008 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * 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 AUTODETECT_H
-#define AUTODETECT_H
-
-#include <inttypes.h>
-
-// flags used for native encoding detection
-enum {
-    kEncodingNone               = 0,
-    kEncodingShiftJIS           = (1 << 0),
-    kEncodingGBK                = (1 << 1),
-    kEncodingBig5               = (1 << 2),
-    kEncodingEUCKR              = (1 << 3),
-
-    kEncodingAll                = (kEncodingShiftJIS | kEncodingGBK | kEncodingBig5 | kEncodingEUCKR),
-};
-
-
-// returns a bitfield containing the possible native encodings for the given character
-extern uint32_t findPossibleEncodings(int ch);
-
-#endif // AUTODETECT_H

media/libmedia/mediametadataretriever.cpp

diff --git a/media/libmedia/mediametadataretriever.cpp b/media/libmedia/mediametadataretriever.cpp
index 110b94c..1d6bb6f 100644
--- a/media/libmedia/mediametadataretriever.cpp
+++ b/media/libmedia/mediametadataretriever.cpp
@@ -21,6 +21,7 @@
 #include <binder/IServiceManager.h>
 #include <binder/IPCThreadState.h>
 #include <media/mediametadataretriever.h>
+#include <media/IMediaHTTPService.h>
 #include <media/IMediaPlayerService.h>
 #include <utils/Log.h>
 #include <dlfcn.h>
@@ -93,7 +94,9 @@
 }
 
 status_t MediaMetadataRetriever::setDataSource(
-        const char *srcUrl, const KeyedVector<String8, String8> *headers)
+        const sp<IMediaHTTPService> &httpService,
+        const char *srcUrl,
+        const KeyedVector<String8, String8> *headers)
 {
     ALOGV("setDataSource");
     Mutex::Autolock _l(mLock);
@@ -106,7 +109,7 @@
         return UNKNOWN_ERROR;
     }
     ALOGV("data source (%s)", srcUrl);
-    return mRetriever->setDataSource(srcUrl, headers);
+    return mRetriever->setDataSource(httpService, srcUrl, headers);
 }
 
 status_t MediaMetadataRetriever::setDataSource(int fd, int64_t offset, int64_t length)
@@ -157,7 +160,7 @@
     return mRetriever->extractAlbumArt();
 }
 
-void MediaMetadataRetriever::DeathNotifier::binderDied(const wp<IBinder>& who) {
+void MediaMetadataRetriever::DeathNotifier::binderDied(const wp<IBinder>& who __unused) {
     Mutex::Autolock lock(MediaMetadataRetriever::sServiceLock);
     MediaMetadataRetriever::sService.clear();
     ALOGW("MediaMetadataRetriever server died!");

media/libmedia/mediaplayer.cpp

diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 0f6d897..d94c7c5 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -58,7 +58,7 @@
     mVideoWidth = mVideoHeight = 0;
     mLockThreadId = 0;
     mAudioSessionId = AudioSystem::newAudioSessionId();
-    AudioSystem::acquireAudioSessionId(mAudioSessionId);
+    AudioSystem::acquireAudioSessionId(mAudioSessionId, -1);
     mSendLevel = 0;
     mRetransmitEndpointValid = false;
 }
@@ -66,7 +66,7 @@
 MediaPlayer::~MediaPlayer()
 {
     ALOGV("destructor");
-    AudioSystem::releaseAudioSessionId(mAudioSessionId);
+    AudioSystem::releaseAudioSessionId(mAudioSessionId, -1);
     disconnect();
     IPCThreadState::self()->flushCommands();
 }
@@ -136,6 +136,7 @@
 }
 
 status_t MediaPlayer::setDataSource(
+        const sp<IMediaHTTPService> &httpService,
         const char *url, const KeyedVector<String8, String8> *headers)
 {
     ALOGV("setDataSource(%s)", url);
@@ -145,7 +146,7 @@
         if (service != 0) {
             sp<IMediaPlayer> player(service->create(this, mAudioSessionId));
             if ((NO_ERROR != doSetRetransmitEndpoint(player)) ||
-                (NO_ERROR != player->setDataSource(url, headers))) {
+                (NO_ERROR != player->setDataSource(httpService, url, headers))) {
                 player.clear();
             }
             err = attachNewPlayer(player);
@@ -575,8 +576,8 @@
         return BAD_VALUE;
     }
     if (sessionId != mAudioSessionId) {
-        AudioSystem::acquireAudioSessionId(sessionId);
-        AudioSystem::releaseAudioSessionId(mAudioSessionId);
+        AudioSystem::acquireAudioSessionId(sessionId, -1);
+        AudioSystem::releaseAudioSessionId(mAudioSessionId, -1);
         mAudioSessionId = sessionId;
     }
     return NO_ERROR;
@@ -654,7 +655,7 @@
         return BAD_VALUE;
     }
 
-    memset(&mRetransmitEndpoint, 0, sizeof(&mRetransmitEndpoint));
+    memset(&mRetransmitEndpoint, 0, sizeof(mRetransmitEndpoint));
     mRetransmitEndpoint.sin_family = AF_INET;
     mRetransmitEndpoint.sin_addr   = saddr;
     mRetransmitEndpoint.sin_port   = htons(port);
@@ -776,15 +777,20 @@
     }
 }
 
-/*static*/ status_t MediaPlayer::decode(const char* url, uint32_t *pSampleRate,
-                                           int* pNumChannels, audio_format_t* pFormat,
-                                           const sp<IMemoryHeap>& heap, size_t *pSize)
+/*static*/ status_t MediaPlayer::decode(
+        const sp<IMediaHTTPService> &httpService,
+        const char* url,
+        uint32_t *pSampleRate,
+        int* pNumChannels,
+        audio_format_t* pFormat,
+        const sp<IMemoryHeap>& heap,
+        size_t *pSize)
 {
     ALOGV("decode(%s)", url);
     status_t status;
     const sp<IMediaPlayerService>& service = getMediaPlayerService();
     if (service != 0) {
-        status = service->decode(url, pSampleRate, pNumChannels, pFormat, heap, pSize);
+        status = service->decode(httpService, url, pSampleRate, pNumChannels, pFormat, heap, pSize);
     } else {
         ALOGE("Unable to locate media service");
         status = DEAD_OBJECT;
@@ -832,15 +838,4 @@
     return mPlayer->setNextPlayer(next == NULL ? NULL : next->mPlayer);
 }
 
-status_t MediaPlayer::updateProxyConfig(
-        const char *host, int32_t port, const char *exclusionList) {
-    const sp<IMediaPlayerService>& service = getMediaPlayerService();
-
-    if (service != NULL) {
-        return service->updateProxyConfig(host, port, exclusionList);
-    }
-
-    return INVALID_OPERATION;
-}
-
 }; // namespace android

media/libmediaplayerservice/Android.mk

diff --git a/media/libmediaplayerservice/Android.mk b/media/libmediaplayerservice/Android.mk
index 8f21632..4189a5e 100644
--- a/media/libmediaplayerservice/Android.mk
+++ b/media/libmediaplayerservice/Android.mk
@@ -45,7 +45,6 @@
     libstagefright_rtsp         \
 
 LOCAL_C_INCLUDES :=                                                 \
-    $(call include-path-for, graphics corecg)                       \
     $(TOP)/frameworks/av/media/libstagefright/include               \
     $(TOP)/frameworks/av/media/libstagefright/rtsp                  \
     $(TOP)/frameworks/av/media/libstagefright/wifi-display          \

media/libmediaplayerservice/MediaPlayerService.cpp

diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 9553458..200c561 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -44,6 +44,7 @@
 #include <utils/SystemClock.h>
 #include <utils/Vector.h>
 
+#include <media/IMediaHTTPService.h>
 #include <media/IRemoteDisplay.h>
 #include <media/IRemoteDisplayClient.h>
 #include <media/MediaPlayerInterface.h>
@@ -306,11 +307,6 @@
     return new RemoteDisplay(client, iface.string());
 }
 
-status_t MediaPlayerService::updateProxyConfig(
-        const char *host, int32_t port, const char *exclusionList) {
-    return HTTPBase::UpdateProxyConfig(host, port, exclusionList);
-}
-
 status_t MediaPlayerService::AudioCache::dump(int fd, const Vector<String16>& args) const
 {
     const size_t SIZE = 256;
@@ -319,7 +315,7 @@
 
     result.append(" AudioCache\n");
     if (mHeap != 0) {
-        snprintf(buffer, 255, "  heap base(%p), size(%d), flags(%d)\n",
+        snprintf(buffer, 255, "  heap base(%p), size(%zu), flags(%d)\n",
                 mHeap->getBase(), mHeap->getSize(), mHeap->getFlags());
         result.append(buffer);
     }
@@ -590,7 +586,8 @@
     }
 
     if (!p->hardwareOutput()) {
-        mAudioOutput = new AudioOutput(mAudioSessionId);
+        mAudioOutput = new AudioOutput(mAudioSessionId, IPCThreadState::self()->getCallingUid(),
+                mPid);
         static_cast<MediaPlayerInterface*>(p.get())->setAudioSink(mAudioOutput);
     }
 
@@ -622,7 +619,9 @@
 }
 
 status_t MediaPlayerService::Client::setDataSource(
-        const char *url, const KeyedVector<String8, String8> *headers)
+        const sp<IMediaHTTPService> &httpService,
+        const char *url,
+        const KeyedVector<String8, String8> *headers)
 {
     ALOGV("setDataSource(%s)", url);
     if (url == NULL)
@@ -657,7 +656,7 @@
             return NO_INIT;
         }
 
-        setDataSource_post(p, p->setDataSource(url, headers));
+        setDataSource_post(p, p->setDataSource(httpService, url, headers));
         return mStatus;
     }
 }
@@ -1176,9 +1175,14 @@
 }
 #endif
 
-status_t MediaPlayerService::decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
-                                       audio_format_t* pFormat,
-                                       const sp<IMemoryHeap>& heap, size_t *pSize)
+status_t MediaPlayerService::decode(
+        const sp<IMediaHTTPService> &httpService,
+        const char* url,
+        uint32_t *pSampleRate,
+        int* pNumChannels,
+        audio_format_t* pFormat,
+        const sp<IMemoryHeap>& heap,
+        size_t *pSize)
 {
     ALOGV("decode(%s)", url);
     sp<MediaPlayerBase> player;
@@ -1206,7 +1210,7 @@
     static_cast<MediaPlayerInterface*>(player.get())->setAudioSink(cache);
 
     // set data source
-    if (player->setDataSource(url) != NO_ERROR) goto Exit;
+    if (player->setDataSource(httpService, url) != NO_ERROR) goto Exit;
 
     ALOGV("prepare");
     player->prepareAsync();
@@ -1296,12 +1300,14 @@
 
 #undef LOG_TAG
 #define LOG_TAG "AudioSink"
-MediaPlayerService::AudioOutput::AudioOutput(int sessionId)
+MediaPlayerService::AudioOutput::AudioOutput(int sessionId, int uid, int pid)
     : mCallback(NULL),
       mCallbackCookie(NULL),
       mCallbackData(NULL),
       mBytesWritten(0),
       mSessionId(sessionId),
+      mUid(uid),
+      mPid(pid),
       mFlags(AUDIO_OUTPUT_FLAG_NONE) {
     ALOGV("AudioOutput(%d)", sessionId);
     mStreamType = AUDIO_STREAM_MUSIC;
@@ -1449,7 +1455,7 @@
                 format, bufferCount, mSessionId, flags);
     uint32_t afSampleRate;
     size_t afFrameCount;
-    uint32_t frameCount;
+    size_t frameCount;
 
     // offloading is only supported in callback mode for now.
     // offloadInfo must be present if offload flag is set
@@ -1549,7 +1555,9 @@
                     0,  // notification frames
                     mSessionId,
                     AudioTrack::TRANSFER_CALLBACK,
-                    offloadInfo);
+                    offloadInfo,
+                    mUid,
+                    mPid);
         } else {
             t = new AudioTrack(
                     mStreamType,
@@ -1558,10 +1566,14 @@
                     channelMask,
                     frameCount,
                     flags,
-                    NULL,
-                    NULL,
-                    0,
-                    mSessionId);
+                    NULL, // callback
+                    NULL, // user data
+                    0, // notification frames
+                    mSessionId,
+                    AudioTrack::TRANSFER_DEFAULT,
+                    NULL, // offload info
+                    mUid,
+                    mPid);
         }
 
         if ((t == 0) || (t->initCheck() != NO_ERROR)) {
@@ -1808,6 +1820,12 @@
     return mSessionId;
 }
 
+uint32_t MediaPlayerService::AudioOutput::getSampleRate() const
+{
+    if (mTrack == 0) return 0;
+    return mTrack->getSampleRate();
+}
+
 #undef LOG_TAG
 #define LOG_TAG "AudioCache"
 MediaPlayerService::AudioCache::AudioCache(const sp<IMemoryHeap>& heap) :
@@ -2010,6 +2028,14 @@
     return 0;
 }
 
+uint32_t MediaPlayerService::AudioCache::getSampleRate() const
+{
+    if (mMsecsPerFrame == 0) {
+        return 0;
+    }
+    return (uint32_t)(1.e3 / mMsecsPerFrame);
+}
+
 void MediaPlayerService::addBatteryData(uint32_t params)
 {
     Mutex::Autolock lock(mLock);

media/libmediaplayerservice/MediaPlayerService.h

diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index 21f4117..448f27a 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -72,7 +72,7 @@
         class CallbackData;
 
      public:
-                                AudioOutput(int sessionId);
+                                AudioOutput(int sessionId, int uid, int pid);
         virtual                 ~AudioOutput();
 
         virtual bool            ready() const { return mTrack != 0; }
@@ -86,6 +86,7 @@
         virtual status_t        getPosition(uint32_t *position) const;
         virtual status_t        getFramesWritten(uint32_t *frameswritten) const;
         virtual int             getSessionId() const;
+        virtual uint32_t        getSampleRate() const;
 
         virtual status_t        open(
                 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
@@ -100,7 +101,10 @@
         virtual void            flush();
         virtual void            pause();
         virtual void            close();
-                void            setAudioStreamType(audio_stream_type_t streamType) { mStreamType = streamType; }
+                void            setAudioStreamType(audio_stream_type_t streamType) {
+                                                                        mStreamType = streamType; }
+        virtual audio_stream_type_t getAudioStreamType() const { return mStreamType; }
+
                 void            setVolume(float left, float right);
         virtual status_t        setPlaybackRatePermille(int32_t ratePermille);
                 status_t        setAuxEffectSendLevel(float level);
@@ -135,6 +139,8 @@
         uint32_t                mSampleRateHz; // sample rate of the content, as set in open()
         float                   mMsecsPerFrame;
         int                     mSessionId;
+        int                     mUid;
+        int                     mPid;
         float                   mSendLevel;
         int                     mAuxEffectId;
         static bool             mIsOnEmulator;
@@ -191,6 +197,7 @@
         virtual status_t        getPosition(uint32_t *position) const;
         virtual status_t        getFramesWritten(uint32_t *frameswritten) const;
         virtual int             getSessionId() const;
+        virtual uint32_t        getSampleRate() const;
 
         virtual status_t        open(
                 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
@@ -205,9 +212,12 @@
         virtual void            flush() {}
         virtual void            pause() {}
         virtual void            close() {}
-                void            setAudioStreamType(audio_stream_type_t streamType) {}
-                void            setVolume(float left, float right) {}
-        virtual status_t        setPlaybackRatePermille(int32_t ratePermille) { return INVALID_OPERATION; }
+                void            setAudioStreamType(audio_stream_type_t streamType __unused) {}
+                // stream type is not used for AudioCache
+        virtual audio_stream_type_t getAudioStreamType() const { return AUDIO_STREAM_DEFAULT; }
+
+                void            setVolume(float left __unused, float right __unused) {}
+        virtual status_t        setPlaybackRatePermille(int32_t ratePermille __unused) { return INVALID_OPERATION; }
                 uint32_t        sampleRate() const { return mSampleRate; }
                 audio_format_t  format() const { return mFormat; }
                 size_t          size() const { return mSize; }
@@ -247,9 +257,15 @@
 
     virtual sp<IMediaPlayer>    create(const sp<IMediaPlayerClient>& client, int audioSessionId);
 
-    virtual status_t            decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
-                                       audio_format_t* pFormat,
-                                       const sp<IMemoryHeap>& heap, size_t *pSize);
+    virtual status_t            decode(
+            const sp<IMediaHTTPService> &httpService,
+            const char* url,
+            uint32_t *pSampleRate,
+            int* pNumChannels,
+            audio_format_t* pFormat,
+            const sp<IMemoryHeap>& heap,
+            size_t *pSize);
+
     virtual status_t            decode(int fd, int64_t offset, int64_t length,
                                        uint32_t *pSampleRate, int* pNumChannels,
                                        audio_format_t* pFormat,
@@ -263,9 +279,6 @@
             const String8& iface);
     virtual status_t            dump(int fd, const Vector<String16>& args);
 
-    virtual status_t        updateProxyConfig(
-            const char *host, int32_t port, const char *exclusionList);
-
             void                removeClient(wp<Client> client);
 
     // For battery usage tracking purpose
@@ -347,6 +360,7 @@
         sp<MediaPlayerBase>     createPlayer(player_type playerType);
 
         virtual status_t        setDataSource(
+                        const sp<IMediaHTTPService> &httpService,
                         const char *url,
                         const KeyedVector<String8, String8> *headers);
 

media/libmediaplayerservice/MetadataRetrieverClient.cpp

diff --git a/media/libmediaplayerservice/MetadataRetrieverClient.cpp b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
index 348957f..c61cf89 100644
--- a/media/libmediaplayerservice/MetadataRetrieverClient.cpp
+++ b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
@@ -31,6 +31,7 @@
 #include <binder/MemoryHeapBase.h>
 #include <binder/IPCThreadState.h>
 #include <binder/IServiceManager.h>
+#include <media/IMediaHTTPService.h>
 #include <media/MediaMetadataRetrieverInterface.h>
 #include <media/MediaPlayerInterface.h>
 #include <private/media/VideoFrame.h>
@@ -106,7 +107,9 @@
 }
 
 status_t MetadataRetrieverClient::setDataSource(
-        const char *url, const KeyedVector<String8, String8> *headers)
+        const sp<IMediaHTTPService> &httpService,
+        const char *url,
+        const KeyedVector<String8, String8> *headers)
 {
     ALOGV("setDataSource(%s)", url);
     Mutex::Autolock lock(mLock);
@@ -127,7 +130,7 @@
     ALOGV("player type = %d", playerType);
     sp<MediaMetadataRetrieverBase> p = createRetriever(playerType);
     if (p == NULL) return NO_INIT;
-    status_t ret = p->setDataSource(url, headers);
+    status_t ret = p->setDataSource(httpService, url, headers);
     if (ret == NO_ERROR) mRetriever = p;
     return ret;
 }

media/libmediaplayerservice/MetadataRetrieverClient.h

diff --git a/media/libmediaplayerservice/MetadataRetrieverClient.h b/media/libmediaplayerservice/MetadataRetrieverClient.h
index f08f933..9d3fbe9 100644
--- a/media/libmediaplayerservice/MetadataRetrieverClient.h
+++ b/media/libmediaplayerservice/MetadataRetrieverClient.h
@@ -30,6 +30,7 @@
 
 namespace android {
 
+struct IMediaHTTPService;
 class IMediaPlayerService;
 class MemoryDealer;
 
@@ -43,7 +44,9 @@
     virtual void                    disconnect();
 
     virtual status_t                setDataSource(
-            const char *url, const KeyedVector<String8, String8> *headers);
+            const sp<IMediaHTTPService> &httpService,
+            const char *url,
+            const KeyedVector<String8, String8> *headers);
 
     virtual status_t                setDataSource(int fd, int64_t offset, int64_t length);
     virtual sp<IMemory>             getFrameAtTime(int64_t timeUs, int option);

media/libmediaplayerservice/MidiFile.cpp

diff --git a/media/libmediaplayerservice/MidiFile.cpp b/media/libmediaplayerservice/MidiFile.cpp
index 0a6aa90..deeddd1 100644
--- a/media/libmediaplayerservice/MidiFile.cpp
+++ b/media/libmediaplayerservice/MidiFile.cpp
@@ -114,7 +114,9 @@
 }
 
 status_t MidiFile::setDataSource(
-        const char* path, const KeyedVector<String8, String8> *) {
+        const sp<IMediaHTTPService> &httpService,
+        const char* path,
+        const KeyedVector<String8, String8> *) {
     ALOGV("MidiFile::setDataSource url=%s", path);
     Mutex::Autolock lock(mMutex);
 

media/libmediaplayerservice/MidiFile.h

diff --git a/media/libmediaplayerservice/MidiFile.h b/media/libmediaplayerservice/MidiFile.h
index 24d59b4..12802ba 100644
--- a/media/libmediaplayerservice/MidiFile.h
+++ b/media/libmediaplayerservice/MidiFile.h
@@ -32,7 +32,9 @@
     virtual status_t    initCheck();
 
     virtual status_t    setDataSource(
-            const char* path, const KeyedVector<String8, String8> *headers);
+            const sp<IMediaHTTPService> &httpService,
+            const char* path,
+            const KeyedVector<String8, String8> *headers);
 
     virtual status_t    setDataSource(int fd, int64_t offset, int64_t length);
     virtual status_t    setVideoSurfaceTexture(

media/libmediaplayerservice/MidiMetadataRetriever.cpp

diff --git a/media/libmediaplayerservice/MidiMetadataRetriever.cpp b/media/libmediaplayerservice/MidiMetadataRetriever.cpp
index 465209f..f3cf6ef 100644
--- a/media/libmediaplayerservice/MidiMetadataRetriever.cpp
+++ b/media/libmediaplayerservice/MidiMetadataRetriever.cpp
@@ -22,6 +22,8 @@
 #include "MidiMetadataRetriever.h"
 #include <media/mediametadataretriever.h>
 
+#include <media/IMediaHTTPService.h>
+
 namespace android {
 
 static status_t ERROR_NOT_OPEN = -1;
@@ -36,7 +38,9 @@
 }
 
 status_t MidiMetadataRetriever::setDataSource(
-        const char *url, const KeyedVector<String8, String8> *headers)
+        const sp<IMediaHTTPService> &httpService,
+        const char *url,
+        const KeyedVector<String8, String8> *headers)
 {
     ALOGV("setDataSource: %s", url? url: "NULL pointer");
     Mutex::Autolock lock(mLock);
@@ -44,7 +48,7 @@
     if (mMidiPlayer == 0) {
         mMidiPlayer = new MidiFile();
     }
-    return mMidiPlayer->setDataSource(url, headers);
+    return mMidiPlayer->setDataSource(httpService, url, headers);
 }
 
 status_t MidiMetadataRetriever::setDataSource(int fd, int64_t offset, int64_t length)

media/libmediaplayerservice/MidiMetadataRetriever.h

diff --git a/media/libmediaplayerservice/MidiMetadataRetriever.h b/media/libmediaplayerservice/MidiMetadataRetriever.h
index 4cee42d..b8214ee 100644
--- a/media/libmediaplayerservice/MidiMetadataRetriever.h
+++ b/media/libmediaplayerservice/MidiMetadataRetriever.h
@@ -32,7 +32,9 @@
                                    ~MidiMetadataRetriever() {}
 
     virtual status_t                setDataSource(
-            const char *url, const KeyedVector<String8, String8> *headers);
+            const sp<IMediaHTTPService> &httpService,
+            const char *url,
+            const KeyedVector<String8, String8> *headers);
 
     virtual status_t                setDataSource(int fd, int64_t offset, int64_t length);
     virtual const char*             extractMetadata(int keyCode);

media/libmediaplayerservice/StagefrightPlayer.cpp

diff --git a/media/libmediaplayerservice/StagefrightPlayer.cpp b/media/libmediaplayerservice/StagefrightPlayer.cpp
index de61d9b..b37aee3 100644
--- a/media/libmediaplayerservice/StagefrightPlayer.cpp
+++ b/media/libmediaplayerservice/StagefrightPlayer.cpp
@@ -54,8 +54,10 @@
 }
 
 status_t StagefrightPlayer::setDataSource(
-        const char *url, const KeyedVector<String8, String8> *headers) {
-    return mPlayer->setDataSource(url, headers);
+        const sp<IMediaHTTPService> &httpService,
+        const char *url,
+        const KeyedVector<String8, String8> *headers) {
+    return mPlayer->setDataSource(httpService, url, headers);
 }
 
 // Warning: The filedescriptor passed into this method will only be valid until
@@ -187,7 +189,7 @@
 }
 
 status_t StagefrightPlayer::getMetadata(
-        const media::Metadata::Filter& ids, Parcel *records) {
+        const media::Metadata::Filter& /* ids */, Parcel *records) {
     using media::Metadata;
 
     uint32_t flags = mPlayer->flags();

media/libmediaplayerservice/StagefrightPlayer.h

diff --git a/media/libmediaplayerservice/StagefrightPlayer.h b/media/libmediaplayerservice/StagefrightPlayer.h
index 600945e..e6c30ff 100644
--- a/media/libmediaplayerservice/StagefrightPlayer.h
+++ b/media/libmediaplayerservice/StagefrightPlayer.h
@@ -34,7 +34,9 @@
     virtual status_t setUID(uid_t uid);
 
     virtual status_t setDataSource(
-            const char *url, const KeyedVector<String8, String8> *headers);
+            const sp<IMediaHTTPService> &httpService,
+            const char *url,
+            const KeyedVector<String8, String8> *headers);
 
     virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
 

media/libmediaplayerservice/StagefrightRecorder.cpp

diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index 095d5ca..5b7a236 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -16,6 +16,7 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "StagefrightRecorder"
+#include <inttypes.h>
 #include <utils/Log.h>
 
 #include "StagefrightRecorder.h"
@@ -24,8 +25,10 @@
 #include <binder/IServiceManager.h>
 
 #include <media/IMediaPlayerService.h>
-#include <media/openmax/OMX_Audio.h>
+#include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/ALooper.h>
 #include <media/stagefright/AudioSource.h>
 #include <media/stagefright/AMRWriter.h>
 #include <media/stagefright/AACWriter.h>
@@ -35,13 +38,12 @@
 #include <media/stagefright/MPEG4Writer.h>
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/MetaData.h>
+#include <media/stagefright/MediaCodecSource.h>
 #include <media/stagefright/OMXClient.h>
 #include <media/stagefright/OMXCodec.h>
-#include <media/stagefright/SurfaceMediaSource.h>
 #include <media/MediaProfiles.h>
 #include <camera/ICamera.h>
 #include <camera/CameraParameters.h>
-#include <gui/Surface.h>
 
 #include <utils/Errors.h>
 #include <sys/types.h>
@@ -70,8 +72,8 @@
       mOutputFd(-1),
       mAudioSource(AUDIO_SOURCE_CNT),
       mVideoSource(VIDEO_SOURCE_LIST_END),
-      mStarted(false), mSurfaceMediaSource(NULL),
-      mCaptureTimeLapse(false) {
+      mCaptureTimeLapse(false),
+      mStarted(false) {
 
     ALOGV("Constructor");
     reset();
@@ -80,10 +82,19 @@
 StagefrightRecorder::~StagefrightRecorder() {
     ALOGV("Destructor");
     stop();
+
+    if (mLooper != NULL) {
+        mLooper->stop();
+    }
 }
 
 status_t StagefrightRecorder::init() {
     ALOGV("init");
+
+    mLooper = new ALooper;
+    mLooper->setName("recorder_looper");
+    mLooper->start();
+
     return OK;
 }
 
@@ -92,7 +103,7 @@
 // while encoding GL Frames
 sp<IGraphicBufferProducer> StagefrightRecorder::querySurfaceMediaSource() const {
     ALOGV("Get SurfaceMediaSource");
-    return mSurfaceMediaSource->getBufferQueue();
+    return mGraphicBufferProducer;
 }
 
 status_t StagefrightRecorder::setAudioSource(audio_source_t as) {
@@ -232,7 +243,7 @@
     return OK;
 }
 
-status_t StagefrightRecorder::setOutputFile(const char *path) {
+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.
@@ -679,10 +690,10 @@
             return setParamTimeLapseEnable(timeLapseEnable);
         }
     } else if (key == "time-between-time-lapse-frame-capture") {
-        int64_t timeBetweenTimeLapseFrameCaptureMs;
-        if (safe_strtoi64(value.string(), &timeBetweenTimeLapseFrameCaptureMs)) {
+        int64_t timeBetweenTimeLapseFrameCaptureUs;
+        if (safe_strtoi64(value.string(), &timeBetweenTimeLapseFrameCaptureUs)) {
             return setParamTimeBetweenTimeLapseFrameCapture(
-                    1000LL * timeBetweenTimeLapseFrameCaptureMs);
+                    timeBetweenTimeLapseFrameCaptureUs);
         }
     } else {
         ALOGE("setParameter: failed to find key %s", key.string());
@@ -737,19 +748,15 @@
     return OK;
 }
 
-status_t StagefrightRecorder::prepare() {
-    return OK;
-}
-
-status_t StagefrightRecorder::start() {
-    CHECK_GE(mOutputFd, 0);
+status_t StagefrightRecorder::prepareInternal() {
+    ALOGV("prepare");
+    if (mOutputFd < 0) {
+        ALOGE("Output file descriptor is invalid");
+        return INVALID_OPERATION;
+    }
 
     // Get UID here for permission checking
     mClientUid = IPCThreadState::self()->getCallingUid();
-    if (mWriter != NULL) {
-        ALOGE("File writer is not avaialble");
-        return UNKNOWN_ERROR;
-    }
 
     status_t status = OK;
 
@@ -757,25 +764,25 @@
         case OUTPUT_FORMAT_DEFAULT:
         case OUTPUT_FORMAT_THREE_GPP:
         case OUTPUT_FORMAT_MPEG_4:
-            status = startMPEG4Recording();
+            status = setupMPEG4Recording();
             break;
 
         case OUTPUT_FORMAT_AMR_NB:
         case OUTPUT_FORMAT_AMR_WB:
-            status = startAMRRecording();
+            status = setupAMRRecording();
             break;
 
         case OUTPUT_FORMAT_AAC_ADIF:
         case OUTPUT_FORMAT_AAC_ADTS:
-            status = startAACRecording();
+            status = setupAACRecording();
             break;
 
         case OUTPUT_FORMAT_RTP_AVP:
-            status = startRTPRecording();
+            status = setupRTPRecording();
             break;
 
         case OUTPUT_FORMAT_MPEG2TS:
-            status = startMPEG2TSRecording();
+            status = setupMPEG2TSRecording();
             break;
 
         default:
@@ -784,6 +791,72 @@
             break;
     }
 
+    return status;
+}
+
+status_t StagefrightRecorder::prepare() {
+    if (mVideoSource == VIDEO_SOURCE_SURFACE) {
+        return prepareInternal();
+    }
+    return OK;
+}
+
+status_t StagefrightRecorder::start() {
+    ALOGV("start");
+    if (mOutputFd < 0) {
+        ALOGE("Output file descriptor is invalid");
+        return INVALID_OPERATION;
+    }
+
+    status_t status = OK;
+
+    if (mVideoSource != VIDEO_SOURCE_SURFACE) {
+        status = prepareInternal();
+        if (status != OK) {
+            return status;
+        }
+    }
+
+    if (mWriter == NULL) {
+        ALOGE("File writer is not avaialble");
+        return UNKNOWN_ERROR;
+    }
+
+    switch (mOutputFormat) {
+        case OUTPUT_FORMAT_DEFAULT:
+        case OUTPUT_FORMAT_THREE_GPP:
+        case OUTPUT_FORMAT_MPEG_4:
+        {
+            sp<MetaData> meta = new MetaData;
+            setupMPEG4MetaData(&meta);
+            status = mWriter->start(meta.get());
+            break;
+        }
+
+        case OUTPUT_FORMAT_AMR_NB:
+        case OUTPUT_FORMAT_AMR_WB:
+        case OUTPUT_FORMAT_AAC_ADIF:
+        case OUTPUT_FORMAT_AAC_ADTS:
+        case OUTPUT_FORMAT_RTP_AVP:
+        case OUTPUT_FORMAT_MPEG2TS:
+        {
+            status = mWriter->start();
+            break;
+        }
+
+        default:
+        {
+            ALOGE("Unsupported output file format: %d", mOutputFormat);
+            status = UNKNOWN_ERROR;
+            break;
+        }
+    }
+
+    if (status != OK) {
+        mWriter.clear();
+        mWriter = NULL;
+    }
+
     if ((status == OK) && (!mStarted)) {
         mStarted = true;
 
@@ -815,58 +888,54 @@
         return NULL;
     }
 
-    sp<MetaData> encMeta = new MetaData;
+    sp<AMessage> format = new AMessage;
     const char *mime;
     switch (mAudioEncoder) {
         case AUDIO_ENCODER_AMR_NB:
         case AUDIO_ENCODER_DEFAULT:
-            mime = MEDIA_MIMETYPE_AUDIO_AMR_NB;
+            format->setString("mime", MEDIA_MIMETYPE_AUDIO_AMR_NB);
             break;
         case AUDIO_ENCODER_AMR_WB:
-            mime = MEDIA_MIMETYPE_AUDIO_AMR_WB;
+            format->setString("mime", MEDIA_MIMETYPE_AUDIO_AMR_WB);
             break;
         case AUDIO_ENCODER_AAC:
-            mime = MEDIA_MIMETYPE_AUDIO_AAC;
-            encMeta->setInt32(kKeyAACProfile, OMX_AUDIO_AACObjectLC);
+            format->setString("mime", MEDIA_MIMETYPE_AUDIO_AAC);
+            format->setInt32("aac-profile", OMX_AUDIO_AACObjectLC);
             break;
         case AUDIO_ENCODER_HE_AAC:
-            mime = MEDIA_MIMETYPE_AUDIO_AAC;
-            encMeta->setInt32(kKeyAACProfile, OMX_AUDIO_AACObjectHE);
+            format->setString("mime", MEDIA_MIMETYPE_AUDIO_AAC);
+            format->setInt32("aac-profile", OMX_AUDIO_AACObjectHE);
             break;
         case AUDIO_ENCODER_AAC_ELD:
-            mime = MEDIA_MIMETYPE_AUDIO_AAC;
-            encMeta->setInt32(kKeyAACProfile, OMX_AUDIO_AACObjectELD);
+            format->setString("mime", MEDIA_MIMETYPE_AUDIO_AAC);
+            format->setInt32("aac-profile", OMX_AUDIO_AACObjectELD);
             break;
 
         default:
             ALOGE("Unknown audio encoder: %d", mAudioEncoder);
             return NULL;
     }
-    encMeta->setCString(kKeyMIMEType, mime);
 
     int32_t maxInputSize;
     CHECK(audioSource->getFormat()->findInt32(
                 kKeyMaxInputSize, &maxInputSize));
 
-    encMeta->setInt32(kKeyMaxInputSize, maxInputSize);
-    encMeta->setInt32(kKeyChannelCount, mAudioChannels);
-    encMeta->setInt32(kKeySampleRate, mSampleRate);
-    encMeta->setInt32(kKeyBitRate, mAudioBitRate);
+    format->setInt32("max-input-size", maxInputSize);
+    format->setInt32("channel-count", mAudioChannels);
+    format->setInt32("sample-rate", mSampleRate);
+    format->setInt32("bitrate", mAudioBitRate);
     if (mAudioTimeScale > 0) {
-        encMeta->setInt32(kKeyTimeScale, mAudioTimeScale);
+        format->setInt32("time-scale", mAudioTimeScale);
     }
 
-    OMXClient client;
-    CHECK_EQ(client.connect(), (status_t)OK);
     sp<MediaSource> audioEncoder =
-        OMXCodec::Create(client.interface(), encMeta,
-                         true /* createEncoder */, audioSource);
+            MediaCodecSource::Create(mLooper, format, audioSource);
     mAudioSourceNode = audioSource;
 
     return audioEncoder;
 }
 
-status_t StagefrightRecorder::startAACRecording() {
+status_t StagefrightRecorder::setupAACRecording() {
     // FIXME:
     // Add support for OUTPUT_FORMAT_AAC_ADIF
     CHECK_EQ(mOutputFormat, OUTPUT_FORMAT_AAC_ADTS);
@@ -877,16 +946,10 @@
     CHECK(mAudioSource != AUDIO_SOURCE_CNT);
 
     mWriter = new AACWriter(mOutputFd);
-    status_t status = startRawAudioRecording();
-    if (status != OK) {
-        mWriter.clear();
-        mWriter = NULL;
-    }
-
-    return status;
+    return setupRawAudioRecording();
 }
 
-status_t StagefrightRecorder::startAMRRecording() {
+status_t StagefrightRecorder::setupAMRRecording() {
     CHECK(mOutputFormat == OUTPUT_FORMAT_AMR_NB ||
           mOutputFormat == OUTPUT_FORMAT_AMR_WB);
 
@@ -906,15 +969,10 @@
     }
 
     mWriter = new AMRWriter(mOutputFd);
-    status_t status = startRawAudioRecording();
-    if (status != OK) {
-        mWriter.clear();
-        mWriter = NULL;
-    }
-    return status;
+    return setupRawAudioRecording();
 }
 
-status_t StagefrightRecorder::startRawAudioRecording() {
+status_t StagefrightRecorder::setupRawAudioRecording() {
     if (mAudioSource >= AUDIO_SOURCE_CNT) {
         ALOGE("Invalid audio source: %d", mAudioSource);
         return BAD_VALUE;
@@ -940,12 +998,11 @@
         mWriter->setMaxFileSize(mMaxFileSizeBytes);
     }
     mWriter->setListener(mListener);
-    mWriter->start();
 
     return OK;
 }
 
-status_t StagefrightRecorder::startRTPRecording() {
+status_t StagefrightRecorder::setupRTPRecording() {
     CHECK_EQ(mOutputFormat, OUTPUT_FORMAT_RTP_AVP);
 
     if ((mAudioSource != AUDIO_SOURCE_CNT
@@ -972,7 +1029,7 @@
             return err;
         }
 
-        err = setupVideoEncoder(mediaSource, mVideoBitRate, &source);
+        err = setupVideoEncoder(mediaSource, &source);
         if (err != OK) {
             return err;
         }
@@ -982,10 +1039,10 @@
     mWriter->addSource(source);
     mWriter->setListener(mListener);
 
-    return mWriter->start();
+    return OK;
 }
 
-status_t StagefrightRecorder::startMPEG2TSRecording() {
+status_t StagefrightRecorder::setupMPEG2TSRecording() {
     CHECK_EQ(mOutputFormat, OUTPUT_FORMAT_MPEG2TS);
 
     sp<MediaWriter> writer = new MPEG2TSWriter(mOutputFd);
@@ -1016,7 +1073,7 @@
         }
 
         sp<MediaSource> encoder;
-        err = setupVideoEncoder(mediaSource, mVideoBitRate, &encoder);
+        err = setupVideoEncoder(mediaSource, &encoder);
 
         if (err != OK) {
             return err;
@@ -1035,7 +1092,7 @@
 
     mWriter = writer;
 
-    return mWriter->start();
+    return OK;
 }
 
 void StagefrightRecorder::clipVideoFrameRate() {
@@ -1089,7 +1146,22 @@
     }
 }
 
-status_t StagefrightRecorder::checkVideoEncoderCapabilities() {
+status_t StagefrightRecorder::checkVideoEncoderCapabilities(
+        bool *supportsCameraSourceMetaDataMode) {
+    /* hardware codecs must support camera source meta data mode */
+    Vector<CodecCapabilities> codecs;
+    OMXClient client;
+    CHECK_EQ(client.connect(), (status_t)OK);
+    QueryCodecs(
+            client.interface(),
+            (mVideoEncoder == VIDEO_ENCODER_H263 ? MEDIA_MIMETYPE_VIDEO_H263 :
+             mVideoEncoder == VIDEO_ENCODER_MPEG_4_SP ? MEDIA_MIMETYPE_VIDEO_MPEG4 :
+             mVideoEncoder == VIDEO_ENCODER_H264 ? MEDIA_MIMETYPE_VIDEO_AVC : ""),
+            false /* decoder */, true /* hwCodec */, &codecs);
+    *supportsCameraSourceMetaDataMode = codecs.size() > 0;
+    ALOGV("encoder %s camera source meta-data mode",
+            *supportsCameraSourceMetaDataMode ? "supports" : "DOES NOT SUPPORT");
+
     if (!mCaptureTimeLapse) {
         // Dont clip for time lapse capture as encoder will have enough
         // time to encode because of slow capture rate of time lapse.
@@ -1261,53 +1333,20 @@
             return err;
         }
         *mediaSource = cameraSource;
-    } else if (mVideoSource == VIDEO_SOURCE_GRALLOC_BUFFER) {
-        // If using GRAlloc buffers, setup surfacemediasource.
-        // Later a handle to that will be passed
-        // to the client side when queried
-        status_t err = setupSurfaceMediaSource();
-        if (err != OK) {
-            return err;
-        }
-        *mediaSource = mSurfaceMediaSource;
+    } else if (mVideoSource == VIDEO_SOURCE_SURFACE) {
+        *mediaSource = NULL;
     } else {
         return INVALID_OPERATION;
     }
     return OK;
 }
 
-// setupSurfaceMediaSource creates a source with the given
-// width and height and framerate.
-// TODO: This could go in a static function inside SurfaceMediaSource
-// similar to that in CameraSource
-status_t StagefrightRecorder::setupSurfaceMediaSource() {
-    status_t err = OK;
-    mSurfaceMediaSource = new SurfaceMediaSource(mVideoWidth, mVideoHeight);
-    if (mSurfaceMediaSource == NULL) {
-        return NO_INIT;
-    }
-
-    if (mFrameRate == -1) {
-        int32_t frameRate = 0;
-        CHECK (mSurfaceMediaSource->getFormat()->findInt32(
-                                        kKeyFrameRate, &frameRate));
-        ALOGI("Frame rate is not explicitly set. Use the current frame "
-             "rate (%d fps)", frameRate);
-        mFrameRate = frameRate;
-    } else {
-        err = mSurfaceMediaSource->setFrameRate(mFrameRate);
-    }
-    CHECK(mFrameRate != -1);
-
-    mIsMetaDataStoredInVideoBuffers =
-        mSurfaceMediaSource->isMetaDataStoredInVideoBuffers();
-    return err;
-}
-
 status_t StagefrightRecorder::setupCameraSource(
         sp<CameraSource> *cameraSource) {
     status_t err = OK;
-    if ((err = checkVideoEncoderCapabilities()) != OK) {
+    bool encoderSupportsCameraSourceMetaDataMode;
+    if ((err = checkVideoEncoderCapabilities(
+                &encoderSupportsCameraSourceMetaDataMode)) != OK) {
         return err;
     }
     Size videoSize;
@@ -1323,13 +1362,14 @@
         mCameraSourceTimeLapse = CameraSourceTimeLapse::CreateFromCamera(
                 mCamera, mCameraProxy, mCameraId, mClientName, mClientUid,
                 videoSize, mFrameRate, mPreviewSurface,
-                mTimeBetweenTimeLapseFrameCaptureUs);
+                mTimeBetweenTimeLapseFrameCaptureUs,
+                encoderSupportsCameraSourceMetaDataMode);
         *cameraSource = mCameraSourceTimeLapse;
     } else {
         *cameraSource = CameraSource::CreateFromCamera(
                 mCamera, mCameraProxy, mCameraId, mClientName, mClientUid,
                 videoSize, mFrameRate,
-                mPreviewSurface, true /*storeMetaDataInVideoBuffers*/);
+                mPreviewSurface, encoderSupportsCameraSourceMetaDataMode);
     }
     mCamera.clear();
     mCameraProxy.clear();
@@ -1364,25 +1404,22 @@
 
 status_t StagefrightRecorder::setupVideoEncoder(
         sp<MediaSource> cameraSource,
-        int32_t videoBitRate,
         sp<MediaSource> *source) {
     source->clear();
 
-    sp<MetaData> enc_meta = new MetaData;
-    enc_meta->setInt32(kKeyBitRate, videoBitRate);
-    enc_meta->setInt32(kKeyFrameRate, mFrameRate);
+    sp<AMessage> format = new AMessage();
 
     switch (mVideoEncoder) {
         case VIDEO_ENCODER_H263:
-            enc_meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_H263);
+            format->setString("mime", MEDIA_MIMETYPE_VIDEO_H263);
             break;
 
         case VIDEO_ENCODER_MPEG_4_SP:
-            enc_meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_MPEG4);
+            format->setString("mime", MEDIA_MIMETYPE_VIDEO_MPEG4);
             break;
 
         case VIDEO_ENCODER_H264:
-            enc_meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_AVC);
+            format->setString("mime", MEDIA_MIMETYPE_VIDEO_AVC);
             break;
 
         default:
@@ -1390,59 +1427,80 @@
             break;
     }
 
-    sp<MetaData> meta = cameraSource->getFormat();
+    if (cameraSource != NULL) {
+        sp<MetaData> meta = cameraSource->getFormat();
 
-    int32_t width, height, stride, sliceHeight, colorFormat;
-    CHECK(meta->findInt32(kKeyWidth, &width));
-    CHECK(meta->findInt32(kKeyHeight, &height));
-    CHECK(meta->findInt32(kKeyStride, &stride));
-    CHECK(meta->findInt32(kKeySliceHeight, &sliceHeight));
-    CHECK(meta->findInt32(kKeyColorFormat, &colorFormat));
+        int32_t width, height, stride, sliceHeight, colorFormat;
+        CHECK(meta->findInt32(kKeyWidth, &width));
+        CHECK(meta->findInt32(kKeyHeight, &height));
+        CHECK(meta->findInt32(kKeyStride, &stride));
+        CHECK(meta->findInt32(kKeySliceHeight, &sliceHeight));
+        CHECK(meta->findInt32(kKeyColorFormat, &colorFormat));
 
-    enc_meta->setInt32(kKeyWidth, width);
-    enc_meta->setInt32(kKeyHeight, height);
-    enc_meta->setInt32(kKeyIFramesInterval, mIFramesIntervalSec);
-    enc_meta->setInt32(kKeyStride, stride);
-    enc_meta->setInt32(kKeySliceHeight, sliceHeight);
-    enc_meta->setInt32(kKeyColorFormat, colorFormat);
+        format->setInt32("width", width);
+        format->setInt32("height", height);
+        format->setInt32("stride", stride);
+        format->setInt32("slice-height", sliceHeight);
+        format->setInt32("color-format", colorFormat);
+    } else {
+        format->setInt32("width", mVideoWidth);
+        format->setInt32("height", mVideoHeight);
+        format->setInt32("stride", mVideoWidth);
+        format->setInt32("slice-height", mVideoWidth);
+        format->setInt32("color-format", OMX_COLOR_FormatAndroidOpaque);
+
+        // set up time lapse/slow motion for surface source
+        if (mCaptureTimeLapse) {
+            if (mTimeBetweenTimeLapseFrameCaptureUs <= 0) {
+                ALOGE("Invalid mTimeBetweenTimeLapseFrameCaptureUs value: %lld",
+                    mTimeBetweenTimeLapseFrameCaptureUs);
+                return BAD_VALUE;
+            }
+            format->setInt64("time-lapse",
+                    mTimeBetweenTimeLapseFrameCaptureUs);
+        }
+    }
+
+    format->setInt32("bitrate", mVideoBitRate);
+    format->setInt32("frame-rate", mFrameRate);
+    format->setInt32("i-frame-interval", mIFramesIntervalSec);
+
     if (mVideoTimeScale > 0) {
-        enc_meta->setInt32(kKeyTimeScale, mVideoTimeScale);
+        format->setInt32("time-scale", mVideoTimeScale);
     }
     if (mVideoEncoderProfile != -1) {
-        enc_meta->setInt32(kKeyVideoProfile, mVideoEncoderProfile);
+        format->setInt32("profile", mVideoEncoderProfile);
     }
     if (mVideoEncoderLevel != -1) {
-        enc_meta->setInt32(kKeyVideoLevel, mVideoEncoderLevel);
+        format->setInt32("level", mVideoEncoderLevel);
     }
 
-    OMXClient client;
-    CHECK_EQ(client.connect(), (status_t)OK);
-
-    uint32_t encoder_flags = 0;
+    uint32_t flags = 0;
     if (mIsMetaDataStoredInVideoBuffers) {
-        encoder_flags |= OMXCodec::kStoreMetaDataInVideoBuffers;
+        flags |= MediaCodecSource::FLAG_USE_METADATA_INPUT;
     }
 
-    // Do not wait for all the input buffers to become available.
-    // This give timelapse video recording faster response in
-    // receiving output from video encoder component.
-    if (mCaptureTimeLapse) {
-        encoder_flags |= OMXCodec::kOnlySubmitOneInputBufferAtOneTime;
+    if (cameraSource == NULL) {
+        flags |= MediaCodecSource::FLAG_USE_SURFACE_INPUT;
     }
 
-    sp<MediaSource> encoder = OMXCodec::Create(
-            client.interface(), enc_meta,
-            true /* createEncoder */, cameraSource,
-            NULL, encoder_flags);
+    sp<MediaCodecSource> encoder =
+            MediaCodecSource::Create(mLooper, format, cameraSource, flags);
     if (encoder == NULL) {
         ALOGW("Failed to create the encoder");
         // When the encoder fails to be created, we need
         // release the camera source due to the camera's lock
         // and unlock mechanism.
-        cameraSource->stop();
+        if (cameraSource != NULL) {
+            cameraSource->stop();
+        }
         return UNKNOWN_ERROR;
     }
 
+    if (cameraSource == NULL) {
+        mGraphicBufferProducer = encoder->getGraphicBufferProducer();
+    }
+
     *source = encoder;
 
     return OK;
@@ -1476,16 +1534,12 @@
     return OK;
 }
 
-status_t StagefrightRecorder::setupMPEG4Recording(
-        int outputFd,
-        int32_t videoWidth, int32_t videoHeight,
-        int32_t videoBitRate,
-        int32_t *totalBitRate,
-        sp<MediaWriter> *mediaWriter) {
-    mediaWriter->clear();
-    *totalBitRate = 0;
+status_t StagefrightRecorder::setupMPEG4Recording() {
+    mWriter.clear();
+    mTotalBitRate = 0;
+
     status_t err = OK;
-    sp<MediaWriter> writer = new MPEG4Writer(outputFd);
+    sp<MediaWriter> writer = new MPEG4Writer(mOutputFd);
 
     if (mVideoSource < VIDEO_SOURCE_LIST_END) {
 
@@ -1496,13 +1550,13 @@
         }
 
         sp<MediaSource> encoder;
-        err = setupVideoEncoder(mediaSource, videoBitRate, &encoder);
+        err = setupVideoEncoder(mediaSource, &encoder);
         if (err != OK) {
             return err;
         }
 
         writer->addSource(encoder);
-        *totalBitRate += videoBitRate;
+        mTotalBitRate += mVideoBitRate;
     }
 
     // Audio source is added at the end if it exists.
@@ -1511,7 +1565,7 @@
     if (!mCaptureTimeLapse && (mAudioSource != AUDIO_SOURCE_CNT)) {
         err = setupAudioEncoder(writer);
         if (err != OK) return err;
-        *totalBitRate += mAudioBitRate;
+        mTotalBitRate += mAudioBitRate;
     }
 
     if (mInterleaveDurationUs > 0) {
@@ -1529,22 +1583,28 @@
         writer->setMaxFileSize(mMaxFileSizeBytes);
     }
 
-    mStartTimeOffsetMs = mEncoderProfiles->getStartTimeOffsetMs(mCameraId);
+    if (mVideoSource == VIDEO_SOURCE_DEFAULT
+            || mVideoSource == VIDEO_SOURCE_CAMERA) {
+        mStartTimeOffsetMs = mEncoderProfiles->getStartTimeOffsetMs(mCameraId);
+    } else if (mVideoSource == VIDEO_SOURCE_SURFACE) {
+        // surface source doesn't need large initial delay
+        mStartTimeOffsetMs = 200;
+    }
     if (mStartTimeOffsetMs > 0) {
         reinterpret_cast<MPEG4Writer *>(writer.get())->
             setStartTimeOffsetMs(mStartTimeOffsetMs);
     }
 
     writer->setListener(mListener);
-    *mediaWriter = writer;
+    mWriter = writer;
     return OK;
 }
 
-void StagefrightRecorder::setupMPEG4MetaData(int64_t startTimeUs, int32_t totalBitRate,
-        sp<MetaData> *meta) {
+void StagefrightRecorder::setupMPEG4MetaData(sp<MetaData> *meta) {
+    int64_t startTimeUs = systemTime() / 1000;
     (*meta)->setInt64(kKeyTime, startTimeUs);
     (*meta)->setInt32(kKeyFileType, mOutputFormat);
-    (*meta)->setInt32(kKeyBitRate, totalBitRate);
+    (*meta)->setInt32(kKeyBitRate, mTotalBitRate);
     (*meta)->setInt32(kKey64BitFileOffset, mUse64BitFileOffset);
     if (mMovieTimeScale > 0) {
         (*meta)->setInt32(kKeyTimeScale, mMovieTimeScale);
@@ -1557,27 +1617,6 @@
     }
 }
 
-status_t StagefrightRecorder::startMPEG4Recording() {
-    int32_t totalBitRate;
-    status_t err = setupMPEG4Recording(
-            mOutputFd, mVideoWidth, mVideoHeight,
-            mVideoBitRate, &totalBitRate, &mWriter);
-    if (err != OK) {
-        return err;
-    }
-
-    int64_t startTimeUs = systemTime() / 1000;
-    sp<MetaData> meta = new MetaData;
-    setupMPEG4MetaData(startTimeUs, totalBitRate, &meta);
-
-    err = mWriter->start(meta.get());
-    if (err != OK) {
-        return err;
-    }
-
-    return OK;
-}
-
 status_t StagefrightRecorder::pause() {
     ALOGV("pause");
     if (mWriter == NULL) {
@@ -1617,6 +1656,8 @@
         mWriter.clear();
     }
 
+    mGraphicBufferProducer.clear();
+
     if (mOutputFd >= 0) {
         ::close(mOutputFd);
         mOutputFd = -1;
@@ -1636,7 +1677,6 @@
         addBatteryData(params);
     }
 
-
     return err;
 }
 
@@ -1688,6 +1728,7 @@
     mRotationDegrees = 0;
     mLatitudex10000 = -3600000;
     mLongitudex10000 = -3600000;
+    mTotalBitRate = 0;
 
     mOutputFd = -1;
 
@@ -1728,15 +1769,15 @@
     result.append(buffer);
     snprintf(buffer, SIZE, "     File format: %d\n", mOutputFormat);
     result.append(buffer);
-    snprintf(buffer, SIZE, "     Max file size (bytes): %lld\n", mMaxFileSizeBytes);
+    snprintf(buffer, SIZE, "     Max file size (bytes): %" PRId64 "\n", mMaxFileSizeBytes);
     result.append(buffer);
-    snprintf(buffer, SIZE, "     Max file duration (us): %lld\n", mMaxFileDurationUs);
+    snprintf(buffer, SIZE, "     Max file duration (us): %" PRId64 "\n", mMaxFileDurationUs);
     result.append(buffer);
     snprintf(buffer, SIZE, "     File offset length (bits): %d\n", mUse64BitFileOffset? 64: 32);
     result.append(buffer);
     snprintf(buffer, SIZE, "     Interleave duration (us): %d\n", mInterleaveDurationUs);
     result.append(buffer);
-    snprintf(buffer, SIZE, "     Progress notification: %lld us\n", mTrackEveryTimeDurationUs);
+    snprintf(buffer, SIZE, "     Progress notification: %" PRId64 " us\n", mTrackEveryTimeDurationUs);
     result.append(buffer);
     snprintf(buffer, SIZE, "   Audio\n");
     result.append(buffer);

media/libmediaplayerservice/StagefrightRecorder.h

diff --git a/media/libmediaplayerservice/StagefrightRecorder.h b/media/libmediaplayerservice/StagefrightRecorder.h
index c864207..377d168 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.h
+++ b/media/libmediaplayerservice/StagefrightRecorder.h
@@ -37,6 +37,7 @@
 class MediaProfiles;
 class IGraphicBufferProducer;
 class SurfaceMediaSource;
+class ALooper;
 
 struct StagefrightRecorder : public MediaRecorderBase {
     StagefrightRecorder();
@@ -106,6 +107,7 @@
     int32_t mLatitudex10000;
     int32_t mLongitudex10000;
     int32_t mStartTimeOffsetMs;
+    int32_t mTotalBitRate;
 
     bool mCaptureTimeLapse;
     int64_t mTimeBetweenTimeLapseFrameCaptureUs;
@@ -122,38 +124,28 @@
     // An <IGraphicBufferProducer> pointer
     // will be sent to the client side using which the
     // frame buffers will be queued and dequeued
-    sp<SurfaceMediaSource> mSurfaceMediaSource;
+    sp<IGraphicBufferProducer> mGraphicBufferProducer;
+    sp<ALooper> mLooper;
 
-    status_t setupMPEG4Recording(
-        int outputFd,
-        int32_t videoWidth, int32_t videoHeight,
-        int32_t videoBitRate,
-        int32_t *totalBitRate,
-        sp<MediaWriter> *mediaWriter);
-    void setupMPEG4MetaData(int64_t startTimeUs, int32_t totalBitRate,
-        sp<MetaData> *meta);
-    status_t startMPEG4Recording();
-    status_t startAMRRecording();
-    status_t startAACRecording();
-    status_t startRawAudioRecording();
-    status_t startRTPRecording();
-    status_t startMPEG2TSRecording();
+    status_t prepareInternal();
+    status_t setupMPEG4Recording();
+    void setupMPEG4MetaData(sp<MetaData> *meta);
+    status_t setupAMRRecording();
+    status_t setupAACRecording();
+    status_t setupRawAudioRecording();
+    status_t setupRTPRecording();
+    status_t setupMPEG2TSRecording();
     sp<MediaSource> createAudioSource();
-    status_t checkVideoEncoderCapabilities();
+    status_t checkVideoEncoderCapabilities(
+            bool *supportsCameraSourceMetaDataMode);
     status_t checkAudioEncoderCapabilities();
     // Generic MediaSource set-up. Returns the appropriate
     // source (CameraSource or SurfaceMediaSource)
     // depending on the videosource type
     status_t setupMediaSource(sp<MediaSource> *mediaSource);
     status_t setupCameraSource(sp<CameraSource> *cameraSource);
-    // setup the surfacemediasource for the encoder
-    status_t setupSurfaceMediaSource();
-
     status_t setupAudioEncoder(const sp<MediaWriter>& writer);
-    status_t setupVideoEncoder(
-            sp<MediaSource> cameraSource,
-            int32_t videoBitRate,
-            sp<MediaSource> *source);
+    status_t setupVideoEncoder(sp<MediaSource> cameraSource, sp<MediaSource> *source);
 
     // Encoding parameter handling utilities
     status_t setParameter(const String8 &key, const String8 &value);

media/libmediaplayerservice/TestPlayerStub.cpp

diff --git a/media/libmediaplayerservice/TestPlayerStub.cpp b/media/libmediaplayerservice/TestPlayerStub.cpp
index 5d9728a..5795773 100644
--- a/media/libmediaplayerservice/TestPlayerStub.cpp
+++ b/media/libmediaplayerservice/TestPlayerStub.cpp
@@ -113,7 +113,9 @@
 // Create the test player.
 // Call setDataSource on the test player with the url in param.
 status_t TestPlayerStub::setDataSource(
-        const char *url, const KeyedVector<String8, String8> *headers) {
+        const sp<IMediaHTTPService> &httpService,
+        const char *url,
+        const KeyedVector<String8, String8> *headers) {
     if (!isTestUrl(url) || NULL != mHandle) {
         return INVALID_OPERATION;
     }
@@ -162,7 +164,7 @@
     }
 
     mPlayer = (*mNewPlayer)();
-    return mPlayer->setDataSource(mContentUrl, headers);
+    return mPlayer->setDataSource(httpService, mContentUrl, headers);
 }
 
 // Internal cleanup.

media/libmediaplayerservice/TestPlayerStub.h

diff --git a/media/libmediaplayerservice/TestPlayerStub.h b/media/libmediaplayerservice/TestPlayerStub.h
index a3802eb..55bf2c8 100644
--- a/media/libmediaplayerservice/TestPlayerStub.h
+++ b/media/libmediaplayerservice/TestPlayerStub.h
@@ -66,7 +66,9 @@
 
     // @param url Should be a test url. See class comment.
     virtual status_t setDataSource(
-            const char* url, const KeyedVector<String8, String8> *headers);
+            const sp<IMediaHTTPService> &httpService,
+            const char* url,
+            const KeyedVector<String8, String8> *headers);
 
     // Test player for a file descriptor source is not supported.
     virtual status_t setDataSource(int, int64_t, int64_t)  {

media/libmediaplayerservice/nuplayer/Android.mk

diff --git a/media/libmediaplayerservice/nuplayer/Android.mk b/media/libmediaplayerservice/nuplayer/Android.mk
index f946c1c..f97ba57 100644
--- a/media/libmediaplayerservice/nuplayer/Android.mk
+++ b/media/libmediaplayerservice/nuplayer/Android.mk
@@ -11,7 +11,6 @@
         NuPlayerStreamListener.cpp      \
         RTSPSource.cpp                  \
         StreamingSource.cpp             \
-        mp4/MP4Source.cpp               \
 
 LOCAL_C_INCLUDES := \
 	$(TOP)/frameworks/av/media/libstagefright/httplive            \

media/libmediaplayerservice/nuplayer/GenericSource.cpp

diff --git a/media/libmediaplayerservice/nuplayer/GenericSource.cpp b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
index b04e7a6..06aac33 100644
--- a/media/libmediaplayerservice/nuplayer/GenericSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
@@ -33,17 +33,16 @@
 
 NuPlayer::GenericSource::GenericSource(
         const sp<AMessage> &notify,
+        const sp<IMediaHTTPService> &httpService,
         const char *url,
-        const KeyedVector<String8, String8> *headers,
-        bool uidValid,
-        uid_t uid)
+        const KeyedVector<String8, String8> *headers)
     : Source(notify),
       mDurationUs(0ll),
       mAudioIsVorbis(false) {
     DataSource::RegisterDefaultSniffers();
 
     sp<DataSource> dataSource =
-        DataSource::CreateFromURI(url, headers);
+        DataSource::CreateFromURI(httpService, url, headers);
     CHECK(dataSource != NULL);
 
     initFromDataSource(dataSource);

media/libmediaplayerservice/nuplayer/GenericSource.h

diff --git a/media/libmediaplayerservice/nuplayer/GenericSource.h b/media/libmediaplayerservice/nuplayer/GenericSource.h
index 2da680c..20d597e 100644
--- a/media/libmediaplayerservice/nuplayer/GenericSource.h
+++ b/media/libmediaplayerservice/nuplayer/GenericSource.h
@@ -33,10 +33,9 @@
 struct NuPlayer::GenericSource : public NuPlayer::Source {
     GenericSource(
             const sp<AMessage> &notify,
+            const sp<IMediaHTTPService> &httpService,
             const char *url,
-            const KeyedVector<String8, String8> *headers,
-            bool uidValid = false,
-            uid_t uid = 0);
+            const KeyedVector<String8, String8> *headers);
 
     GenericSource(
             const sp<AMessage> &notify,

media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp

diff --git a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
index d8b35d7..cbedf5c 100644
--- a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
@@ -24,6 +24,7 @@
 #include "LiveDataSource.h"
 #include "LiveSession.h"
 
+#include <media/IMediaHTTPService.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AMessage.h>
@@ -34,13 +35,12 @@
 
 NuPlayer::HTTPLiveSource::HTTPLiveSource(
         const sp<AMessage> &notify,
+        const sp<IMediaHTTPService> &httpService,
         const char *url,
-        const KeyedVector<String8, String8> *headers,
-        bool uidValid, uid_t uid)
+        const KeyedVector<String8, String8> *headers)
     : Source(notify),
+      mHTTPService(httpService),
       mURL(url),
-      mUIDValid(uidValid),
-      mUID(uid),
       mFlags(0),
       mFinalResult(OK),
       mOffset(0),
@@ -79,8 +79,7 @@
     mLiveSession = new LiveSession(
             notify,
             (mFlags & kFlagIncognito) ? LiveSession::kFlagIncognito : 0,
-            mUIDValid,
-            mUID);
+            mHTTPService);
 
     mLiveLooper->registerHandler(mLiveSession);
 
@@ -140,7 +139,7 @@
     // LiveSession::selectTrack returns BAD_VALUE when selecting the currently
     // selected track, or unselecting a non-selected track. In this case it's an
     // no-op so we return OK.
-    return (err == OK || err == BAD_VALUE) ? OK : err;
+    return (err == OK || err == BAD_VALUE) ? (status_t)OK : err;
 }
 
 status_t NuPlayer::HTTPLiveSource::seekTo(int64_t seekTimeUs) {
@@ -201,7 +200,16 @@
     switch (what) {
         case LiveSession::kWhatPrepared:
         {
-            notifyVideoSizeChanged(0, 0);
+            // notify the current size here if we have it, otherwise report an initial size of (0,0)
+            sp<AMessage> format = getFormat(false /* audio */);
+            int32_t width;
+            int32_t height;
+            if (format != NULL &&
+                    format->findInt32("width", &width) && format->findInt32("height", &height)) {
+                notifyVideoSizeChanged(width, height);
+            } else {
+                notifyVideoSizeChanged(0, 0);
+            }
 
             uint32_t flags = FLAG_CAN_PAUSE;
             if (mLiveSession->isSeekable()) {

media/libmediaplayerservice/nuplayer/HTTPLiveSource.h

diff --git a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h
index bcc3f8b..4d7251f 100644
--- a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h
+++ b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h
@@ -28,10 +28,9 @@
 struct NuPlayer::HTTPLiveSource : public NuPlayer::Source {
     HTTPLiveSource(
             const sp<AMessage> &notify,
+            const sp<IMediaHTTPService> &httpService,
             const char *url,
-            const KeyedVector<String8, String8> *headers,
-            bool uidValid = false,
-            uid_t uid = 0);
+            const KeyedVector<String8, String8> *headers);
 
     virtual void prepareAsync();
     virtual void start();
@@ -61,10 +60,9 @@
         kWhatFetchSubtitleData,
     };
 
+    sp<IMediaHTTPService> mHTTPService;
     AString mURL;
     KeyedVector<String8, String8> mExtraHeaders;
-    bool mUIDValid;
-    uid_t mUID;
     uint32_t mFlags;
     status_t mFinalResult;
     off64_t mOffset;

media/libmediaplayerservice/nuplayer/NuPlayer.cpp

diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index 750287f..a750ad0 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -28,11 +28,11 @@
 #include "RTSPSource.h"
 #include "StreamingSource.h"
 #include "GenericSource.h"
-#include "mp4/MP4Source.h"
 
 #include "ATSParser.h"
 
-#include <cutils/properties.h> // for property_get
+#include "SoftwareRenderer.h"
+
 #include <media/stagefright/foundation/hexdump.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -146,6 +146,7 @@
     : mUIDValid(false),
       mSourceFlags(0),
       mVideoIsAVC(false),
+      mNeedsSwRenderer(false),
       mAudioEOS(false),
       mVideoEOS(false),
       mScanSourcesPending(false),
@@ -180,14 +181,7 @@
 
     sp<AMessage> notify = new AMessage(kWhatSourceNotify, id());
 
-    char prop[PROPERTY_VALUE_MAX];
-    if (property_get("media.stagefright.use-mp4source", prop, NULL)
-            && (!strcmp(prop, "1") || !strcasecmp(prop, "true"))) {
-        msg->setObject("source", new MP4Source(notify, source));
-    } else {
-        msg->setObject("source", new StreamingSource(notify, source));
-    }
-
+    msg->setObject("source", new StreamingSource(notify, source));
     msg->post();
 }
 
@@ -209,7 +203,9 @@
 }
 
 void NuPlayer::setDataSourceAsync(
-        const char *url, const KeyedVector<String8, String8> *headers) {
+        const sp<IMediaHTTPService> &httpService,
+        const char *url,
+        const KeyedVector<String8, String8> *headers) {
     sp<AMessage> msg = new AMessage(kWhatSetDataSource, id());
     size_t len = strlen(url);
 
@@ -217,16 +213,18 @@
 
     sp<Source> source;
     if (IsHTTPLiveURL(url)) {
-        source = new HTTPLiveSource(notify, url, headers, mUIDValid, mUID);
+        source = new HTTPLiveSource(notify, httpService, url, headers);
     } else if (!strncasecmp(url, "rtsp://", 7)) {
-        source = new RTSPSource(notify, url, headers, mUIDValid, mUID);
+        source = new RTSPSource(
+                notify, httpService, url, headers, mUIDValid, mUID);
     } else if ((!strncasecmp(url, "http://", 7)
                 || !strncasecmp(url, "https://", 8))
                     && ((len >= 4 && !strcasecmp(".sdp", &url[len - 4]))
                     || strstr(url, ".sdp?"))) {
-        source = new RTSPSource(notify, url, headers, mUIDValid, mUID, true);
+        source = new RTSPSource(
+                notify, httpService, url, headers, mUIDValid, mUID, true);
     } else {
-        source = new GenericSource(notify, url, headers, mUIDValid, mUID);
+        source = new GenericSource(notify, httpService, url, headers);
     }
 
     msg->setObject("source", source);
@@ -444,6 +442,7 @@
             ALOGV("kWhatStart");
 
             mVideoIsAVC = false;
+            mNeedsSwRenderer = false;
             mAudioEOS = false;
             mVideoEOS = false;
             mSkipRenderingAudioUntilMediaTimeUs = -1;
@@ -680,6 +679,20 @@
 
                     notifyListener(
                             MEDIA_SET_VIDEO_SIZE, displayWidth, displayHeight);
+
+                    if (mNeedsSwRenderer && mNativeWindow != NULL) {
+                        int32_t colorFormat;
+                        CHECK(codecRequest->findInt32("color-format", &colorFormat));
+
+                        sp<MetaData> meta = new MetaData;
+                        meta->setInt32(kKeyWidth, width);
+                        meta->setInt32(kKeyHeight, height);
+                        meta->setRect(kKeyCropRect, cropLeft, cropTop, cropRight, cropBottom);
+                        meta->setInt32(kKeyColorFormat, colorFormat);
+
+                        mRenderer->setSoftRenderer(
+                                new SoftwareRenderer(mNativeWindow->getNativeWindow(), meta));
+                    }
                 }
             } else if (what == ACodec::kWhatShutdownCompleted) {
                 ALOGV("%s shutdown completed", audio ? "audio" : "video");
@@ -703,8 +716,13 @@
                 mRenderer->queueEOS(audio, UNKNOWN_ERROR);
             } else if (what == ACodec::kWhatDrainThisBuffer) {
                 renderBuffer(audio, codecRequest);
-            } else if (what != ACodec::kWhatComponentAllocated
-                    && what != ACodec::kWhatComponentConfigured
+            } else if (what == ACodec::kWhatComponentAllocated) {
+                if (!audio) {
+                    AString name;
+                    CHECK(codecRequest->findString("componentName", &name));
+                    mNeedsSwRenderer = name.startsWith("OMX.google.");
+                }
+            } else if (what != ACodec::kWhatComponentConfigured
                     && what != ACodec::kWhatBuffersAllocated) {
                 ALOGV("Unhandled codec notification %d '%c%c%c%c'.",
                       what,
@@ -988,7 +1006,14 @@
                                     &NuPlayer::performScanSources));
                     }
 
-                    flushDecoder(audio, formatChange);
+                    sp<AMessage> newFormat = mSource->getFormat(audio);
+                    sp<Decoder> &decoder = audio ? mAudioDecoder : mVideoDecoder;
+                    if (formatChange && !decoder->supportsSeamlessFormatChange(newFormat)) {
+                        flushDecoder(audio, /* needShutdown = */ true);
+                    } else {
+                        flushDecoder(audio, /* needShutdown = */ false);
+                        err = OK;
+                    }
                 } else {
                     // This stream is unaffected by the discontinuity
 
@@ -1501,7 +1526,7 @@
     notify->post();
 }
 
-void NuPlayer::Source::onMessageReceived(const sp<AMessage> &msg) {
+void NuPlayer::Source::onMessageReceived(const sp<AMessage> & /* msg */) {
     TRESPASS();
 }
 

media/libmediaplayerservice/nuplayer/NuPlayer.h

diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.h b/media/libmediaplayerservice/nuplayer/NuPlayer.h
index 13350f3..9dfe4a0 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.h
@@ -38,7 +38,9 @@
     void setDataSourceAsync(const sp<IStreamSource> &source);
 
     void setDataSourceAsync(
-            const char *url, const KeyedVector<String8, String8> *headers);
+            const sp<IMediaHTTPService> &httpService,
+            const char *url,
+            const KeyedVector<String8, String8> *headers);
 
     void setDataSourceAsync(int fd, int64_t offset, int64_t length);
 
@@ -116,6 +118,7 @@
     sp<MediaPlayerBase::AudioSink> mAudioSink;
     sp<Decoder> mVideoDecoder;
     bool mVideoIsAVC;
+    bool mNeedsSwRenderer;
     sp<Decoder> mAudioDecoder;
     sp<Renderer> mRenderer;
 

media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp

diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
index 22f699e..2423fd5 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
@@ -67,6 +67,7 @@
     // queue.
     bool needDedicatedLooper = !strncasecmp(mime.c_str(), "video/", 6);
 
+    mFormat = format;
     mCodec = new ACodec;
 
     if (needDedicatedLooper && mCodecLooper == NULL) {
@@ -147,5 +148,65 @@
     }
 }
 
+bool NuPlayer::Decoder::supportsSeamlessAudioFormatChange(const sp<AMessage> &targetFormat) const {
+    if (targetFormat == NULL) {
+        return true;
+    }
+
+    AString mime;
+    if (!targetFormat->findString("mime", &mime)) {
+        return false;
+    }
+
+    if (!strcasecmp(mime.c_str(), MEDIA_MIMETYPE_AUDIO_AAC)) {
+        // field-by-field comparison
+        const char * keys[] = { "channel-count", "sample-rate", "is-adts" };
+        for (unsigned int i = 0; i < sizeof(keys) / sizeof(keys[0]); i++) {
+            int32_t oldVal, newVal;
+            if (!mFormat->findInt32(keys[i], &oldVal) || !targetFormat->findInt32(keys[i], &newVal)
+                    || oldVal != newVal) {
+                return false;
+            }
+        }
+
+        sp<ABuffer> oldBuf, newBuf;
+        if (mFormat->findBuffer("csd-0", &oldBuf) && targetFormat->findBuffer("csd-0", &newBuf)) {
+            if (oldBuf->size() != newBuf->size()) {
+                return false;
+            }
+            return !memcmp(oldBuf->data(), newBuf->data(), oldBuf->size());
+        }
+    }
+    return false;
+}
+
+bool NuPlayer::Decoder::supportsSeamlessFormatChange(const sp<AMessage> &targetFormat) const {
+    if (mFormat == NULL) {
+        return false;
+    }
+
+    if (targetFormat == NULL) {
+        return true;
+    }
+
+    AString oldMime, newMime;
+    if (!mFormat->findString("mime", &oldMime)
+            || !targetFormat->findString("mime", &newMime)
+            || !(oldMime == newMime)) {
+        return false;
+    }
+
+    bool audio = !strncasecmp(oldMime.c_str(), "audio/", strlen("audio/"));
+    bool seamless;
+    if (audio) {
+        seamless = supportsSeamlessAudioFormatChange(targetFormat);
+    } else {
+        seamless = mCodec != NULL && mCodec->isConfiguredForAdaptivePlayback();
+    }
+
+    ALOGV("%s seamless support for %s", seamless ? "yes" : "no", oldMime.c_str());
+    return seamless;
+}
+
 }  // namespace android
 

media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h

diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
index a876148..78ea74a 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
@@ -36,6 +36,8 @@
     void signalResume();
     void initiateShutdown();
 
+    bool supportsSeamlessFormatChange(const sp<AMessage> &to) const;
+
 protected:
     virtual ~Decoder();
 
@@ -49,6 +51,7 @@
     sp<AMessage> mNotify;
     sp<NativeWindowWrapper> mNativeWindow;
 
+    sp<AMessage> mFormat;
     sp<ACodec> mCodec;
     sp<ALooper> mCodecLooper;
 
@@ -59,6 +62,8 @@
 
     void onFillThisBuffer(const sp<AMessage> &msg);
 
+    bool supportsSeamlessAudioFormatChange(const sp<AMessage> &targetFormat) const;
+
     DISALLOW_EVIL_CONSTRUCTORS(Decoder);
 };
 

media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp

diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
index 47834fd..e4850f0 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
@@ -16,6 +16,7 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "NuPlayerDriver"
+#include <inttypes.h>
 #include <utils/Log.h>
 
 #include "NuPlayerDriver.h"
@@ -70,7 +71,9 @@
 }
 
 status_t NuPlayerDriver::setDataSource(
-        const char *url, const KeyedVector<String8, String8> *headers) {
+        const sp<IMediaHTTPService> &httpService,
+        const char *url,
+        const KeyedVector<String8, String8> *headers) {
     Mutex::Autolock autoLock(mLock);
 
     if (mState != STATE_IDLE) {
@@ -79,7 +82,7 @@
 
     mState = STATE_SET_DATASOURCE_PENDING;
 
-    mPlayer->setDataSourceAsync(url, headers);
+    mPlayer->setDataSourceAsync(httpService, url, headers);
 
     while (mState == STATE_SET_DATASOURCE_PENDING) {
         mCondition.wait(mLock);
@@ -364,7 +367,7 @@
     return OK;
 }
 
-status_t NuPlayerDriver::setLooping(int loop) {
+status_t NuPlayerDriver::setLooping(int /* loop */) {
     return INVALID_OPERATION;
 }
 
@@ -420,16 +423,17 @@
     mPlayer->setAudioSink(audioSink);
 }
 
-status_t NuPlayerDriver::setParameter(int key, const Parcel &request) {
+status_t NuPlayerDriver::setParameter(
+        int /* key */, const Parcel & /* request */) {
     return INVALID_OPERATION;
 }
 
-status_t NuPlayerDriver::getParameter(int key, Parcel *reply) {
+status_t NuPlayerDriver::getParameter(int /* key */, Parcel * /* reply */) {
     return INVALID_OPERATION;
 }
 
 status_t NuPlayerDriver::getMetadata(
-        const media::Metadata::Filter& ids, Parcel *records) {
+        const media::Metadata::Filter& /* ids */, Parcel *records) {
     Mutex::Autolock autoLock(mLock);
 
     using media::Metadata;
@@ -493,13 +497,14 @@
     mNumFramesDropped = numFramesDropped;
 }
 
-status_t NuPlayerDriver::dump(int fd, const Vector<String16> &args) const {
+status_t NuPlayerDriver::dump(
+        int fd, const Vector<String16> & /* args */) const {
     Mutex::Autolock autoLock(mLock);
 
     FILE *out = fdopen(dup(fd), "w");
 
     fprintf(out, " NuPlayer\n");
-    fprintf(out, "  numFramesTotal(%lld), numFramesDropped(%lld), "
+    fprintf(out, "  numFramesTotal(%" PRId64 "), numFramesDropped(%" PRId64 "), "
                  "percentageDropped(%.2f)\n",
                  mNumFramesTotal,
                  mNumFramesDropped,

media/libmediaplayerservice/nuplayer/NuPlayerDriver.h

diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h
index 99f72a6..0148fb1 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h
@@ -31,7 +31,9 @@
     virtual status_t setUID(uid_t uid);
 
     virtual status_t setDataSource(
-            const char *url, const KeyedVector<String8, String8> *headers);
+            const sp<IMediaHTTPService> &httpService,
+            const char *url,
+            const KeyedVector<String8, String8> *headers);
 
     virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
 

media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp

diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index 3b2784b..bf5271e 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -20,6 +20,8 @@
 
 #include "NuPlayerRenderer.h"
 
+#include "SoftwareRenderer.h"
+
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AMessage.h>
@@ -34,6 +36,7 @@
         const sp<AMessage> &notify,
         uint32_t flags)
     : mAudioSink(sink),
+      mSoftRenderer(NULL),
       mNotify(notify),
       mFlags(flags),
       mNumFramesWritten(0),
@@ -57,6 +60,12 @@
 }
 
 NuPlayer::Renderer::~Renderer() {
+    delete mSoftRenderer;
+}
+
+void NuPlayer::Renderer::setSoftRenderer(SoftwareRenderer *softRenderer) {
+    delete mSoftRenderer;
+    mSoftRenderer = softRenderer;
 }
 
 void NuPlayer::Renderer::queueBuffer(
@@ -413,7 +422,12 @@
         ALOGV("video late by %lld us (%.2f secs)",
              mVideoLateByUs, mVideoLateByUs / 1E6);
     } else {
-        ALOGV("rendering video at media time %.2f secs", mediaTimeUs / 1E6);
+        ALOGV("rendering video at media time %.2f secs",
+                (mFlags & FLAG_REAL_TIME ? realTimeUs :
+                (realTimeUs + mAnchorTimeMediaUs - mAnchorTimeRealUs)) / 1E6);
+        if (mSoftRenderer != NULL) {
+            mSoftRenderer->render(entry->mBuffer->data(), entry->mBuffer->size(), NULL);
+        }
     }
 
     entry->mNotifyConsumed->setInt32("render", !tooLate);

media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h

diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h
index 94a05ea..9124e03 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h
@@ -23,6 +23,7 @@
 namespace android {
 
 struct ABuffer;
+class SoftwareRenderer;
 
 struct NuPlayer::Renderer : public AHandler {
     enum Flags {
@@ -56,6 +57,8 @@
         kWhatMediaRenderingStart = 'mdrd',
     };
 
+    void setSoftRenderer(SoftwareRenderer *softRenderer);
+
 protected:
     virtual ~Renderer();
 
@@ -83,6 +86,7 @@
     static const int64_t kMinPositionUpdateDelayUs;
 
     sp<MediaPlayerBase::AudioSink> mAudioSink;
+    SoftwareRenderer *mSoftRenderer;
     sp<AMessage> mNotify;
     uint32_t mFlags;
     List<QueueEntry> mAudioQueue;

media/libmediaplayerservice/nuplayer/NuPlayerSource.h

diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerSource.h b/media/libmediaplayerservice/nuplayer/NuPlayerSource.h
index e50533a..11279fc 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerSource.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerSource.h
@@ -68,19 +68,19 @@
     virtual status_t dequeueAccessUnit(
             bool audio, sp<ABuffer> *accessUnit) = 0;
 
-    virtual status_t getDuration(int64_t *durationUs) {
+    virtual status_t getDuration(int64_t * /* durationUs */) {
         return INVALID_OPERATION;
     }
 
-    virtual status_t getTrackInfo(Parcel* reply) const {
+    virtual status_t getTrackInfo(Parcel* /* reply */) const {
         return INVALID_OPERATION;
     }
 
-    virtual status_t selectTrack(size_t trackIndex, bool select) {
+    virtual status_t selectTrack(size_t /* trackIndex */, bool /* select */) {
         return INVALID_OPERATION;
     }
 
-    virtual status_t seekTo(int64_t seekTimeUs) {
+    virtual status_t seekTo(int64_t /* seekTimeUs */) {
         return INVALID_OPERATION;
     }
 
@@ -93,7 +93,7 @@
 
     virtual void onMessageReceived(const sp<AMessage> &msg);
 
-    virtual sp<MetaData> getFormatMeta(bool audio) { return NULL; }
+    virtual sp<MetaData> getFormatMeta(bool /* audio */) { return NULL; }
 
     sp<AMessage> dupNotify() const { return mNotify->dup(); }
 

media/libmediaplayerservice/nuplayer/RTSPSource.cpp

diff --git a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
index 18cf6d1..94800ba 100644
--- a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
@@ -24,6 +24,7 @@
 #include "MyHandler.h"
 #include "SDPLoader.h"
 
+#include <media/IMediaHTTPService.h>
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/MetaData.h>
 
@@ -33,12 +34,14 @@
 
 NuPlayer::RTSPSource::RTSPSource(
         const sp<AMessage> &notify,
+        const sp<IMediaHTTPService> &httpService,
         const char *url,
         const KeyedVector<String8, String8> *headers,
         bool uidValid,
         uid_t uid,
         bool isSDP)
     : Source(notify),
+      mHTTPService(httpService),
       mURL(url),
       mUIDValid(uidValid),
       mUID(uid),
@@ -92,7 +95,7 @@
     if (mIsSDP) {
         mSDPLoader = new SDPLoader(notify,
                 (mFlags & kFlagIncognito) ? SDPLoader::kFlagIncognito : 0,
-                mUIDValid, mUID);
+                mHTTPService);
 
         mSDPLoader->load(
                 mURL.c_str(), mExtraHeaders.isEmpty() ? NULL : &mExtraHeaders);

media/libmediaplayerservice/nuplayer/RTSPSource.h

diff --git a/media/libmediaplayerservice/nuplayer/RTSPSource.h b/media/libmediaplayerservice/nuplayer/RTSPSource.h
index 8cf34a0..3718bf9 100644
--- a/media/libmediaplayerservice/nuplayer/RTSPSource.h
+++ b/media/libmediaplayerservice/nuplayer/RTSPSource.h
@@ -34,6 +34,7 @@
 struct NuPlayer::RTSPSource : public NuPlayer::Source {
     RTSPSource(
             const sp<AMessage> &notify,
+            const sp<IMediaHTTPService> &httpService,
             const char *url,
             const KeyedVector<String8, String8> *headers,
             bool uidValid = false,
@@ -88,6 +89,7 @@
         bool mNPTMappingValid;
     };
 
+    sp<IMediaHTTPService> mHTTPService;
     AString mURL;
     KeyedVector<String8, String8> mExtraHeaders;
     bool mUIDValid;

media/libmediaplayerservice/nuplayer/mp4/MP4Source.cpp

diff --git a/media/libmediaplayerservice/nuplayer/mp4/MP4Source.cpp b/media/libmediaplayerservice/nuplayer/mp4/MP4Source.cpp
deleted file mode 100644
index d31d947..0000000
--- a/media/libmediaplayerservice/nuplayer/mp4/MP4Source.cpp
+++ /dev/null
@@ -1,144 +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 "MP4Source.h"
-
-#include "FragmentedMP4Parser.h"
-#include "../NuPlayerStreamListener.h"
-
-#include <media/IStreamSource.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/MetaData.h>
-
-namespace android {
-
-struct StreamSource : public FragmentedMP4Parser::Source {
-    StreamSource(const sp<IStreamSource> &source)
-        : mListener(new NuPlayer::NuPlayerStreamListener(source, 0)),
-          mPosition(0) {
-        mListener->start();
-    }
-
-    virtual ssize_t readAt(off64_t offset, void *data, size_t size) {
-        if (offset < mPosition) {
-            return -EPIPE;
-        }
-
-        while (offset > mPosition) {
-            char buffer[1024];
-            off64_t skipBytes = offset - mPosition;
-            if (skipBytes > sizeof(buffer)) {
-                skipBytes = sizeof(buffer);
-            }
-
-            sp<AMessage> extra;
-            ssize_t n;
-            for (;;) {
-                n = mListener->read(buffer, skipBytes, &extra);
-
-                if (n == -EWOULDBLOCK) {
-                    usleep(10000);
-                    continue;
-                }
-
-                break;
-            }
-
-            ALOGV("skipped %ld bytes at offset %lld", n, mPosition);
-
-            if (n < 0) {
-                return n;
-            }
-
-            mPosition += n;
-        }
-
-        sp<AMessage> extra;
-        size_t total = 0;
-        while (total < size) {
-            ssize_t n = mListener->read(
-                    (uint8_t *)data + total, size - total, &extra);
-
-            if (n == -EWOULDBLOCK) {
-                usleep(10000);
-                continue;
-            } else if (n == 0) {
-                break;
-            } else if (n < 0) {
-                mPosition += total;
-                return n;
-            }
-
-            total += n;
-        }
-
-        ALOGV("read %ld bytes at offset %lld", n, mPosition);
-
-        mPosition += total;
-
-        return total;
-    }
-
-    bool isSeekable() {
-        return false;
-    }
-
-private:
-    sp<NuPlayer::NuPlayerStreamListener> mListener;
-    off64_t mPosition;
-
-    DISALLOW_EVIL_CONSTRUCTORS(StreamSource);
-};
-
-MP4Source::MP4Source(
-        const sp<AMessage> &notify, const sp<IStreamSource> &source)
-    : Source(notify),
-      mSource(source),
-      mLooper(new ALooper),
-      mParser(new FragmentedMP4Parser),
-      mEOS(false) {
-    mLooper->registerHandler(mParser);
-}
-
-MP4Source::~MP4Source() {
-}
-
-void MP4Source::prepareAsync() {
-    notifyVideoSizeChanged(0, 0);
-    notifyFlagsChanged(0);
-    notifyPrepared();
-}
-
-void MP4Source::start() {
-    mLooper->start(false /* runOnCallingThread */);
-    mParser->start(new StreamSource(mSource));
-}
-
-status_t MP4Source::feedMoreTSData() {
-    return mEOS ? ERROR_END_OF_STREAM : (status_t)OK;
-}
-
-sp<AMessage> MP4Source::getFormat(bool audio) {
-    return mParser->getFormat(audio);
-}
-
-status_t MP4Source::dequeueAccessUnit(
-        bool audio, sp<ABuffer> *accessUnit) {
-    return mParser->dequeueAccessUnit(audio, accessUnit);
-}
-
-}  // namespace android

media/libmediaplayerservice/nuplayer/mp4/MP4Source.h

diff --git a/media/libmediaplayerservice/nuplayer/mp4/MP4Source.h b/media/libmediaplayerservice/nuplayer/mp4/MP4Source.h
deleted file mode 100644
index a6ef622..0000000
--- a/media/libmediaplayerservice/nuplayer/mp4/MP4Source.h
+++ /dev/null
@@ -1,53 +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 MP4_SOURCE_H
-#define MP4_SOURCE_H
-
-#include "NuPlayerSource.h"
-
-namespace android {
-
-struct FragmentedMP4Parser;
-
-struct MP4Source : public NuPlayer::Source {
-    MP4Source(const sp<AMessage> &notify, const sp<IStreamSource> &source);
-
-    virtual void prepareAsync();
-    virtual void start();
-
-    virtual status_t feedMoreTSData();
-
-    virtual sp<AMessage> getFormat(bool audio);
-
-    virtual status_t dequeueAccessUnit(
-            bool audio, sp<ABuffer> *accessUnit);
-
-protected:
-    virtual ~MP4Source();
-
-private:
-    sp<IStreamSource> mSource;
-    sp<ALooper> mLooper;
-    sp<FragmentedMP4Parser> mParser;
-    bool mEOS;
-
-    DISALLOW_EVIL_CONSTRUCTORS(MP4Source);
-};
-
-}  // namespace android
-
-#endif // MP4_SOURCE_H

media/libnbaio/Android.mk

diff --git a/media/libnbaio/Android.mk b/media/libnbaio/Android.mk
index 69c75b8..9707c4a 100644
--- a/media/libnbaio/Android.mk
+++ b/media/libnbaio/Android.mk
@@ -31,9 +31,8 @@
     libcommon_time_client \
     libcutils \
     libutils \
-    liblog \
-    libmedia
-# This dependency on libmedia is for SingleStateQueueInstantiations.
-# Consider a separate a library for SingleStateQueueInstantiations.
+    liblog
+
+LOCAL_STATIC_LIBRARIES += libinstantssq
 
 include $(BUILD_SHARED_LIBRARY)

media/libnbaio/AudioBufferProviderSource.cpp

diff --git a/media/libnbaio/AudioBufferProviderSource.cpp b/media/libnbaio/AudioBufferProviderSource.cpp
index 74a6fdb..551f516 100644
--- a/media/libnbaio/AudioBufferProviderSource.cpp
+++ b/media/libnbaio/AudioBufferProviderSource.cpp
@@ -24,11 +24,11 @@
 namespace android {
 
 AudioBufferProviderSource::AudioBufferProviderSource(AudioBufferProvider *provider,
-                                                     NBAIO_Format format) :
+                                                     const NBAIO_Format& format) :
     NBAIO_Source(format), mProvider(provider), mConsumed(0)
 {
     ALOG_ASSERT(provider != NULL);
-    ALOG_ASSERT(format != Format_Invalid);
+    ALOG_ASSERT(Format_isValid(format));
 }
 
 AudioBufferProviderSource::~AudioBufferProviderSource()
@@ -68,7 +68,7 @@
     }
     // count could be zero, either because count was zero on entry or
     // available is zero, but both are unlikely so don't check for that
-    memcpy(buffer, (char *) mBuffer.raw + (mConsumed << mBitShift), count << mBitShift);
+    memcpy(buffer, (char *) mBuffer.raw + (mConsumed * mFrameSize), count * mFrameSize);
     if (CC_UNLIKELY((mConsumed += count) >= mBuffer.frameCount)) {
         mProvider->releaseBuffer(&mBuffer);
         mBuffer.raw = NULL;
@@ -120,7 +120,7 @@
             count = available;
         }
         if (CC_LIKELY(count > 0)) {
-            char* readTgt = (char *) mBuffer.raw + (mConsumed << mBitShift);
+            char* readTgt = (char *) mBuffer.raw + (mConsumed * mFrameSize);
             ssize_t ret = via(user, readTgt, count, readPTS);
             if (CC_UNLIKELY(ret <= 0)) {
                 if (CC_LIKELY(accumulator > 0)) {

media/libnbaio/AudioStreamInSource.cpp

diff --git a/media/libnbaio/AudioStreamInSource.cpp b/media/libnbaio/AudioStreamInSource.cpp
index 05273f6..efbc1a7 100644
--- a/media/libnbaio/AudioStreamInSource.cpp
+++ b/media/libnbaio/AudioStreamInSource.cpp
@@ -40,7 +40,7 @@
 ssize_t AudioStreamInSource::negotiate(const NBAIO_Format offers[], size_t numOffers,
                                       NBAIO_Format counterOffers[], size_t& numCounterOffers)
 {
-    if (mFormat == Format_Invalid) {
+    if (!Format_isValid(mFormat)) {
         mStreamBufferSizeBytes = mStream->common.get_buffer_size(&mStream->common);
         audio_format_t streamFormat = mStream->common.get_format(&mStream->common);
         if (streamFormat == AUDIO_FORMAT_PCM_16_BIT) {
@@ -48,7 +48,7 @@
             audio_channel_mask_t channelMask =
                     (audio_channel_mask_t) mStream->common.get_channels(&mStream->common);
             mFormat = Format_from_SR_C(sampleRate, popcount(channelMask));
-            mBitShift = Format_frameBitShift(mFormat);
+            mFrameSize = Format_frameSize(mFormat);
         }
     }
     return NBAIO_Source::negotiate(offers, numOffers, counterOffers, numCounterOffers);
@@ -67,12 +67,12 @@
 
 ssize_t AudioStreamInSource::read(void *buffer, size_t count)
 {
-    if (CC_UNLIKELY(mFormat == Format_Invalid)) {
+    if (CC_UNLIKELY(!Format_isValid(mFormat))) {
         return NEGOTIATE;
     }
-    ssize_t bytesRead = mStream->read(mStream, buffer, count << mBitShift);
+    ssize_t bytesRead = mStream->read(mStream, buffer, count * mFrameSize);
     if (bytesRead > 0) {
-        size_t framesRead = bytesRead >> mBitShift;
+        size_t framesRead = bytesRead / mFrameSize;
         mFramesRead += framesRead;
         return framesRead;
     } else {

media/libnbaio/AudioStreamOutSink.cpp

diff --git a/media/libnbaio/AudioStreamOutSink.cpp b/media/libnbaio/AudioStreamOutSink.cpp
index e4341d7..f7f764e 100644
--- a/media/libnbaio/AudioStreamOutSink.cpp
+++ b/media/libnbaio/AudioStreamOutSink.cpp
@@ -37,7 +37,7 @@
 ssize_t AudioStreamOutSink::negotiate(const NBAIO_Format offers[], size_t numOffers,
                                       NBAIO_Format counterOffers[], size_t& numCounterOffers)
 {
-    if (mFormat == Format_Invalid) {
+    if (!Format_isValid(mFormat)) {
         mStreamBufferSizeBytes = mStream->common.get_buffer_size(&mStream->common);
         audio_format_t streamFormat = mStream->common.get_format(&mStream->common);
         if (streamFormat == AUDIO_FORMAT_PCM_16_BIT) {
@@ -45,7 +45,7 @@
             audio_channel_mask_t channelMask =
                     (audio_channel_mask_t) mStream->common.get_channels(&mStream->common);
             mFormat = Format_from_SR_C(sampleRate, popcount(channelMask));
-            mBitShift = Format_frameBitShift(mFormat);
+            mFrameSize = Format_frameSize(mFormat);
         }
     }
     return NBAIO_Sink::negotiate(offers, numOffers, counterOffers, numCounterOffers);
@@ -56,10 +56,10 @@
     if (!mNegotiated) {
         return NEGOTIATE;
     }
-    ALOG_ASSERT(mFormat != Format_Invalid);
-    ssize_t ret = mStream->write(mStream, buffer, count << mBitShift);
+    ALOG_ASSERT(Format_isValid(mFormat));
+    ssize_t ret = mStream->write(mStream, buffer, count * mFrameSize);
     if (ret > 0) {
-        ret >>= mBitShift;
+        ret /= mFrameSize;
         mFramesWritten += ret;
     } else {
         // FIXME verify HAL implementations are returning the correct error codes e.g. WOULD_BLOCK

media/libnbaio/MonoPipe.cpp

diff --git a/media/libnbaio/MonoPipe.cpp b/media/libnbaio/MonoPipe.cpp
index de0ad28..9c8461c 100644
--- a/media/libnbaio/MonoPipe.cpp
+++ b/media/libnbaio/MonoPipe.cpp
@@ -30,7 +30,7 @@
 
 namespace android {
 
-MonoPipe::MonoPipe(size_t reqFrames, NBAIO_Format format, bool writeCanBlock) :
+MonoPipe::MonoPipe(size_t reqFrames, const NBAIO_Format& format, bool writeCanBlock) :
         NBAIO_Sink(format),
         mUpdateSeq(0),
         mReqFrames(reqFrames),
@@ -115,11 +115,11 @@
             part1 = written;
         }
         if (CC_LIKELY(part1 > 0)) {
-            memcpy((char *) mBuffer + (rear << mBitShift), buffer, part1 << mBitShift);
+            memcpy((char *) mBuffer + (rear * mFrameSize), buffer, part1 * mFrameSize);
             if (CC_UNLIKELY(rear + part1 == mMaxFrames)) {
                 size_t part2 = written - part1;
                 if (CC_LIKELY(part2 > 0)) {
-                    memcpy(mBuffer, (char *) buffer + (part1 << mBitShift), part2 << mBitShift);
+                    memcpy(mBuffer, (char *) buffer + (part1 * mFrameSize), part2 * mFrameSize);
                 }
             }
             android_atomic_release_store(written + mRear, &mRear);
@@ -129,7 +129,7 @@
             break;
         }
         count -= written;
-        buffer = (char *) buffer + (written << mBitShift);
+        buffer = (char *) buffer + (written * mFrameSize);
         // Simulate blocking I/O by sleeping at different rates, depending on a throttle.
         // The throttle tries to keep the mean pipe depth near the setpoint, with a slight jitter.
         uint32_t ns;
@@ -183,7 +183,7 @@
             }
         }
         if (ns > 0) {
-            const struct timespec req = {0, ns};
+            const struct timespec req = {0, static_cast<long>(ns)};
             nanosleep(&req, NULL);
         }
         // record the time that this write() completed

media/libnbaio/MonoPipeReader.cpp

diff --git a/media/libnbaio/MonoPipeReader.cpp b/media/libnbaio/MonoPipeReader.cpp
index 851341a..de82229 100644
--- a/media/libnbaio/MonoPipeReader.cpp
+++ b/media/libnbaio/MonoPipeReader.cpp
@@ -73,11 +73,11 @@
         part1 = red;
     }
     if (CC_LIKELY(part1 > 0)) {
-        memcpy(buffer, (char *) mPipe->mBuffer + (front << mBitShift), part1 << mBitShift);
+        memcpy(buffer, (char *) mPipe->mBuffer + (front * mFrameSize), part1 * mFrameSize);
         if (CC_UNLIKELY(front + part1 == mPipe->mMaxFrames)) {
             size_t part2 = red - part1;
             if (CC_LIKELY(part2 > 0)) {
-                memcpy((char *) buffer + (part1 << mBitShift), mPipe->mBuffer, part2 << mBitShift);
+                memcpy((char *) buffer + (part1 * mFrameSize), mPipe->mBuffer, part2 * mFrameSize);
             }
         }
         mPipe->updateFrontAndNRPTS(red + mPipe->mFront, nextReadPTS);

media/libnbaio/NBAIO.cpp

diff --git a/media/libnbaio/NBAIO.cpp b/media/libnbaio/NBAIO.cpp
index e0d2c21..4f6591d 100644
--- a/media/libnbaio/NBAIO.cpp
+++ b/media/libnbaio/NBAIO.cpp
@@ -22,16 +22,22 @@
 
 namespace android {
 
-size_t Format_frameSize(NBAIO_Format format)
+size_t Format_frameSize(const NBAIO_Format& format)
 {
-    return Format_channelCount(format) * sizeof(short);
+    return format.mFrameSize;
 }
 
-size_t Format_frameBitShift(NBAIO_Format format)
+#if 0
+int Format_frameBitShift(const NBAIO_Format& format)
 {
+    // FIXME The sample format is hard-coded to AUDIO_FORMAT_PCM_16_BIT
     // sizeof(short) == 2, so frame size == 1 << channels
     return Format_channelCount(format);
+    // FIXME must return -1 for non-power of 2
 }
+#endif
+
+const NBAIO_Format Format_Invalid = { 0, 0, AUDIO_FORMAT_INVALID, 0 };
 
 enum {
     Format_SR_8000,
@@ -51,90 +57,35 @@
     Format_C_Mask = 0x18
 };
 
-unsigned Format_sampleRate(NBAIO_Format format)
+unsigned Format_sampleRate(const NBAIO_Format& format)
 {
-    if (format == Format_Invalid) {
+    if (!Format_isValid(format)) {
         return 0;
     }
-    switch (format & Format_SR_Mask) {
-    case Format_SR_8000:
-        return 8000;
-    case Format_SR_11025:
-        return 11025;
-    case Format_SR_16000:
-        return 16000;
-    case Format_SR_22050:
-        return 22050;
-    case Format_SR_24000:
-        return 24000;
-    case Format_SR_32000:
-        return 32000;
-    case Format_SR_44100:
-        return 44100;
-    case Format_SR_48000:
-        return 48000;
-    default:
-        return 0;
-    }
+    return format.mSampleRate;
 }
 
-unsigned Format_channelCount(NBAIO_Format format)
+unsigned Format_channelCount(const NBAIO_Format& format)
 {
-    if (format == Format_Invalid) {
+    if (!Format_isValid(format)) {
         return 0;
     }
-    switch (format & Format_C_Mask) {
-    case Format_C_1:
-        return 1;
-    case Format_C_2:
-        return 2;
-    default:
-        return 0;
-    }
+    return format.mChannelCount;
 }
 
-NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount)
+NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount,
+        audio_format_t format)
 {
-    NBAIO_Format format;
-    switch (sampleRate) {
-    case 8000:
-        format = Format_SR_8000;
-        break;
-    case 11025:
-        format = Format_SR_11025;
-        break;
-    case 16000:
-        format = Format_SR_16000;
-        break;
-    case 22050:
-        format = Format_SR_22050;
-        break;
-    case 24000:
-        format = Format_SR_24000;
-        break;
-    case 32000:
-        format = Format_SR_32000;
-        break;
-    case 44100:
-        format = Format_SR_44100;
-        break;
-    case 48000:
-        format = Format_SR_48000;
-        break;
-    default:
+    if (sampleRate == 0 || channelCount == 0 || !audio_is_valid_format(format)) {
         return Format_Invalid;
     }
-    switch (channelCount) {
-    case 1:
-        format |= Format_C_1;
-        break;
-    case 2:
-        format |= Format_C_2;
-        break;
-    default:
-        return Format_Invalid;
-    }
-    return format;
+    NBAIO_Format ret;
+    ret.mSampleRate = sampleRate;
+    ret.mChannelCount = channelCount;
+    ret.mFormat = format;
+    ret.mFrameSize = audio_is_linear_pcm(format) ?
+            channelCount * audio_bytes_per_sample(format) : sizeof(uint8_t);
+    return ret;
 }
 
 // This is a default implementation; it is expected that subclasses will optimize this.
@@ -216,9 +167,9 @@
 {
     ALOGV("negotiate offers=%p numOffers=%u countersOffers=%p numCounterOffers=%u",
             offers, numOffers, counterOffers, numCounterOffers);
-    if (mFormat != Format_Invalid) {
+    if (Format_isValid(mFormat)) {
         for (size_t i = 0; i < numOffers; ++i) {
-            if (offers[i] == mFormat) {
+            if (Format_isEqual(offers[i], mFormat)) {
                 mNegotiated = true;
                 return i;
             }
@@ -233,4 +184,17 @@
     return (ssize_t) NEGOTIATE;
 }
 
+bool Format_isValid(const NBAIO_Format& format)
+{
+    return format.mSampleRate != 0 && format.mChannelCount != 0 &&
+            format.mFormat != AUDIO_FORMAT_INVALID && format.mFrameSize != 0;
+}
+
+bool Format_isEqual(const NBAIO_Format& format1, const NBAIO_Format& format2)
+{
+    return format1.mSampleRate == format2.mSampleRate &&
+            format1.mChannelCount == format2.mChannelCount && format1.mFormat == format2.mFormat &&
+            format1.mFrameSize == format2.mFrameSize;
+}
+
 }   // namespace android

media/libnbaio/NBLog.cpp

diff --git a/media/libnbaio/NBLog.cpp b/media/libnbaio/NBLog.cpp
index 045bf64..96738a7 100644
--- a/media/libnbaio/NBLog.cpp
+++ b/media/libnbaio/NBLog.cpp
@@ -26,6 +26,7 @@
 #include <cutils/atomic.h>
 #include <media/nbaio/NBLog.h>
 #include <utils/Log.h>
+#include <utils/String8.h>
 
 namespace android {
 
@@ -337,25 +338,25 @@
         }
         i -= length + 3;
     }
-    if (i > 0) {
-        lost += i;
-        if (fd >= 0) {
-            fdprintf(fd, "%*swarning: lost %u bytes worth of events\n", indent, "", lost);
-        } else {
-            ALOGI("%*swarning: lost %u bytes worth of events\n", indent, "", lost);
-        }
+    mFd = fd;
+    mIndent = indent;
+    String8 timestamp, body;
+    lost += i;
+    if (lost > 0) {
+        body.appendFormat("warning: lost %u bytes worth of events", lost);
+        // TODO timestamp empty here, only other choice to wait for the first timestamp event in the
+        //      log to push it out.  Consider keeping the timestamp/body between calls to readAt().
+        dumpLine(timestamp, body);
     }
     size_t width = 1;
     while (maxSec >= 10) {
         ++width;
         maxSec /= 10;
     }
-    char prefix[32];
     if (maxSec >= 0) {
-        snprintf(prefix, sizeof(prefix), "[%*s] ", width + 4, "");
-    } else {
-        prefix[0] = '\0';
+        timestamp.appendFormat("[%*s]", width + 4, "");
     }
+    bool deferredTimestamp = false;
     while (i < avail) {
         event = (Event) copy[i];
         length = copy[i + 1];
@@ -363,11 +364,8 @@
         size_t advance = length + 3;
         switch (event) {
         case EVENT_STRING:
-            if (fd >= 0) {
-                fdprintf(fd, "%*s%s%.*s\n", indent, "", prefix, length, (const char *) data);
-            } else {
-                ALOGI("%*s%s%.*s", indent, "", prefix, length, (const char *) data);
-            } break;
+            body.appendFormat("%.*s", length, (const char *) data);
+            break;
         case EVENT_TIMESTAMP: {
             // already checked that length == sizeof(struct timespec);
             memcpy(&ts, data, sizeof(struct timespec));
@@ -400,48 +398,56 @@
                 prevNsec = tsNext.tv_nsec;
             }
             size_t n = (j - i) / (sizeof(struct timespec) + 3);
+            if (deferredTimestamp) {
+                dumpLine(timestamp, body);
+                deferredTimestamp = false;
+            }
+            timestamp.clear();
             if (n >= kSquashTimestamp) {
-                if (fd >= 0) {
-                    fdprintf(fd, "%*s[%d.%03d to .%.03d by .%.03d to .%.03d]\n", indent, "",
-                            (int) ts.tv_sec, (int) (ts.tv_nsec / 1000000),
-                            (int) ((ts.tv_nsec + deltaTotal) / 1000000),
-                            (int) (deltaMin / 1000000), (int) (deltaMax / 1000000));
-                } else {
-                    ALOGI("%*s[%d.%03d to .%.03d by .%.03d to .%.03d]\n", indent, "",
-                            (int) ts.tv_sec, (int) (ts.tv_nsec / 1000000),
-                            (int) ((ts.tv_nsec + deltaTotal) / 1000000),
-                            (int) (deltaMin / 1000000), (int) (deltaMax / 1000000));
-                }
+                timestamp.appendFormat("[%d.%03d to .%.03d by .%.03d to .%.03d]",
+                        (int) ts.tv_sec, (int) (ts.tv_nsec / 1000000),
+                        (int) ((ts.tv_nsec + deltaTotal) / 1000000),
+                        (int) (deltaMin / 1000000), (int) (deltaMax / 1000000));
                 i = j;
                 advance = 0;
                 break;
             }
-            if (fd >= 0) {
-                fdprintf(fd, "%*s[%d.%03d]\n", indent, "", (int) ts.tv_sec,
-                        (int) (ts.tv_nsec / 1000000));
-            } else {
-                ALOGI("%*s[%d.%03d]", indent, "", (int) ts.tv_sec,
-                        (int) (ts.tv_nsec / 1000000));
-            }
+            timestamp.appendFormat("[%d.%03d]", (int) ts.tv_sec,
+                    (int) (ts.tv_nsec / 1000000));
+            deferredTimestamp = true;
             } break;
         case EVENT_RESERVED:
         default:
-            if (fd >= 0) {
-                fdprintf(fd, "%*s%swarning: unknown event %d\n", indent, "", prefix, event);
-            } else {
-                ALOGI("%*s%swarning: unknown event %d", indent, "", prefix, event);
-            }
+            body.appendFormat("warning: unknown event %d", event);
             break;
         }
         i += advance;
+
+        if (!body.isEmpty()) {
+            dumpLine(timestamp, body);
+            deferredTimestamp = false;
+        }
+    }
+    if (deferredTimestamp) {
+        dumpLine(timestamp, body);
     }
     // FIXME it would be more efficient to put a char mCopy[256] as a member variable of the dumper
     delete[] copy;
 }
 
+void NBLog::Reader::dumpLine(const String8& timestamp, String8& body)
+{
+    if (mFd >= 0) {
+        fdprintf(mFd, "%.*s%s %s\n", mIndent, "", timestamp.string(), body.string());
+    } else {
+        ALOGI("%.*s%s %s", mIndent, "", timestamp.string(), body.string());
+    }
+    body.clear();
+}
+
 bool NBLog::Reader::isIMemory(const sp<IMemory>& iMemory) const
 {
-    return iMemory.get() == mIMemory.get();
+    return iMemory != 0 && mIMemory != 0 && iMemory->pointer() == mIMemory->pointer();
 }
 
 }   // namespace android

media/libnbaio/Pipe.cpp

diff --git a/media/libnbaio/Pipe.cpp b/media/libnbaio/Pipe.cpp
index 1c21f9c..28a034c 100644
--- a/media/libnbaio/Pipe.cpp
+++ b/media/libnbaio/Pipe.cpp
@@ -25,7 +25,7 @@
 
 namespace android {
 
-Pipe::Pipe(size_t maxFrames, NBAIO_Format format) :
+Pipe::Pipe(size_t maxFrames, const NBAIO_Format& format) :
         NBAIO_Sink(format),
         mMaxFrames(roundup(maxFrames)),
         mBuffer(malloc(mMaxFrames * Format_frameSize(format))),
@@ -52,13 +52,13 @@
     if (CC_LIKELY(written > count)) {
         written = count;
     }
-    memcpy((char *) mBuffer + (rear << mBitShift), buffer, written << mBitShift);
+    memcpy((char *) mBuffer + (rear * mFrameSize), buffer, written * mFrameSize);
     if (CC_UNLIKELY(rear + written == mMaxFrames)) {
         if (CC_UNLIKELY((count -= written) > rear)) {
             count = rear;
         }
         if (CC_LIKELY(count > 0)) {
-            memcpy(mBuffer, (char *) buffer + (written << mBitShift), count << mBitShift);
+            memcpy(mBuffer, (char *) buffer + (written * mFrameSize), count * mFrameSize);
             written += count;
         }
     }

media/libnbaio/PipeReader.cpp

diff --git a/media/libnbaio/PipeReader.cpp b/media/libnbaio/PipeReader.cpp
index d786b84..c8e4953 100644
--- a/media/libnbaio/PipeReader.cpp
+++ b/media/libnbaio/PipeReader.cpp
@@ -59,7 +59,7 @@
     return avail;
 }
 
-ssize_t PipeReader::read(void *buffer, size_t count, int64_t readPTS)
+ssize_t PipeReader::read(void *buffer, size_t count, int64_t readPTS __unused)
 {
     ssize_t avail = availableToRead();
     if (CC_UNLIKELY(avail <= 0)) {
@@ -76,14 +76,14 @@
         red = count;
     }
     // In particular, an overrun during the memcpy will result in reading corrupt data
-    memcpy(buffer, (char *) mPipe.mBuffer + (front << mBitShift), red << mBitShift);
+    memcpy(buffer, (char *) mPipe.mBuffer + (front * mFrameSize), red * mFrameSize);
     // We could re-read the rear pointer here to detect the corruption, but why bother?
     if (CC_UNLIKELY(front + red == mPipe.mMaxFrames)) {
         if (CC_UNLIKELY((count -= red) > front)) {
             count = front;
         }
         if (CC_LIKELY(count > 0)) {
-            memcpy((char *) buffer + (red << mBitShift), mPipe.mBuffer, count << mBitShift);
+            memcpy((char *) buffer + (red * mFrameSize), mPipe.mBuffer, count * mFrameSize);
             red += count;
         }
     }

media/libnbaio/SourceAudioBufferProvider.cpp

diff --git a/media/libnbaio/SourceAudioBufferProvider.cpp b/media/libnbaio/SourceAudioBufferProvider.cpp
index 062fa0f..e21ef48 100644
--- a/media/libnbaio/SourceAudioBufferProvider.cpp
+++ b/media/libnbaio/SourceAudioBufferProvider.cpp
@@ -24,7 +24,7 @@
 
 SourceAudioBufferProvider::SourceAudioBufferProvider(const sp<NBAIO_Source>& source) :
     mSource(source),
-    // mFrameBitShiftFormat below
+    // mFrameSize below
     mAllocated(NULL), mSize(0), mOffset(0), mRemaining(0), mGetCount(0), mFramesReleased(0)
 {
     ALOG_ASSERT(source != 0);
@@ -37,7 +37,7 @@
     numCounterOffers = 0;
     index = source->negotiate(counterOffers, 1, NULL, numCounterOffers);
     ALOG_ASSERT(index == 0);
-    mFrameBitShift = Format_frameBitShift(source->format());
+    mFrameSize = Format_frameSize(source->format());
 }
 
 SourceAudioBufferProvider::~SourceAudioBufferProvider()
@@ -54,14 +54,14 @@
         if (mRemaining < buffer->frameCount) {
             buffer->frameCount = mRemaining;
         }
-        buffer->raw = (char *) mAllocated + (mOffset << mFrameBitShift);
+        buffer->raw = (char *) mAllocated + (mOffset * mFrameSize);
         mGetCount = buffer->frameCount;
         return OK;
     }
     // do we need to reallocate?
     if (buffer->frameCount > mSize) {
         free(mAllocated);
-        mAllocated = malloc(buffer->frameCount << mFrameBitShift);
+        mAllocated = malloc(buffer->frameCount * mFrameSize);
         mSize = buffer->frameCount;
     }
     // read from source
@@ -84,7 +84,7 @@
 void SourceAudioBufferProvider::releaseBuffer(Buffer *buffer)
 {
     ALOG_ASSERT((buffer != NULL) &&
-            (buffer->raw == (char *) mAllocated + (mOffset << mFrameBitShift)) &&
+            (buffer->raw == (char *) mAllocated + (mOffset * mFrameSize)) &&
             (buffer->frameCount <= mGetCount) &&
             (mGetCount <= mRemaining) &&
             (mOffset + mRemaining <= mSize));

media/libstagefright/AACExtractor.cpp

diff --git a/media/libstagefright/AACExtractor.cpp b/media/libstagefright/AACExtractor.cpp
index 4d1072f..196f6ee 100644
--- a/media/libstagefright/AACExtractor.cpp
+++ b/media/libstagefright/AACExtractor.cpp
@@ -219,7 +219,7 @@
     return new AACSource(mDataSource, mMeta, mOffsetVector, mFrameDurationUs);
 }
 
-sp<MetaData> AACExtractor::getTrackMetaData(size_t index, uint32_t flags) {
+sp<MetaData> AACExtractor::getTrackMetaData(size_t index, uint32_t /* flags */) {
     if (mInitCheck != OK || index != 0) {
         return NULL;
     }
@@ -252,7 +252,7 @@
     }
 }
 
-status_t AACSource::start(MetaData *params) {
+status_t AACSource::start(MetaData * /* params */) {
     CHECK(!mStarted);
 
     if (mOffsetVector.empty()) {

media/libstagefright/AACWriter.cpp

diff --git a/media/libstagefright/AACWriter.cpp b/media/libstagefright/AACWriter.cpp
index a6f7cfb..deee8e7 100644
--- a/media/libstagefright/AACWriter.cpp
+++ b/media/libstagefright/AACWriter.cpp
@@ -111,7 +111,7 @@
     return OK;
 }
 
-status_t AACWriter::start(MetaData *params) {
+status_t AACWriter::start(MetaData * /* params */) {
     if (mInitCheck != OK) {
         return mInitCheck;
     }
@@ -171,7 +171,7 @@
     void *dummy;
     pthread_join(mThread, &dummy);
 
-    status_t err = (status_t) dummy;
+    status_t err = static_cast<status_t>(reinterpret_cast<uintptr_t>(dummy));
     {
         status_t status = mSource->stop();
         if (err == OK &&
@@ -200,7 +200,7 @@
 
 // static
 void *AACWriter::ThreadWrapper(void *me) {
-    return (void *) static_cast<AACWriter *>(me)->threadFunc();
+    return (void *)(uintptr_t)static_cast<AACWriter *>(me)->threadFunc();
 }
 
 /*

media/libstagefright/ACodec.cpp

diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index bfb730c..9c48587 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -35,7 +35,9 @@
 
 #include <media/hardware/HardwareAPI.h>
 
+#include <OMX_AudioExt.h>
 #include <OMX_Component.h>
+#include <OMX_IndexExt.h>
 
 #include "include/avc_utils.h"
 
@@ -363,6 +365,7 @@
       mIsEncoder(false),
       mUseMetadataOnEncoderOutput(false),
       mShutdownInProgress(false),
+      mIsConfiguredForAdaptivePlayback(false),
       mEncoderDelay(0),
       mEncoderPadding(0),
       mChannelMaskPresent(false),
@@ -370,7 +373,11 @@
       mDequeueCounter(0),
       mStoreMetaDataInOutputBuffers(false),
       mMetaDataBuffersToSubmit(0),
-      mRepeatFrameDelayUs(-1ll) {
+      mRepeatFrameDelayUs(-1ll),
+      mMaxPtsGapUs(-1ll),
+      mTimePerCaptureUs(-1ll),
+      mTimePerFrameUs(-1ll),
+      mCreateInputBuffersSuspended(false) {
     mUninitializedState = new UninitializedState(this);
     mLoadedState = new LoadedState(this);
     mLoadedToIdleState = new LoadedToIdleState(this);
@@ -452,6 +459,18 @@
     (new AMessage(kWhatRequestIDRFrame, id()))->post();
 }
 
+// *** NOTE: THE FOLLOWING WORKAROUND WILL BE REMOVED ***
+// Some codecs may return input buffers before having them processed.
+// This causes a halt if we already signaled an EOS on the input
+// port.  For now keep submitting an output buffer if there was an
+// EOS on the input port, but not yet on the output port.
+void ACodec::signalSubmitOutputMetaDataBufferIfEOS_workaround() {
+    if (mPortEOS[kPortIndexInput] && !mPortEOS[kPortIndexOutput] &&
+            mMetaDataBuffersToSubmit > 0) {
+        (new AMessage(kWhatSubmitOutputMetaDataBufferIfEOS, id()))->post();
+    }
+}
+
 status_t ACodec::allocateBuffersOnPort(OMX_U32 portIndex) {
     CHECK(portIndex == kPortIndexInput || portIndex == kPortIndexOutput);
 
@@ -945,6 +964,8 @@
             "audio_decoder.aac", "audio_encoder.aac" },
         { MEDIA_MIMETYPE_AUDIO_VORBIS,
             "audio_decoder.vorbis", "audio_encoder.vorbis" },
+        { MEDIA_MIMETYPE_AUDIO_OPUS,
+            "audio_decoder.opus", "audio_encoder.opus" },
         { MEDIA_MIMETYPE_AUDIO_G711_MLAW,
             "audio_decoder.g711mlaw", "audio_encoder.g711mlaw" },
         { MEDIA_MIMETYPE_AUDIO_G711_ALAW,
@@ -965,6 +986,10 @@
             "audio_decoder.flac", "audio_encoder.flac" },
         { MEDIA_MIMETYPE_AUDIO_MSGSM,
             "audio_decoder.gsm", "audio_encoder.gsm" },
+        { MEDIA_MIMETYPE_VIDEO_MPEG2,
+            "video_decoder.mpeg2", "video_encoder.mpeg2" },
+        { MEDIA_MIMETYPE_AUDIO_AC3,
+            "audio_decoder.ac3", "audio_encoder.ac3" },
     };
 
     static const size_t kNumMimeToRole =
@@ -1096,6 +1121,20 @@
                     &mRepeatFrameDelayUs)) {
             mRepeatFrameDelayUs = -1ll;
         }
+
+        if (!msg->findInt64("max-pts-gap-to-encoder", &mMaxPtsGapUs)) {
+            mMaxPtsGapUs = -1ll;
+        }
+
+        if (!msg->findInt64("time-lapse", &mTimePerCaptureUs)) {
+            mTimePerCaptureUs = -1ll;
+        }
+
+        if (!msg->findInt32(
+                    "create-input-buffers-suspended",
+                    (int32_t*)&mCreateInputBuffersSuspended)) {
+            mCreateInputBuffersSuspended = false;
+        }
     }
 
     // Always try to enable dynamic output buffers on native surface
@@ -1103,16 +1142,59 @@
     int32_t haveNativeWindow = msg->findObject("native-window", &obj) &&
             obj != NULL;
     mStoreMetaDataInOutputBuffers = false;
+    mIsConfiguredForAdaptivePlayback = false;
     if (!encoder && video && haveNativeWindow) {
         err = mOMX->storeMetaDataInBuffers(mNode, kPortIndexOutput, OMX_TRUE);
         if (err != OK) {
-            // allow failure
             ALOGE("[%s] storeMetaDataInBuffers failed w/ err %d",
                   mComponentName.c_str(), err);
+
+            // if adaptive playback has been requested, try JB fallback
+            // NOTE: THIS FALLBACK MECHANISM WILL BE REMOVED DUE TO ITS
+            // LARGE MEMORY REQUIREMENT
+
+            // we will not do adaptive playback on software accessed
+            // surfaces as they never had to respond to changes in the
+            // crop window, and we don't trust that they will be able to.
+            int usageBits = 0;
+            bool canDoAdaptivePlayback;
+
+            sp<NativeWindowWrapper> windowWrapper(
+                    static_cast<NativeWindowWrapper *>(obj.get()));
+            sp<ANativeWindow> nativeWindow = windowWrapper->getNativeWindow();
+
+            if (nativeWindow->query(
+                    nativeWindow.get(),
+                    NATIVE_WINDOW_CONSUMER_USAGE_BITS,
+                    &usageBits) != OK) {
+                canDoAdaptivePlayback = false;
+            } else {
+                canDoAdaptivePlayback =
+                    (usageBits &
+                            (GRALLOC_USAGE_SW_READ_MASK |
+                             GRALLOC_USAGE_SW_WRITE_MASK)) == 0;
+            }
+
+            int32_t maxWidth = 0, maxHeight = 0;
+            if (canDoAdaptivePlayback &&
+                msg->findInt32("max-width", &maxWidth) &&
+                msg->findInt32("max-height", &maxHeight)) {
+                ALOGV("[%s] prepareForAdaptivePlayback(%dx%d)",
+                      mComponentName.c_str(), maxWidth, maxHeight);
+
+                err = mOMX->prepareForAdaptivePlayback(
+                        mNode, kPortIndexOutput, OMX_TRUE, maxWidth, maxHeight);
+                ALOGW_IF(err != OK,
+                        "[%s] prepareForAdaptivePlayback failed w/ err %d",
+                        mComponentName.c_str(), err);
+                mIsConfiguredForAdaptivePlayback = (err == OK);
+            }
+            // allow failure
             err = OK;
         } else {
             ALOGV("[%s] storeMetaDataInBuffers succeeded", mComponentName.c_str());
             mStoreMetaDataInOutputBuffers = true;
+            mIsConfiguredForAdaptivePlayback = true;
         }
 
         int32_t push;
@@ -1216,6 +1298,15 @@
         } else {
             err = setupRawAudioFormat(kPortIndexInput, sampleRate, numChannels);
         }
+    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AC3)) {
+        int32_t numChannels;
+        int32_t sampleRate;
+        if (!msg->findInt32("channel-count", &numChannels)
+                || !msg->findInt32("sample-rate", &sampleRate)) {
+            err = INVALID_OPERATION;
+        } else {
+            err = setupAC3Codec(encoder, numChannels, sampleRate);
+        }
     }
 
     if (err != OK) {
@@ -1412,6 +1503,44 @@
             mNode, OMX_IndexParamAudioAac, &profile, sizeof(profile));
 }
 
+status_t ACodec::setupAC3Codec(
+        bool encoder, int32_t numChannels, int32_t sampleRate) {
+    status_t err = setupRawAudioFormat(
+            encoder ? kPortIndexInput : kPortIndexOutput, sampleRate, numChannels);
+
+    if (err != OK) {
+        return err;
+    }
+
+    if (encoder) {
+        ALOGW("AC3 encoding is not supported.");
+        return INVALID_OPERATION;
+    }
+
+    OMX_AUDIO_PARAM_ANDROID_AC3TYPE def;
+    InitOMXParams(&def);
+    def.nPortIndex = kPortIndexInput;
+
+    err = mOMX->getParameter(
+            mNode,
+            (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3,
+            &def,
+            sizeof(def));
+
+    if (err != OK) {
+        return err;
+    }
+
+    def.nChannels = numChannels;
+    def.nSampleRate = sampleRate;
+
+    return mOMX->setParameter(
+            mNode,
+            (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3,
+            &def,
+            sizeof(def));
+}
+
 static OMX_AUDIO_AMRBANDMODETYPE pickModeFromBitRate(
         bool isAMRWB, int32_t bps) {
     if (isAMRWB) {
@@ -1795,6 +1924,7 @@
             return INVALID_OPERATION;
         }
         frameRate = (float)tmp;
+        mTimePerFrameUs = (int64_t) (1000000.0f / frameRate);
     }
 
     video_def->xFramerate = (OMX_U32)(frameRate * 65536.0f);
@@ -1869,6 +1999,11 @@
             err = setupAVCEncoderParameters(msg);
             break;
 
+        case OMX_VIDEO_CodingVP8:
+        case OMX_VIDEO_CodingVP9:
+            err = setupVPXEncoderParameters(msg);
+            break;
+
         default:
             break;
     }
@@ -2200,6 +2335,17 @@
     return configureBitrate(bitrate, bitrateMode);
 }
 
+status_t ACodec::setupVPXEncoderParameters(const sp<AMessage> &msg) {
+    int32_t bitrate;
+    if (!msg->findInt32("bitrate", &bitrate)) {
+        return INVALID_OPERATION;
+    }
+
+    OMX_VIDEO_CONTROLRATETYPE bitrateMode = getBitrateMode(msg);
+
+    return configureBitrate(bitrate, bitrateMode);
+}
+
 status_t ACodec::verifySupportForProfileAndLevel(
         int32_t profile, int32_t level) {
     OMX_VIDEO_PARAM_PROFILELEVELTYPE params;
@@ -2490,7 +2636,7 @@
         {
             OMX_AUDIO_PORTDEFINITIONTYPE *audioDef = &def.format.audio;
 
-            switch (audioDef->eEncoding) {
+            switch ((int)audioDef->eEncoding) {
                 case OMX_AUDIO_CodingPCM:
                 {
                     OMX_AUDIO_PARAM_PCMMODETYPE params;
@@ -2596,6 +2742,24 @@
                     break;
                 }
 
+                case OMX_AUDIO_CodingAndroidAC3:
+                {
+                    OMX_AUDIO_PARAM_ANDROID_AC3TYPE params;
+                    InitOMXParams(&params);
+                    params.nPortIndex = kPortIndexOutput;
+
+                    CHECK_EQ((status_t)OK, mOMX->getParameter(
+                            mNode,
+                            (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3,
+                            &params,
+                            sizeof(params)));
+
+                    notify->setString("mime", MEDIA_MIMETYPE_AUDIO_AC3);
+                    notify->setInt32("channel-count", params.nChannels);
+                    notify->setInt32("sample-rate", params.nSampleRate);
+                    break;
+                }
+
                 default:
                     TRESPASS();
             }
@@ -2812,7 +2976,8 @@
       mCodec(codec) {
 }
 
-ACodec::BaseState::PortMode ACodec::BaseState::getPortMode(OMX_U32 portIndex) {
+ACodec::BaseState::PortMode ACodec::BaseState::getPortMode(
+        OMX_U32 /* portIndex */) {
     return KEEP_BUFFERS;
 }
 
@@ -3026,16 +3191,22 @@
     sp<ABuffer> buffer;
     int32_t err = OK;
     bool eos = false;
+    PortMode mode = getPortMode(kPortIndexInput);
 
     if (!msg->findBuffer("buffer", &buffer)) {
+        /* these are unfilled buffers returned by client */
         CHECK(msg->findInt32("err", &err));
 
-        ALOGV("[%s] saw error %d instead of an input buffer",
-             mCodec->mComponentName.c_str(), err);
+        if (err == OK) {
+            /* buffers with no errors are returned on MediaCodec.flush */
+            mode = KEEP_BUFFERS;
+        } else {
+            ALOGV("[%s] saw error %d instead of an input buffer",
+                 mCodec->mComponentName.c_str(), err);
+            eos = true;
+        }
 
         buffer.clear();
-
-        eos = true;
     }
 
     int32_t tmp;
@@ -3049,8 +3220,6 @@
 
     info->mStatus = BufferInfo::OWNED_BY_US;
 
-    PortMode mode = getPortMode(kPortIndexInput);
-
     switch (mode) {
         case KEEP_BUFFERS:
         {
@@ -3175,11 +3344,11 @@
                 mCodec->mInputEOSResult = err;
             }
             break;
-
-            default:
-                CHECK_EQ((int)mode, (int)FREE_BUFFERS);
-                break;
         }
+
+        default:
+            CHECK_EQ((int)mode, (int)FREE_BUFFERS);
+            break;
     }
 }
 
@@ -3217,8 +3386,8 @@
         size_t rangeOffset, size_t rangeLength,
         OMX_U32 flags,
         int64_t timeUs,
-        void *platformPrivate,
-        void *dataPtr) {
+        void * /* platformPrivate */,
+        void * /* dataPtr */) {
     ALOGV("[%s] onOMXFillBufferDone %p time %lld us, flags = 0x%08lx",
          mCodec->mComponentName.c_str(), bufferID, timeUs, flags);
 
@@ -3270,7 +3439,7 @@
             sp<AMessage> reply =
                 new AMessage(kWhatOutputBufferDrained, mCodec->id());
 
-            if (!mCodec->mSentFormat) {
+            if (!mCodec->mSentFormat && rangeLength > 0) {
                 mCodec->sendFormatChange(reply);
             }
 
@@ -3355,7 +3524,7 @@
     int32_t render;
     if (mCodec->mNativeWindow != NULL
             && msg->findInt32("render", &render) && render != 0
-            && (info->mData == NULL || info->mData->size() != 0)) {
+            && info->mData != NULL && info->mData->size() != 0) {
         // The client wants this buffer to be rendered.
 
         status_t err;
@@ -3630,6 +3799,7 @@
     mCodec->mDequeueCounter = 0;
     mCodec->mMetaDataBuffersToSubmit = 0;
     mCodec->mRepeatFrameDelayUs = -1ll;
+    mCodec->mIsConfiguredForAdaptivePlayback = false;
 
     if (mCodec->mShutdownInProgress) {
         bool keepComponentAllocated = mCodec->mKeepComponentAllocated;
@@ -3750,7 +3920,7 @@
 }
 
 void ACodec::LoadedState::onCreateInputSurface(
-        const sp<AMessage> &msg) {
+        const sp<AMessage> & /* msg */) {
     ALOGV("onCreateInputSurface");
 
     sp<AMessage> notify = mCodec->mNotify->dup();
@@ -3778,6 +3948,56 @@
         }
     }
 
+    if (err == OK && mCodec->mMaxPtsGapUs > 0ll) {
+        err = mCodec->mOMX->setInternalOption(
+                mCodec->mNode,
+                kPortIndexInput,
+                IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP,
+                &mCodec->mMaxPtsGapUs,
+                sizeof(mCodec->mMaxPtsGapUs));
+
+        if (err != OK) {
+            ALOGE("[%s] Unable to configure max timestamp gap (err %d)",
+                    mCodec->mComponentName.c_str(),
+                    err);
+        }
+    }
+
+    if (err == OK && mCodec->mTimePerCaptureUs > 0ll
+            && mCodec->mTimePerFrameUs > 0ll) {
+        int64_t timeLapse[2];
+        timeLapse[0] = mCodec->mTimePerFrameUs;
+        timeLapse[1] = mCodec->mTimePerCaptureUs;
+        err = mCodec->mOMX->setInternalOption(
+                mCodec->mNode,
+                kPortIndexInput,
+                IOMX::INTERNAL_OPTION_TIME_LAPSE,
+                &timeLapse[0],
+                sizeof(timeLapse));
+
+        if (err != OK) {
+            ALOGE("[%s] Unable to configure time lapse (err %d)",
+                    mCodec->mComponentName.c_str(),
+                    err);
+        }
+    }
+
+    if (err == OK && mCodec->mCreateInputBuffersSuspended) {
+        bool suspend = true;
+        err = mCodec->mOMX->setInternalOption(
+                mCodec->mNode,
+                kPortIndexInput,
+                IOMX::INTERNAL_OPTION_SUSPEND,
+                &suspend,
+                sizeof(suspend));
+
+        if (err != OK) {
+            ALOGE("[%s] Unable to configure option to suspend (err %d)",
+                  mCodec->mComponentName.c_str(),
+                  err);
+        }
+    }
+
     if (err == OK) {
         notify->setObject("input-surface",
                 new BufferProducerWrapper(bufferProducer));
@@ -3835,6 +4055,7 @@
 
 bool ACodec::LoadedToIdleState::onMessageReceived(const sp<AMessage> &msg) {
     switch (msg->what()) {
+        case kWhatSetParameters:
         case kWhatShutdown:
         {
             mCodec->deferMessage(msg);
@@ -3901,6 +4122,7 @@
 
 bool ACodec::IdleToExecutingState::onMessageReceived(const sp<AMessage> &msg) {
     switch (msg->what()) {
+        case kWhatSetParameters:
         case kWhatShutdown:
         {
             mCodec->deferMessage(msg);
@@ -3961,7 +4183,7 @@
 }
 
 ACodec::BaseState::PortMode ACodec::ExecutingState::getPortMode(
-        OMX_U32 portIndex) {
+        OMX_U32 /* portIndex */) {
     return RESUBMIT_BUFFERS;
 }
 
@@ -3976,6 +4198,9 @@
                 break;
         }
     }
+
+    // *** NOTE: THE FOLLOWING WORKAROUND WILL BE REMOVED ***
+    mCodec->signalSubmitOutputMetaDataBufferIfEOS_workaround();
 }
 
 void ACodec::ExecutingState::submitRegularOutputBuffers() {
@@ -4124,6 +4349,19 @@
             break;
         }
 
+        // *** NOTE: THE FOLLOWING WORKAROUND WILL BE REMOVED ***
+        case kWhatSubmitOutputMetaDataBufferIfEOS:
+        {
+            if (mCodec->mPortEOS[kPortIndexInput] &&
+                    !mCodec->mPortEOS[kPortIndexOutput]) {
+                status_t err = mCodec->submitOutputMetaDataBuffer();
+                if (err == OK) {
+                    mCodec->signalSubmitOutputMetaDataBufferIfEOS_workaround();
+                }
+            }
+            return true;
+        }
+
         default:
             handled = BaseState::onMessageReceived(msg);
             break;
@@ -4154,6 +4392,22 @@
         }
     }
 
+    int64_t skipFramesBeforeUs;
+    if (params->findInt64("skip-frames-before", &skipFramesBeforeUs)) {
+        status_t err =
+            mOMX->setInternalOption(
+                     mNode,
+                     kPortIndexInput,
+                     IOMX::INTERNAL_OPTION_START_TIME,
+                     &skipFramesBeforeUs,
+                     sizeof(skipFramesBeforeUs));
+
+        if (err != OK) {
+            ALOGE("Failed to set parameter 'skip-frames-before' (err %d)", err);
+            return err;
+        }
+    }
+
     int32_t dropInputFrames;
     if (params->findInt32("drop-input-frames", &dropInputFrames)) {
         bool suspend = dropInputFrames != 0;

media/libstagefright/AMRExtractor.cpp

diff --git a/media/libstagefright/AMRExtractor.cpp b/media/libstagefright/AMRExtractor.cpp
index 03dcbf9..3f592ed 100644
--- a/media/libstagefright/AMRExtractor.cpp
+++ b/media/libstagefright/AMRExtractor.cpp
@@ -189,7 +189,7 @@
             mOffsetTable, mOffsetTableLength);
 }
 
-sp<MetaData> AMRExtractor::getTrackMetaData(size_t index, uint32_t flags) {
+sp<MetaData> AMRExtractor::getTrackMetaData(size_t index, uint32_t /* flags */) {
     if (mInitCheck != OK || index != 0) {
         return NULL;
     }
@@ -221,7 +221,7 @@
     }
 }
 
-status_t AMRSource::start(MetaData *params) {
+status_t AMRSource::start(MetaData * /* params */) {
     CHECK(!mStarted);
 
     mOffset = mIsWide ? 9 : 6;
@@ -258,7 +258,7 @@
         int64_t seekFrame = seekTimeUs / 20000ll;  // 20ms per frame.
         mCurrentTimeUs = seekFrame * 20000ll;
 
-        int index = seekFrame / 50;
+        size_t index = seekFrame < 0 ? 0 : seekFrame / 50;
         if (index >= mOffsetTableLength) {
             index = mOffsetTableLength - 1;
         }

media/libstagefright/AMRWriter.cpp

diff --git a/media/libstagefright/AMRWriter.cpp b/media/libstagefright/AMRWriter.cpp
index 8d5eec8..653ca36 100644
--- a/media/libstagefright/AMRWriter.cpp
+++ b/media/libstagefright/AMRWriter.cpp
@@ -105,7 +105,7 @@
     return OK;
 }
 
-status_t AMRWriter::start(MetaData *params) {
+status_t AMRWriter::start(MetaData * /* params */) {
     if (mInitCheck != OK) {
         return mInitCheck;
     }
@@ -162,7 +162,7 @@
     void *dummy;
     pthread_join(mThread, &dummy);
 
-    status_t err = (status_t) dummy;
+    status_t err = static_cast<status_t>(reinterpret_cast<uintptr_t>(dummy));
     {
         status_t status = mSource->stop();
         if (err == OK &&
@@ -191,7 +191,7 @@
 
 // static
 void *AMRWriter::ThreadWrapper(void *me) {
-    return (void *) static_cast<AMRWriter *>(me)->threadFunc();
+    return (void *)(uintptr_t) static_cast<AMRWriter *>(me)->threadFunc();
 }
 
 status_t AMRWriter::threadFunc() {

media/libstagefright/Android.mk

diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index 1f68b51..0fd1e69 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -15,6 +15,7 @@
         CameraSource.cpp                  \
         CameraSourceTimeLapse.cpp         \
         DataSource.cpp                    \
+        DataURISource.cpp                 \
         DRMExtractor.cpp                  \
         ESDS.cpp                          \
         FileSource.cpp                    \
@@ -30,8 +31,10 @@
         MediaBufferGroup.cpp              \
         MediaCodec.cpp                    \
         MediaCodecList.cpp                \
+        MediaCodecSource.cpp              \
         MediaDefs.cpp                     \
         MediaExtractor.cpp                \
+        http/MediaHTTP.cpp                \
         MediaMuxer.cpp                    \
         MediaSource.cpp                   \
         MetaData.cpp                      \
@@ -55,8 +58,6 @@
         WVMExtractor.cpp                  \
         XINGSeeker.cpp                    \
         avc_utils.cpp                     \
-        mp4/FragmentedMP4Parser.cpp       \
-        mp4/TrackFragment.cpp             \
 
 LOCAL_C_INCLUDES:= \
         $(TOP)/frameworks/av/include/media/stagefright/timedtext \
@@ -80,6 +81,7 @@
         libicuuc \
         liblog \
         libmedia \
+        libopus \
         libsonivox \
         libssl \
         libstagefright_omx \
@@ -89,6 +91,7 @@
         libutils \
         libvorbisidec \
         libz \
+        libpowermanager
 
 LOCAL_STATIC_LIBRARIES := \
         libstagefright_color_conversion \
@@ -102,13 +105,6 @@
         libFLAC \
         libmedia_helper
 
-LOCAL_SRC_FILES += \
-        chromium_http_stub.cpp
-LOCAL_CPPFLAGS += -DCHROMIUM_AVAILABLE=1
-
-LOCAL_SHARED_LIBRARIES += libstlport
-include external/stlport/libstlport.mk
-
 LOCAL_SHARED_LIBRARIES += \
         libstagefright_enc_common \
         libstagefright_avc_common \

media/libstagefright/AudioPlayer.cpp

diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index e38e261..8623100 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -363,6 +363,7 @@
     mPositionTimeMediaUs = -1;
     mPositionTimeRealUs = -1;
     mSeeking = false;
+    mSeekTimeUs = 0;
     mReachedEOS = false;
     mFinalStatus = OK;
     mStarted = false;
@@ -409,7 +410,7 @@
 
 // static
 size_t AudioPlayer::AudioSinkCallback(
-        MediaPlayerBase::AudioSink *audioSink,
+        MediaPlayerBase::AudioSink * /* audioSink */,
         void *buffer, size_t size, void *cookie,
         MediaPlayerBase::AudioSink::cb_event_t event) {
     AudioPlayer *me = (AudioPlayer *)cookie;
@@ -602,15 +603,24 @@
 
             // need to adjust the mStartPosUs for offload decoding since parser
             // might not be able to get the exact seek time requested.
-            if (refreshSeekTime && useOffload()) {
-                if (postSeekComplete) {
-                    ALOGV("fillBuffer is going to post SEEK_COMPLETE");
-                    mObserver->postAudioSeekComplete();
-                    postSeekComplete = false;
-                }
+            if (refreshSeekTime) {
+                if (useOffload()) {
+                    if (postSeekComplete) {
+                        ALOGV("fillBuffer is going to post SEEK_COMPLETE");
+                        mObserver->postAudioSeekComplete();
+                        postSeekComplete = false;
+                    }
 
-                mStartPosUs = mPositionTimeMediaUs;
-                ALOGV("adjust seek time to: %.2f", mStartPosUs/ 1E6);
+                    mStartPosUs = mPositionTimeMediaUs;
+                    ALOGV("adjust seek time to: %.2f", mStartPosUs/ 1E6);
+                }
+                // clear seek time with mLock locked and once we have valid mPositionTimeMediaUs
+                // and mPositionTimeRealUs
+                // before clearing mSeekTimeUs check if a new seek request has been received while
+                // we were reading from the source with mLock released.
+                if (!mSeeking) {
+                    mSeekTimeUs = 0;
+                }
             }
 
             if (!useOffload()) {
@@ -711,16 +721,27 @@
     return result + diffUs;
 }
 
-int64_t AudioPlayer::getOutputPlayPositionUs_l() const
+int64_t AudioPlayer::getOutputPlayPositionUs_l()
 {
     uint32_t playedSamples = 0;
+    uint32_t sampleRate;
     if (mAudioSink != NULL) {
         mAudioSink->getPosition(&playedSamples);
+        sampleRate = mAudioSink->getSampleRate();
     } else {
         mAudioTrack->getPosition(&playedSamples);
+        sampleRate = mAudioTrack->getSampleRate();
+    }
+    if (sampleRate != 0) {
+        mSampleRate = sampleRate;
     }
 
-    const int64_t playedUs = (static_cast<int64_t>(playedSamples) * 1000000 ) / mSampleRate;
+    int64_t playedUs;
+    if (mSampleRate != 0) {
+        playedUs = (static_cast<int64_t>(playedSamples) * 1000000 ) / mSampleRate;
+    } else {
+        playedUs = 0;
+    }
 
     // HAL position is relative to the first buffer we sent at mStartPosUs
     const int64_t renderedDuration = mStartPosUs + playedUs;
@@ -741,12 +762,10 @@
         return mPositionTimeRealUs;
     }
 
-    if (mPositionTimeMediaUs < 0 || mPositionTimeRealUs < 0) {
-        if (mSeeking) {
-            return mSeekTimeUs;
-        }
 
-        return 0;
+    if (mPositionTimeMediaUs < 0 || mPositionTimeRealUs < 0) {
+        // mSeekTimeUs is either seek time while seeking or 0 if playback did not start.
+        return mSeekTimeUs;
     }
 
     int64_t realTimeOffset = getRealTimeUsLocked() - mPositionTimeRealUs;

media/libstagefright/AudioSource.cpp

diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index bdd842f..d0e0e8e 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -65,10 +65,10 @@
     if (status == OK) {
         // make sure that the AudioRecord callback never returns more than the maximum
         // buffer size
-        int frameCount = kMaxBufferSize / sizeof(int16_t) / channelCount;
+        uint32_t frameCount = kMaxBufferSize / sizeof(int16_t) / channelCount;
 
         // make sure that the AudioRecord total buffer size is large enough
-        int bufCount = 2;
+        size_t bufCount = 2;
         while ((bufCount * frameCount) < minFrameCount) {
             bufCount++;
         }
@@ -76,10 +76,10 @@
         mRecord = new AudioRecord(
                     inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
                     audio_channel_in_mask_from_count(channelCount),
-                    bufCount * frameCount,
+                    (size_t) (bufCount * frameCount),
                     AudioRecordCallbackFunction,
                     this,
-                    frameCount);
+                    frameCount /*notificationFrames*/);
         mInitCheck = mRecord->initCheck();
     } else {
         mInitCheck = status;
@@ -208,7 +208,7 @@
 }
 
 status_t AudioSource::read(
-        MediaBuffer **out, const ReadOptions *options) {
+        MediaBuffer **out, const ReadOptions * /* options */) {
     Mutex::Autolock autoLock(mLock);
     *out = NULL;
 
@@ -236,10 +236,10 @@
         memset((uint8_t *) buffer->data(), 0, buffer->range_length());
     } else if (elapsedTimeUs < kAutoRampStartUs + kAutoRampDurationUs) {
         int32_t autoRampDurationFrames =
-                    (kAutoRampDurationUs * mSampleRate + 500000LL) / 1000000LL;
+                    ((int64_t)kAutoRampDurationUs * mSampleRate + 500000LL) / 1000000LL; //Need type casting
 
         int32_t autoRampStartFrames =
-                    (kAutoRampStartUs * mSampleRate + 500000LL) / 1000000LL;
+                    ((int64_t)kAutoRampStartUs * mSampleRate + 500000LL) / 1000000LL; //Need type casting
 
         int32_t nFrames = mNumFramesReceived - autoRampStartFrames;
         rampVolume(nFrames, autoRampDurationFrames,
@@ -278,7 +278,7 @@
 
     // Drop retrieved and previously lost audio data.
     if (mNumFramesReceived == 0 && timeUs < mStartTimeUs) {
-        mRecord->getInputFramesLost();
+        (void) mRecord->getInputFramesLost();
         ALOGV("Drop audio data at %lld/%lld us", timeUs, mStartTimeUs);
         return OK;
     }
@@ -308,7 +308,7 @@
     if (numLostBytes > 0) {
         // Loss of audio frames should happen rarely; thus the LOGW should
         // not cause a logging spam
-        ALOGW("Lost audio record data: %d bytes", numLostBytes);
+        ALOGW("Lost audio record data: %zu bytes", numLostBytes);
     }
 
     while (numLostBytes > 0) {

media/libstagefright/AwesomePlayer.cpp

diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp
index 5fbee7e..4bad14b 100644
--- a/media/libstagefright/AwesomePlayer.cpp
+++ b/media/libstagefright/AwesomePlayer.cpp
@@ -19,6 +19,7 @@
 //#define LOG_NDEBUG 0
 #define LOG_TAG "AwesomePlayer"
 #define ATRACE_TAG ATRACE_TAG_VIDEO
+#include <inttypes.h>
 #include <utils/Log.h>
 #include <utils/Trace.h>
 
@@ -34,6 +35,8 @@
 
 #include <binder/IPCThreadState.h>
 #include <binder/IServiceManager.h>
+#include <media/IMediaHTTPConnection.h>
+#include <media/IMediaHTTPService.h>
 #include <media/IMediaPlayerService.h>
 #include <media/stagefright/foundation/hexdump.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -44,6 +47,7 @@
 #include <media/stagefright/MediaBuffer.h>
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/MediaExtractor.h>
+#include <media/stagefright/MediaHTTP.h>
 #include <media/stagefright/MediaSource.h>
 #include <media/stagefright/MetaData.h>
 #include <media/stagefright/OMXCodec.h>
@@ -82,7 +86,7 @@
 protected:
     virtual ~AwesomeEvent() {}
 
-    virtual void fire(TimedEventQueue *queue, int64_t /* now_us */) {
+    virtual void fire(TimedEventQueue * /* queue */, int64_t /* now_us */) {
         (mPlayer->*mMethod)();
     }
 
@@ -259,6 +263,7 @@
 
         mQueue.cancelEvent(mBufferingEvent->eventID());
         mBufferingEventPending = false;
+        mAudioTearDown = false;
     }
 }
 
@@ -275,15 +280,20 @@
 }
 
 status_t AwesomePlayer::setDataSource(
-        const char *uri, const KeyedVector<String8, String8> *headers) {
+        const sp<IMediaHTTPService> &httpService,
+        const char *uri,
+        const KeyedVector<String8, String8> *headers) {
     Mutex::Autolock autoLock(mLock);
-    return setDataSource_l(uri, headers);
+    return setDataSource_l(httpService, uri, headers);
 }
 
 status_t AwesomePlayer::setDataSource_l(
-        const char *uri, const KeyedVector<String8, String8> *headers) {
+        const sp<IMediaHTTPService> &httpService,
+        const char *uri,
+        const KeyedVector<String8, String8> *headers) {
     reset_l();
 
+    mHTTPService = httpService;
     mUri = uri;
 
     if (headers) {
@@ -580,6 +590,7 @@
     mSeekNotificationSent = true;
     mSeekTimeUs = 0;
 
+    mHTTPService.clear();
     mUri.setTo("");
     mUriHeaders.clear();
 
@@ -605,6 +616,9 @@
 
     mWatchForAudioSeekComplete = false;
     mWatchForAudioEOS = false;
+
+    mMediaRenderingStartGeneration = 0;
+    mStartGeneration = 0;
 }
 
 void AwesomePlayer::notifyListener_l(int msg, int ext1, int ext2) {
@@ -721,7 +735,7 @@
 
                 if ((mFlags & PLAYING) && !eos
                         && (cachedDataRemaining < kLowWaterMarkBytes)) {
-                    ALOGI("cache is running low (< %d) , pausing.",
+                    ALOGI("cache is running low (< %zu) , pausing.",
                          kLowWaterMarkBytes);
                     modifyFlags(CACHE_UNDERRUN, SET);
                     pause_l();
@@ -730,12 +744,12 @@
                     notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_START);
                 } else if (eos || cachedDataRemaining > kHighWaterMarkBytes) {
                     if (mFlags & CACHE_UNDERRUN) {
-                        ALOGI("cache has filled up (> %d), resuming.",
+                        ALOGI("cache has filled up (> %zu), resuming.",
                              kHighWaterMarkBytes);
                         modifyFlags(CACHE_UNDERRUN, CLEAR);
                         play_l();
                     } else if (mFlags & PREPARING) {
-                        ALOGV("cache has filled up (> %d), prepare is done",
+                        ALOGV("cache has filled up (> %zu), prepare is done",
                              kHighWaterMarkBytes);
                         finishAsyncPrepare_l();
                     }
@@ -797,7 +811,9 @@
         }
     }
 
-    postBufferingEvent_l();
+    if (mFlags & (PLAYING | PREPARING | CACHE_UNDERRUN)) {
+        postBufferingEvent_l();
+    }
 }
 
 void AwesomePlayer::sendCacheStats() {
@@ -892,6 +908,8 @@
         return OK;
     }
 
+    mMediaRenderingStartGeneration = ++mStartGeneration;
+
     if (!(mFlags & PREPARED)) {
         status_t err = prepare_l();
 
@@ -927,6 +945,9 @@
 
             if ((err != OK) && mOffloadAudio) {
                 ALOGI("play_l() cannot create offload output, fallback to sw decode");
+                int64_t curTimeUs;
+                getPosition(&curTimeUs);
+
                 delete mAudioPlayer;
                 mAudioPlayer = NULL;
                 // if the player was started it will take care of stopping the source when destroyed
@@ -942,6 +963,10 @@
                     if (err != OK) {
                         mAudioSource.clear();
                     } else {
+                        mSeekNotificationSent = true;
+                        if (mExtractorFlags & MediaExtractor::CAN_SEEK) {
+                            seekTo_l(curTimeUs);
+                        }
                         createAudioPlayer_l();
                         err = startAudioPlayer_l(false);
                     }
@@ -993,6 +1018,10 @@
     }
     addBatteryData(params);
 
+    if (isStreamingHTTP()) {
+        postBufferingEvent_l();
+    }
+
     return OK;
 }
 
@@ -1183,8 +1212,7 @@
     setVideoScalingMode_l(mVideoScalingMode);
     if (USE_SURFACE_ALLOC
             && !strncmp(component, "OMX.", 4)
-            && strncmp(component, "OMX.google.", 11)
-            && strcmp(component, "OMX.Nvidia.mpeg2v.decode")) {
+            && strncmp(component, "OMX.google.", 11)) {
         // Hardware decoders avoid the CPU color conversion by decoding
         // directly to ANativeBuffers, so we must use a renderer that
         // just pushes those buffers to the ANativeWindow.
@@ -1211,6 +1239,12 @@
 
 status_t AwesomePlayer::pause_l(bool at_eos) {
     if (!(mFlags & PLAYING)) {
+        if (mAudioTearDown && mAudioTearDownWasPlaying) {
+            ALOGV("pause_l() during teardown and finishSetDataSource_l() mFlags %x" , mFlags);
+            mAudioTearDownWasPlaying = false;
+            notifyListener_l(MEDIA_PAUSED);
+            mMediaRenderingStartGeneration = ++mStartGeneration;
+        }
         return OK;
     }
 
@@ -1405,8 +1439,10 @@
     mSeekTimeUs = timeUs;
     modifyFlags((AT_EOS | AUDIO_AT_EOS | VIDEO_AT_EOS), CLEAR);
 
-    notifyListener_l(MEDIA_PAUSED);
-    mMediaRenderingStartGeneration = ++mStartGeneration;
+    if (mFlags & PLAYING) {
+        notifyListener_l(MEDIA_PAUSED);
+        mMediaRenderingStartGeneration = ++mStartGeneration;
+    }
 
     seekAudioIfNecessary_l();
 
@@ -1456,7 +1492,7 @@
     CHECK(source != NULL);
 
     if (mTextDriver == NULL) {
-        mTextDriver = new TimedTextDriver(mListener);
+        mTextDriver = new TimedTextDriver(mListener, mHTTPService);
     }
 
     mTextDriver->addInBandTextSource(trackIndex, source);
@@ -1473,7 +1509,13 @@
     // This doesn't guarantee that the hardware has a free stream
     // but it avoids us attempting to open (and re-open) an offload
     // stream to hardware that doesn't have the necessary codec
-    mOffloadAudio = canOffloadStream(meta, (mVideoSource != NULL), isStreamingHTTP());
+    audio_stream_type_t streamType = AUDIO_STREAM_MUSIC;
+    if (mAudioSink != NULL) {
+        streamType = mAudioSink->getAudioStreamType();
+    }
+
+    mOffloadAudio = canOffloadStream(meta, (mVideoSource != NULL),
+                                     isStreamingHTTP(), streamType);
 
     if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_RAW)) {
         ALOGV("createAudioPlayer: bypass OMX (raw)");
@@ -1651,6 +1693,16 @@
         return;
     }
 
+    // If we paused, then seeked, then resumed, it is possible that we have
+    // signaled SEEK_COMPLETE at a copmletely different media time than where
+    // we are now resuming.  Signal new position to media time provider.
+    // Cannot signal another SEEK_COMPLETE, as existing clients may not expect
+    // multiple SEEK_COMPLETE responses to a single seek() request.
+    if (mSeekNotificationSent && abs(mSeekTimeUs - videoTimeUs) > 10000) {
+        // notify if we are resuming more than 10ms away from desired seek time
+        notifyListener_l(MEDIA_SKIPPED);
+    }
+
     if (mAudioPlayer != NULL) {
         ALOGV("seeking audio to %lld us (%.2f secs).", videoTimeUs, videoTimeUs / 1E6);
 
@@ -1895,7 +1947,7 @@
                     ++mStats.mNumVideoFramesDropped;
                 }
 
-                postVideoEvent_l();
+                postVideoEvent_l(0);
                 return;
             }
         }
@@ -1922,7 +1974,9 @@
             notifyListener_l(MEDIA_INFO, MEDIA_INFO_RENDERING_START);
         }
 
-        notifyIfMediaStarted_l();
+        if (mFlags & PLAYING) {
+            notifyIfMediaStarted_l();
+        }
     }
 
     mVideoBuffer->release();
@@ -1933,6 +1987,41 @@
         return;
     }
 
+    /* get next frame time */
+    if (wasSeeking == NO_SEEK) {
+        MediaSource::ReadOptions options;
+        for (;;) {
+            status_t err = mVideoSource->read(&mVideoBuffer, &options);
+            if (err != OK) {
+                // deal with any errors next time
+                CHECK(mVideoBuffer == NULL);
+                postVideoEvent_l(0);
+                return;
+            }
+
+            if (mVideoBuffer->range_length() != 0) {
+                break;
+            }
+
+            // Some decoders, notably the PV AVC software decoder
+            // return spurious empty buffers that we just want to ignore.
+
+            mVideoBuffer->release();
+            mVideoBuffer = NULL;
+        }
+
+        {
+            Mutex::Autolock autoLock(mStatsLock);
+            ++mStats.mNumVideoFramesDecoded;
+        }
+
+        int64_t nextTimeUs;
+        CHECK(mVideoBuffer->meta_data()->findInt64(kKeyTime, &nextTimeUs));
+        int64_t delayUs = nextTimeUs - ts->getRealTimeUs() + mTimeSourceDeltaUs;
+        postVideoEvent_l(delayUs > 10000 ? 10000 : delayUs < 0 ? 0 : delayUs);
+        return;
+    }
+
     postVideoEvent_l();
 }
 
@@ -2017,7 +2106,10 @@
             mSeekNotificationSent = true;
         }
 
-        mSeeking = NO_SEEK;
+        if (mVideoSource == NULL) {
+            // For video the mSeeking flag is always reset in finishSeekIfNecessary
+            mSeeking = NO_SEEK;
+        }
 
         notifyIfMediaStarted_l();
     }
@@ -2110,15 +2202,14 @@
     if (!strncasecmp("http://", mUri.string(), 7)
             || !strncasecmp("https://", mUri.string(), 8)
             || isWidevineStreaming) {
-        mConnectingDataSource = HTTPBase::Create(
-                (mFlags & INCOGNITO)
-                    ? HTTPBase::kFlagIncognito
-                    : 0);
-
-        if (mUIDValid) {
-            mConnectingDataSource->setUID(mUID);
+        if (mHTTPService == NULL) {
+            ALOGE("Attempt to play media from http URI without HTTP service.");
+            return UNKNOWN_ERROR;
         }
 
+        sp<IMediaHTTPConnection> conn = mHTTPService->makeHTTPConnection();
+        mConnectingDataSource = new MediaHTTP(conn);
+
         String8 cacheConfig;
         bool disconnectAtHighwatermark;
         NuCachedSource2::RemoveCacheSpecificHeaders(
@@ -2126,6 +2217,10 @@
 
         mLock.unlock();
         status_t err = mConnectingDataSource->connect(mUri, &mUriHeaders);
+        // force connection at this point, to avoid a race condition between getMIMEType and the
+        // caching datasource constructed below, which could result in multiple requests to the
+        // server, and/or failed connections.
+        String8 contentType = mConnectingDataSource->getMIMEType();
         mLock.lock();
 
         if (err != OK) {
@@ -2156,8 +2251,6 @@
 
         mConnectingDataSource.clear();
 
-        String8 contentType = dataSource->getMIMEType();
-
         if (strncasecmp(contentType.string(), "audio/", 6)) {
             // We're not doing this for streams that appear to be audio-only
             // streams to ensure that even low bandwidth streams start
@@ -2213,8 +2306,8 @@
                         sniffedMIME = tmp.string();
 
                         if (meta == NULL
-                                || !meta->findInt64(
-                                    "meta-data-size", &metaDataSize)) {
+                                || !meta->findInt64("meta-data-size",
+                                     reinterpret_cast<int64_t*>(&metaDataSize))) {
                             metaDataSize = kHighWaterMarkBytes;
                         }
 
@@ -2234,7 +2327,8 @@
             }
         }
     } else {
-        dataSource = DataSource::CreateFromURI(mUri.string(), &mUriHeaders);
+        dataSource = DataSource::CreateFromURI(
+                mHTTPService, mUri.string(), &mUriHeaders);
     }
 
     if (dataSource == NULL) {
@@ -2301,6 +2395,7 @@
     modifyFlags((PREPARING|PREPARE_CANCELLED|PREPARING_CONNECTED), CLEAR);
     mAsyncPrepareEvent = NULL;
     mPreparedCondition.broadcast();
+    mAudioTearDown = false;
 }
 
 // static
@@ -2374,6 +2469,20 @@
     modifyFlags(PREPARED, SET);
     mAsyncPrepareEvent = NULL;
     mPreparedCondition.broadcast();
+
+    if (mAudioTearDown) {
+        if (mPrepareResult == OK) {
+            if (mExtractorFlags & MediaExtractor::CAN_SEEK) {
+                seekTo_l(mAudioTearDownPosition);
+            }
+
+            if (mAudioTearDownWasPlaying) {
+                modifyFlags(CACHE_UNDERRUN, CLEAR);
+                play_l();
+            }
+        }
+        mAudioTearDown = false;
+    }
 }
 
 uint32_t AwesomePlayer::flags() const {
@@ -2486,12 +2595,12 @@
 status_t AwesomePlayer::selectAudioTrack_l(
         const sp<MediaSource>& source, size_t trackIndex) {
 
-    ALOGI("selectAudioTrack_l: trackIndex=%d, mFlags=0x%x", trackIndex, mFlags);
+    ALOGI("selectAudioTrack_l: trackIndex=%zu, mFlags=0x%x", trackIndex, mFlags);
 
     {
         Mutex::Autolock autoLock(mStatsLock);
         if ((ssize_t)trackIndex == mActiveAudioTrackIndex) {
-            ALOGI("Track %d is active. Does nothing.", trackIndex);
+            ALOGI("Track %zu is active. Does nothing.", trackIndex);
             return OK;
         }
         //mStats.mFlags = mFlags;
@@ -2564,7 +2673,7 @@
         trackCount += mTextDriver->countExternalTracks();
     }
     if (trackIndex >= trackCount) {
-        ALOGE("Track index (%d) is out of range [0, %d)", trackIndex, trackCount);
+        ALOGE("Track index (%zu) is out of range [0, %zu)", trackIndex, trackCount);
         return ERROR_OUT_OF_RANGE;
     }
 
@@ -2576,14 +2685,14 @@
         isAudioTrack = !strncasecmp(mime, "audio/", 6);
 
         if (!isAudioTrack && strcasecmp(mime, MEDIA_MIMETYPE_TEXT_3GPP) != 0) {
-            ALOGE("Track %d is not either audio or timed text", trackIndex);
+            ALOGE("Track %zu is not either audio or timed text", trackIndex);
             return ERROR_UNSUPPORTED;
         }
     }
 
     if (isAudioTrack) {
         if (!select) {
-            ALOGE("Deselect an audio track (%d) is not supported", trackIndex);
+            ALOGE("Deselect an audio track (%zu) is not supported", trackIndex);
             return ERROR_UNSUPPORTED;
         }
         return selectAudioTrack_l(mExtractor->getTrack(trackIndex), trackIndex);
@@ -2661,7 +2770,7 @@
         {
             Mutex::Autolock autoLock(mLock);
             if (mTextDriver == NULL) {
-                mTextDriver = new TimedTextDriver(mListener);
+                mTextDriver = new TimedTextDriver(mListener, mHTTPService);
             }
             // String values written in Parcel are UTF-16 values.
             String8 uri(request.readString16());
@@ -2673,7 +2782,7 @@
         {
             Mutex::Autolock autoLock(mLock);
             if (mTextDriver == NULL) {
-                mTextDriver = new TimedTextDriver(mListener);
+                mTextDriver = new TimedTextDriver(mListener, mHTTPService);
             }
             int fd         = request.readFileDescriptor();
             off64_t offset = request.readInt64();
@@ -2706,7 +2815,8 @@
     return mCachedSource != NULL || mWVMExtractor != NULL;
 }
 
-status_t AwesomePlayer::dump(int fd, const Vector<String16> &args) const {
+status_t AwesomePlayer::dump(
+        int fd, const Vector<String16> & /* args */) const {
     Mutex::Autolock autoLock(mStatsLock);
 
     FILE *out = fdopen(dup(fd), "w");
@@ -2721,7 +2831,7 @@
     fprintf(out, ", flags(0x%08x)", mStats.mFlags);
 
     if (mStats.mBitrate >= 0) {
-        fprintf(out, ", bitrate(%lld bps)", mStats.mBitrate);
+        fprintf(out, ", bitrate(%" PRId64 " bps)", mStats.mBitrate);
     }
 
     fprintf(out, "\n");
@@ -2729,7 +2839,7 @@
     for (size_t i = 0; i < mStats.mTracks.size(); ++i) {
         const TrackStat &stat = mStats.mTracks.itemAt(i);
 
-        fprintf(out, "  Track %d\n", i + 1);
+        fprintf(out, "  Track %zu\n", i + 1);
         fprintf(out, "   MIME(%s)", stat.mMIME.string());
 
         if (!stat.mDecoderName.isEmpty()) {
@@ -2741,8 +2851,8 @@
         if ((ssize_t)i == mStats.mVideoTrackIndex) {
             fprintf(out,
                     "   videoDimensions(%d x %d), "
-                    "numVideoFramesDecoded(%lld), "
-                    "numVideoFramesDropped(%lld)\n",
+                    "numVideoFramesDecoded(%" PRId64 "), "
+                    "numVideoFramesDropped(%" PRId64 ")\n",
                     mStats.mVideoWidth,
                     mStats.mVideoHeight,
                     mStats.mNumVideoFramesDecoded,
@@ -2791,7 +2901,7 @@
     ALOGV("onAudioTearDownEvent");
 
     // stream info is cleared by reset_l() so copy what we need
-    const bool wasPlaying = (mFlags & PLAYING);
+    mAudioTearDownWasPlaying = (mFlags & PLAYING);
     KeyedVector<String8, String8> uriHeaders(mUriHeaders);
     sp<DataSource> fileSource(mFileSource);
 
@@ -2800,8 +2910,9 @@
     mStatsLock.unlock();
 
     // get current position so we can start recreated stream from here
-    int64_t position = 0;
-    getPosition(&position);
+    getPosition(&mAudioTearDownPosition);
+
+    sp<IMediaHTTPService> savedHTTPService = mHTTPService;
 
     // Reset and recreate
     reset_l();
@@ -2812,7 +2923,7 @@
         mFileSource = fileSource;
         err = setDataSource_l(fileSource);
     } else {
-        err = setDataSource_l(uri, &uriHeaders);
+        err = setDataSource_l(savedHTTPService, uri, &uriHeaders);
     }
 
     mFlags |= PREPARING;
@@ -2825,21 +2936,8 @@
     mAudioTearDown = true;
     mIsAsyncPrepare = true;
 
-    // Call parepare for the host decoding
+    // Call prepare for the host decoding
     beginPrepareAsync_l();
-
-    if (mPrepareResult == OK) {
-        if (mExtractorFlags & MediaExtractor::CAN_SEEK) {
-            seekTo_l(position);
-        }
-
-        if (wasPlaying) {
-            modifyFlags(CACHE_UNDERRUN, CLEAR);
-            play_l();
-        }
-    }
-
-    mAudioTearDown = false;
 }
 
 }  // namespace android

media/libstagefright/CameraSource.cpp

diff --git a/media/libstagefright/CameraSource.cpp b/media/libstagefright/CameraSource.cpp
index 3017fe7..b31e9e8 100644
--- a/media/libstagefright/CameraSource.cpp
+++ b/media/libstagefright/CameraSource.cpp
@@ -31,6 +31,12 @@
 #include <utils/String8.h>
 #include <cutils/properties.h>
 
+#if LOG_NDEBUG
+#define UNUSED_UNLESS_VERBOSE(x) (void)(x)
+#else
+#define UNUSED_UNLESS_VERBOSE(x)
+#endif
+
 namespace android {
 
 static const int64_t CAMERA_SOURCE_TIMEOUT_NS = 3000000000LL;
@@ -63,11 +69,14 @@
 }
 
 void CameraSourceListener::notify(int32_t msgType, int32_t ext1, int32_t ext2) {
+    UNUSED_UNLESS_VERBOSE(msgType);
+    UNUSED_UNLESS_VERBOSE(ext1);
+    UNUSED_UNLESS_VERBOSE(ext2);
     ALOGV("notify(%d, %d, %d)", msgType, ext1, ext2);
 }
 
 void CameraSourceListener::postData(int32_t msgType, const sp<IMemory> &dataPtr,
-                                    camera_frame_metadata_t *metadata) {
+                                    camera_frame_metadata_t * /* metadata */) {
     ALOGV("postData(%d, ptr:%p, size:%d)",
          msgType, dataPtr->pointer(), dataPtr->size());
 
@@ -577,14 +586,15 @@
     }
 }
 
-void CameraSource::startCameraRecording() {
+status_t CameraSource::startCameraRecording() {
     ALOGV("startCameraRecording");
     // Reset the identity to the current thread because media server owns the
     // camera and recording is started by the applications. The applications
     // will connect to the camera in ICameraRecordingProxy::startRecording.
     int64_t token = IPCThreadState::self()->clearCallingIdentity();
+    status_t err;
     if (mNumInputBuffers > 0) {
-        status_t err = mCamera->sendCommand(
+        err = mCamera->sendCommand(
             CAMERA_CMD_SET_VIDEO_BUFFER_COUNT, mNumInputBuffers, 0);
 
         // This could happen for CameraHAL1 clients; thus the failure is
@@ -595,17 +605,25 @@
         }
     }
 
+    err = OK;
     if (mCameraFlags & FLAGS_HOT_CAMERA) {
         mCamera->unlock();
         mCamera.clear();
-        CHECK_EQ((status_t)OK,
-            mCameraRecordingProxy->startRecording(new ProxyListener(this)));
+        if ((err = mCameraRecordingProxy->startRecording(
+                new ProxyListener(this))) != OK) {
+            ALOGE("Failed to start recording, received error: %s (%d)",
+                    strerror(-err), err);
+        }
     } else {
         mCamera->setListener(new CameraSourceListener(this));
         mCamera->startRecording();
-        CHECK(mCamera->recordingEnabled());
+        if (!mCamera->recordingEnabled()) {
+            err = -EINVAL;
+            ALOGE("Failed to start recording");
+        }
     }
     IPCThreadState::self()->restoreCallingIdentity(token);
+    return err;
 }
 
 status_t CameraSource::start(MetaData *meta) {
@@ -637,10 +655,12 @@
         }
     }
 
-    startCameraRecording();
+    status_t err;
+    if ((err = startCameraRecording()) == OK) {
+        mStarted = true;
+    }
 
-    mStarted = true;
-    return OK;
+    return err;
 }
 
 void CameraSource::stopCameraRecording() {

media/libstagefright/CameraSourceTimeLapse.cpp

diff --git a/media/libstagefright/CameraSourceTimeLapse.cpp b/media/libstagefright/CameraSourceTimeLapse.cpp
index 20214e8..60cdf66 100644
--- a/media/libstagefright/CameraSourceTimeLapse.cpp
+++ b/media/libstagefright/CameraSourceTimeLapse.cpp
@@ -41,13 +41,15 @@
         Size videoSize,
         int32_t videoFrameRate,
         const sp<IGraphicBufferProducer>& surface,
-        int64_t timeBetweenFrameCaptureUs) {
+        int64_t timeBetweenFrameCaptureUs,
+        bool storeMetaDataInVideoBuffers) {
 
     CameraSourceTimeLapse *source = new
             CameraSourceTimeLapse(camera, proxy, cameraId,
                 clientName, clientUid,
                 videoSize, videoFrameRate, surface,
-                timeBetweenFrameCaptureUs);
+                timeBetweenFrameCaptureUs,
+                storeMetaDataInVideoBuffers);
 
     if (source != NULL) {
         if (source->initCheck() != OK) {
@@ -67,9 +69,11 @@
         Size videoSize,
         int32_t videoFrameRate,
         const sp<IGraphicBufferProducer>& surface,
-        int64_t timeBetweenFrameCaptureUs)
+        int64_t timeBetweenFrameCaptureUs,
+        bool storeMetaDataInVideoBuffers)
       : CameraSource(camera, proxy, cameraId, clientName, clientUid,
-                videoSize, videoFrameRate, surface, true),
+                videoSize, videoFrameRate, surface,
+                storeMetaDataInVideoBuffers),
       mTimeBetweenTimeLapseVideoFramesUs(1E6/videoFrameRate),
       mLastTimeLapseFrameRealTimestampUs(0),
       mSkipCurrentFrame(false) {
@@ -81,7 +85,8 @@
     mVideoWidth = videoSize.width;
     mVideoHeight = videoSize.height;
 
-    if (!trySettingVideoSize(videoSize.width, videoSize.height)) {
+    if (OK == mInitCheck && !trySettingVideoSize(videoSize.width, videoSize.height)) {
+        releaseCamera();
         mInitCheck = NO_INIT;
     }
 
@@ -227,7 +232,7 @@
     return newMemory;
 }
 
-bool CameraSourceTimeLapse::skipCurrentFrame(int64_t timestampUs) {
+bool CameraSourceTimeLapse::skipCurrentFrame(int64_t /* timestampUs */) {
     ALOGV("skipCurrentFrame");
     if (mSkipCurrentFrame) {
         mSkipCurrentFrame = false;

media/libstagefright/DataSource.cpp

diff --git a/media/libstagefright/DataSource.cpp b/media/libstagefright/DataSource.cpp
index fc6fd9c..6e0f37a 100644
--- a/media/libstagefright/DataSource.cpp
+++ b/media/libstagefright/DataSource.cpp
@@ -16,10 +16,6 @@
 
 #include "include/AMRExtractor.h"
 
-#if CHROMIUM_AVAILABLE
-#include "include/chromium_http_stub.h"
-#endif
-
 #include "include/AACExtractor.h"
 #include "include/DRMExtractor.h"
 #include "include/FLACExtractor.h"
@@ -35,10 +31,14 @@
 
 #include "matroska/MatroskaExtractor.h"
 
+#include <media/IMediaHTTPConnection.h>
+#include <media/IMediaHTTPService.h>
 #include <media/stagefright/foundation/AMessage.h>
 #include <media/stagefright/DataSource.h>
+#include <media/stagefright/DataURISource.h>
 #include <media/stagefright/FileSource.h>
 #include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaHTTP.h>
 #include <utils/String8.h>
 
 #include <cutils/properties.h>
@@ -107,6 +107,7 @@
 
 Mutex DataSource::gSnifferMutex;
 List<DataSource::SnifferFunc> DataSource::gSniffers;
+bool DataSource::gSniffersRegistered = false;
 
 bool DataSource::sniff(
         String8 *mimeType, float *confidence, sp<AMessage> *meta) {
@@ -114,7 +115,13 @@
     *confidence = 0.0f;
     meta->clear();
 
-    Mutex::Autolock autoLock(gSnifferMutex);
+    {
+        Mutex::Autolock autoLock(gSnifferMutex);
+        if (!gSniffersRegistered) {
+            return false;
+        }
+    }
+
     for (List<SnifferFunc>::iterator it = gSniffers.begin();
          it != gSniffers.end(); ++it) {
         String8 newMimeType;
@@ -133,9 +140,7 @@
 }
 
 // static
-void DataSource::RegisterSniffer(SnifferFunc func) {
-    Mutex::Autolock autoLock(gSnifferMutex);
-
+void DataSource::RegisterSniffer_l(SnifferFunc func) {
     for (List<SnifferFunc>::iterator it = gSniffers.begin();
          it != gSniffers.end(); ++it) {
         if (*it == func) {
@@ -148,28 +153,36 @@
 
 // static
 void DataSource::RegisterDefaultSniffers() {
-    RegisterSniffer(SniffMPEG4);
-    RegisterSniffer(SniffMatroska);
-    RegisterSniffer(SniffOgg);
-    RegisterSniffer(SniffWAV);
-    RegisterSniffer(SniffFLAC);
-    RegisterSniffer(SniffAMR);
-    RegisterSniffer(SniffMPEG2TS);
-    RegisterSniffer(SniffMP3);
-    RegisterSniffer(SniffAAC);
-    RegisterSniffer(SniffMPEG2PS);
-    RegisterSniffer(SniffWVM);
+    Mutex::Autolock autoLock(gSnifferMutex);
+    if (gSniffersRegistered) {
+        return;
+    }
+
+    RegisterSniffer_l(SniffMPEG4);
+    RegisterSniffer_l(SniffMatroska);
+    RegisterSniffer_l(SniffOgg);
+    RegisterSniffer_l(SniffWAV);
+    RegisterSniffer_l(SniffFLAC);
+    RegisterSniffer_l(SniffAMR);
+    RegisterSniffer_l(SniffMPEG2TS);
+    RegisterSniffer_l(SniffMP3);
+    RegisterSniffer_l(SniffAAC);
+    RegisterSniffer_l(SniffMPEG2PS);
+    RegisterSniffer_l(SniffWVM);
 
     char value[PROPERTY_VALUE_MAX];
     if (property_get("drm.service.enabled", value, NULL)
             && (!strcmp(value, "1") || !strcasecmp(value, "true"))) {
-        RegisterSniffer(SniffDRM);
+        RegisterSniffer_l(SniffDRM);
     }
+    gSniffersRegistered = true;
 }
 
 // static
 sp<DataSource> DataSource::CreateFromURI(
-        const char *uri, const KeyedVector<String8, String8> *headers) {
+        const sp<IMediaHTTPService> &httpService,
+        const char *uri,
+        const KeyedVector<String8, String8> *headers) {
     bool isWidevine = !strncasecmp("widevine://", uri, 11);
 
     sp<DataSource> source;
@@ -178,7 +191,7 @@
     } else if (!strncasecmp("http://", uri, 7)
             || !strncasecmp("https://", uri, 8)
             || isWidevine) {
-        sp<HTTPBase> httpSource = HTTPBase::Create();
+        sp<HTTPBase> httpSource = new MediaHTTP(httpService->makeHTTPConnection());
 
         String8 tmp;
         if (isWidevine) {
@@ -209,11 +222,8 @@
             // in the widevine:// case.
             source = httpSource;
         }
-
-# if CHROMIUM_AVAILABLE
     } else if (!strncasecmp("data:", uri, 5)) {
-        source = createDataUriSource(uri);
-#endif
+        source = DataURISource::Create(uri);
     } else {
         // Assume it's a filename.
         source = new FileSource(uri);

media/libstagefright/DataURISource.cpp

diff --git a/media/libstagefright/DataURISource.cpp b/media/libstagefright/DataURISource.cpp
new file mode 100644
index 0000000..377bc85
--- /dev/null
+++ b/media/libstagefright/DataURISource.cpp
@@ -0,0 +1,109 @@
+/*
+ * 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/DataURISource.h>
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/AString.h>
+#include <media/stagefright/foundation/base64.h>
+
+namespace android {
+
+// static
+sp<DataURISource> DataURISource::Create(const char *uri) {
+    if (strncasecmp("data:", uri, 5)) {
+        return NULL;
+    }
+
+    char *commaPos = strrchr(uri, ',');
+
+    if (commaPos == NULL) {
+        return NULL;
+    }
+
+    sp<ABuffer> buffer;
+
+    AString tmp(&uri[5], commaPos - &uri[5]);
+
+    if (tmp.endsWith(";base64")) {
+        AString encoded(commaPos + 1);
+
+        // Strip CR and LF...
+        for (size_t i = encoded.size(); i-- > 0;) {
+            if (encoded.c_str()[i] == '\r' || encoded.c_str()[i] == '\n') {
+                encoded.erase(i, 1);
+            }
+        }
+
+        buffer = decodeBase64(encoded);
+
+        if (buffer == NULL) {
+            ALOGE("Malformed base64 encoded content found.");
+            return NULL;
+        }
+    } else {
+#if 0
+        size_t dataLen = strlen(uri) - tmp.size() - 6;
+        buffer = new ABuffer(dataLen);
+        memcpy(buffer->data(), commaPos + 1, dataLen);
+
+        // unescape
+#else
+        // MediaPlayer doesn't care for this right now as we don't
+        // play any text-based media.
+        return NULL;
+#endif
+    }
+
+    // We don't really care about charset or mime type.
+
+    return new DataURISource(buffer);
+}
+
+DataURISource::DataURISource(const sp<ABuffer> &buffer)
+    : mBuffer(buffer) {
+}
+
+DataURISource::~DataURISource() {
+}
+
+status_t DataURISource::initCheck() const {
+    return OK;
+}
+
+ssize_t DataURISource::readAt(off64_t offset, void *data, size_t size) {
+    if (offset >= mBuffer->size()) {
+        return 0;
+    }
+
+    size_t copy = mBuffer->size() - offset;
+    if (copy > size) {
+        copy = size;
+    }
+
+    memcpy(data, mBuffer->data() + offset, copy);
+
+    return copy;
+}
+
+status_t DataURISource::getSize(off64_t *size) {
+    *size = mBuffer->size();
+
+    return OK;
+}
+
+}  // namespace android
+

media/libstagefright/FLACExtractor.cpp

diff --git a/media/libstagefright/FLACExtractor.cpp b/media/libstagefright/FLACExtractor.cpp
index 098fcf9..fa7251c 100644
--- a/media/libstagefright/FLACExtractor.cpp
+++ b/media/libstagefright/FLACExtractor.cpp
@@ -208,55 +208,55 @@
 // with the same parameter list, but discard redundant information.
 
 FLAC__StreamDecoderReadStatus FLACParser::read_callback(
-        const FLAC__StreamDecoder *decoder, FLAC__byte buffer[],
+        const FLAC__StreamDecoder * /* decoder */, FLAC__byte buffer[],
         size_t *bytes, void *client_data)
 {
     return ((FLACParser *) client_data)->readCallback(buffer, bytes);
 }
 
 FLAC__StreamDecoderSeekStatus FLACParser::seek_callback(
-        const FLAC__StreamDecoder *decoder,
+        const FLAC__StreamDecoder * /* decoder */,
         FLAC__uint64 absolute_byte_offset, void *client_data)
 {
     return ((FLACParser *) client_data)->seekCallback(absolute_byte_offset);
 }
 
 FLAC__StreamDecoderTellStatus FLACParser::tell_callback(
-        const FLAC__StreamDecoder *decoder,
+        const FLAC__StreamDecoder * /* decoder */,
         FLAC__uint64 *absolute_byte_offset, void *client_data)
 {
     return ((FLACParser *) client_data)->tellCallback(absolute_byte_offset);
 }
 
 FLAC__StreamDecoderLengthStatus FLACParser::length_callback(
-        const FLAC__StreamDecoder *decoder,
+        const FLAC__StreamDecoder * /* decoder */,
         FLAC__uint64 *stream_length, void *client_data)
 {
     return ((FLACParser *) client_data)->lengthCallback(stream_length);
 }
 
 FLAC__bool FLACParser::eof_callback(
-        const FLAC__StreamDecoder *decoder, void *client_data)
+        const FLAC__StreamDecoder * /* decoder */, void *client_data)
 {
     return ((FLACParser *) client_data)->eofCallback();
 }
 
 FLAC__StreamDecoderWriteStatus FLACParser::write_callback(
-        const FLAC__StreamDecoder *decoder, const FLAC__Frame *frame,
+        const FLAC__StreamDecoder * /* decoder */, const FLAC__Frame *frame,
         const FLAC__int32 * const buffer[], void *client_data)
 {
     return ((FLACParser *) client_data)->writeCallback(frame, buffer);
 }
 
 void FLACParser::metadata_callback(
-        const FLAC__StreamDecoder *decoder,
+        const FLAC__StreamDecoder * /* decoder */,
         const FLAC__StreamMetadata *metadata, void *client_data)
 {
     ((FLACParser *) client_data)->metadataCallback(metadata);
 }
 
 void FLACParser::error_callback(
-        const FLAC__StreamDecoder *decoder,
+        const FLAC__StreamDecoder * /* decoder */,
         FLAC__StreamDecoderErrorStatus status, void *client_data)
 {
     ((FLACParser *) client_data)->errorCallback(status);
@@ -380,15 +380,21 @@
 // Copy samples from FLAC native 32-bit non-interleaved to 16-bit interleaved.
 // These are candidates for optimization if needed.
 
-static void copyMono8(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
-{
+static void copyMono8(
+        short *dst,
+        const int *const *src,
+        unsigned nSamples,
+        unsigned /* nChannels */) {
     for (unsigned i = 0; i < nSamples; ++i) {
         *dst++ = src[0][i] << 8;
     }
 }
 
-static void copyStereo8(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
-{
+static void copyStereo8(
+        short *dst,
+        const int *const *src,
+        unsigned nSamples,
+        unsigned /* nChannels */) {
     for (unsigned i = 0; i < nSamples; ++i) {
         *dst++ = src[0][i] << 8;
         *dst++ = src[1][i] << 8;
@@ -404,15 +410,21 @@
     }
 }
 
-static void copyMono16(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
-{
+static void copyMono16(
+        short *dst,
+        const int *const *src,
+        unsigned nSamples,
+        unsigned /* nChannels */) {
     for (unsigned i = 0; i < nSamples; ++i) {
         *dst++ = src[0][i];
     }
 }
 
-static void copyStereo16(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
-{
+static void copyStereo16(
+        short *dst,
+        const int *const *src,
+        unsigned nSamples,
+        unsigned /* nChannels */) {
     for (unsigned i = 0; i < nSamples; ++i) {
         *dst++ = src[0][i];
         *dst++ = src[1][i];
@@ -430,15 +442,21 @@
 
 // 24-bit versions should do dithering or noise-shaping, here or in AudioFlinger
 
-static void copyMono24(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
-{
+static void copyMono24(
+        short *dst,
+        const int *const *src,
+        unsigned nSamples,
+        unsigned /* nChannels */) {
     for (unsigned i = 0; i < nSamples; ++i) {
         *dst++ = src[0][i] >> 8;
     }
 }
 
-static void copyStereo24(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
-{
+static void copyStereo24(
+        short *dst,
+        const int *const *src,
+        unsigned nSamples,
+        unsigned /* nChannels */) {
     for (unsigned i = 0; i < nSamples; ++i) {
         *dst++ = src[0][i] >> 8;
         *dst++ = src[1][i] >> 8;
@@ -454,8 +472,11 @@
     }
 }
 
-static void copyTrespass(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
-{
+static void copyTrespass(
+        short * /* dst */,
+        const int *const * /* src */,
+        unsigned /* nSamples */,
+        unsigned /* nChannels */) {
     TRESPASS();
 }
 
@@ -700,7 +721,7 @@
     }
 }
 
-status_t FLACSource::start(MetaData *params)
+status_t FLACSource::start(MetaData * /* params */)
 {
     ALOGV("FLACSource::start");
 
@@ -792,8 +813,7 @@
 }
 
 sp<MetaData> FLACExtractor::getTrackMetaData(
-        size_t index, uint32_t flags)
-{
+        size_t index, uint32_t /* flags */) {
     if (mInitCheck != OK || index > 0) {
         return NULL;
     }

media/libstagefright/HTTPBase.cpp

diff --git a/media/libstagefright/HTTPBase.cpp b/media/libstagefright/HTTPBase.cpp
index 5fa4b6f..ca68c3d 100644
--- a/media/libstagefright/HTTPBase.cpp
+++ b/media/libstagefright/HTTPBase.cpp
@@ -20,10 +20,6 @@
 
 #include "include/HTTPBase.h"
 
-#if CHROMIUM_AVAILABLE
-#include "include/chromium_http_stub.h"
-#endif
-
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/ALooper.h>
 
@@ -40,34 +36,7 @@
       mTotalTransferBytes(0),
       mPrevBandwidthMeasureTimeUs(0),
       mPrevEstimatedBandWidthKbps(0),
-      mBandWidthCollectFreqMs(5000),
-      mUIDValid(false),
-      mUID(0) {
-}
-
-// static
-sp<HTTPBase> HTTPBase::Create(uint32_t flags) {
-#if CHROMIUM_AVAILABLE
-        HTTPBase *dataSource = createChromiumHTTPDataSource(flags);
-        if (dataSource) {
-           return dataSource;
-        }
-#endif
-    {
-        TRESPASS();
-
-        return NULL;
-    }
-}
-
-// static
-status_t HTTPBase::UpdateProxyConfig(
-        const char *host, int32_t port, const char *exclusionList) {
-#if CHROMIUM_AVAILABLE
-    return UpdateChromiumHTTPDataSourceProxyConfig(host, port, exclusionList);
-#else
-    return INVALID_OPERATION;
-#endif
+      mBandWidthCollectFreqMs(5000) {
 }
 
 void HTTPBase::addBandwidthMeasurement(
@@ -135,21 +104,6 @@
     return OK;
 }
 
-void HTTPBase::setUID(uid_t uid) {
-    mUIDValid = true;
-    mUID = uid;
-}
-
-bool HTTPBase::getUID(uid_t *uid) const {
-    if (!mUIDValid) {
-        return false;
-    }
-
-    *uid = mUID;
-
-    return true;
-}
-
 // static
 void HTTPBase::RegisterSocketUserTag(int sockfd, uid_t uid, uint32_t kTag) {
     int res = qtaguid_tagSocket(sockfd, kTag, uid);

media/libstagefright/MP3Extractor.cpp

diff --git a/media/libstagefright/MP3Extractor.cpp b/media/libstagefright/MP3Extractor.cpp
index 380dab4..4a63152 100644
--- a/media/libstagefright/MP3Extractor.cpp
+++ b/media/libstagefright/MP3Extractor.cpp
@@ -398,7 +398,8 @@
             mSeeker);
 }
 
-sp<MetaData> MP3Extractor::getTrackMetaData(size_t index, uint32_t flags) {
+sp<MetaData> MP3Extractor::getTrackMetaData(
+        size_t index, uint32_t /* flags */) {
     if (mInitCheck != OK || index != 0) {
         return NULL;
     }

media/libstagefright/MPEG2TSWriter.cpp

diff --git a/media/libstagefright/MPEG2TSWriter.cpp b/media/libstagefright/MPEG2TSWriter.cpp
index c9ed5bb..78c12e1 100644
--- a/media/libstagefright/MPEG2TSWriter.cpp
+++ b/media/libstagefright/MPEG2TSWriter.cpp
@@ -555,7 +555,7 @@
     return OK;
 }
 
-status_t MPEG2TSWriter::start(MetaData *param) {
+status_t MPEG2TSWriter::start(MetaData * /* param */) {
     CHECK(!mStarted);
 
     mStarted = true;
@@ -596,7 +596,8 @@
     return !mStarted || (mNumSourcesDone == mSources.size() ? true : false);
 }
 
-status_t MPEG2TSWriter::dump(int fd, const Vector<String16> &args) {
+status_t MPEG2TSWriter::dump(
+        int /* fd */, const Vector<String16> & /* args */) {
     return OK;
 }
 

media/libstagefright/MPEG4Extractor.cpp

diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index cbc169b..2a3fa04 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -39,6 +39,7 @@
 #include <utils/String8.h>
 
 #include <byteswap.h>
+#include "include/ID3.h"
 
 namespace android {
 
@@ -259,7 +260,7 @@
     const uint8_t *data = (const uint8_t *)_data;
     size_t offset = 0;
     while (offset < size) {
-        printf("0x%04x  ", offset);
+        printf("0x%04zx  ", offset);
 
         size_t n = size - offset;
         if (n > 16) {
@@ -487,12 +488,12 @@
             break;
         }
         uint32_t chunk_type = ntohl(hdr[1]);
-        if (chunk_type == FOURCC('s', 'i', 'd', 'x')) {
-            // parse the sidx box too
-            continue;
-        } else if (chunk_type == FOURCC('m', 'o', 'o', 'f')) {
+        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;
     }
@@ -570,7 +571,8 @@
     return size;
 }
 
-status_t MPEG4Extractor::parseDrmSINF(off64_t *offset, off64_t data_offset) {
+status_t MPEG4Extractor::parseDrmSINF(
+        off64_t * /* offset */, off64_t data_offset) {
     uint8_t updateIdTag;
     if (mDataSource->readAt(data_offset, &updateIdTag, 1) < 1) {
         return ERROR_IO;
@@ -682,8 +684,9 @@
             }
             sinf->len = dataLen - 3;
             sinf->IPMPData = new char[sinf->len];
+            data_offset += 2;
 
-            if (mDataSource->readAt(data_offset + 2, sinf->IPMPData, sinf->len) < sinf->len) {
+            if (mDataSource->readAt(data_offset, sinf->IPMPData, sinf->len) < sinf->len) {
                 return ERROR_IO;
             }
             data_offset += sinf->len;
@@ -910,6 +913,8 @@
 
         case FOURCC('e', 'l', 's', 't'):
         {
+            *offset += chunk_size;
+
             // See 14496-12 8.6.6
             uint8_t version;
             if (mDataSource->readAt(data_offset, &version, 1) < 1) {
@@ -962,16 +967,23 @@
                     mLastTrack->meta->setInt32(kKeyEncoderDelay, delay);
 
                     int64_t paddingus = duration - (segment_duration + media_time);
+                    if (paddingus < 0) {
+                        // track duration from media header (which is what kKeyDuration is) might
+                        // be slightly shorter than the segment duration, which would make the
+                        // padding negative. Clamp to zero.
+                        paddingus = 0;
+                    }
                     int64_t paddingsamples = (paddingus * samplerate + 500000) / 1000000;
                     mLastTrack->meta->setInt32(kKeyEncoderPadding, paddingsamples);
                 }
             }
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('f', 'r', 'm', 'a'):
         {
+            *offset += chunk_size;
+
             uint32_t original_fourcc;
             if (mDataSource->readAt(data_offset, &original_fourcc, 4) < 4) {
                 return ERROR_IO;
@@ -985,12 +997,13 @@
                 mLastTrack->meta->setInt32(kKeyChannelCount, num_channels);
                 mLastTrack->meta->setInt32(kKeySampleRate, sample_rate);
             }
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('t', 'e', 'n', 'c'):
         {
+            *offset += chunk_size;
+
             if (chunk_size < 32) {
                 return ERROR_MALFORMED;
             }
@@ -1035,23 +1048,25 @@
             mLastTrack->meta->setInt32(kKeyCryptoMode, defaultAlgorithmId);
             mLastTrack->meta->setInt32(kKeyCryptoDefaultIVSize, defaultIVSize);
             mLastTrack->meta->setData(kKeyCryptoKey, 'tenc', defaultKeyId, 16);
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('t', 'k', 'h', 'd'):
         {
+            *offset += chunk_size;
+
             status_t err;
             if ((err = parseTrackHeader(data_offset, chunk_data_size)) != OK) {
                 return err;
             }
 
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('p', 's', 's', 'h'):
         {
+            *offset += chunk_size;
+
             PsshInfo pssh;
 
             if (mDataSource->readAt(data_offset + 4, &pssh.uuid, 16) < 16) {
@@ -1077,12 +1092,13 @@
             }
             mPssh.push_back(pssh);
 
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('m', 'd', 'h', 'd'):
         {
+            *offset += chunk_size;
+
             if (chunk_data_size < 4) {
                 return ERROR_MALFORMED;
             }
@@ -1163,7 +1179,6 @@
             mLastTrack->meta->setCString(
                     kKeyMediaLanguage, lang_code);
 
-            *offset += chunk_size;
             break;
         }
 
@@ -1330,11 +1345,12 @@
                 mLastTrack->sampleTable->setChunkOffsetParams(
                         chunk_type, data_offset, chunk_data_size);
 
+            *offset += chunk_size;
+
             if (err != OK) {
                 return err;
             }
 
-            *offset += chunk_size;
             break;
         }
 
@@ -1344,11 +1360,12 @@
                 mLastTrack->sampleTable->setSampleToChunkParams(
                         data_offset, chunk_data_size);
 
+            *offset += chunk_size;
+
             if (err != OK) {
                 return err;
             }
 
-            *offset += chunk_size;
             break;
         }
 
@@ -1359,6 +1376,8 @@
                 mLastTrack->sampleTable->setSampleSizeParams(
                         chunk_type, data_offset, chunk_data_size);
 
+            *offset += chunk_size;
+
             if (err != OK) {
                 return err;
             }
@@ -1379,19 +1398,32 @@
             } else {
                 // No size was specified. Pick a conservatively large size.
                 int32_t width, height;
-                if (mLastTrack->meta->findInt32(kKeyWidth, &width) &&
-                        mLastTrack->meta->findInt32(kKeyHeight, &height)) {
-                    mLastTrack->meta->setInt32(kKeyMaxInputSize, width * height * 3 / 2);
-                } else {
+                if (!mLastTrack->meta->findInt32(kKeyWidth, &width) ||
+                    !mLastTrack->meta->findInt32(kKeyHeight, &height)) {
                     ALOGE("No width or height, assuming worst case 1080p");
-                    mLastTrack->meta->setInt32(kKeyMaxInputSize, 3110400);
+                    width = 1920;
+                    height = 1080;
                 }
-            }
-            *offset += chunk_size;
 
-            // Calculate average frame rate.
+                const char *mime;
+                CHECK(mLastTrack->meta->findCString(kKeyMIMEType, &mime));
+                if (!strcmp(mime, MEDIA_MIMETYPE_VIDEO_AVC)) {
+                    // AVC requires compression ratio of at least 2, and uses
+                    // macroblocks
+                    max_size = ((width + 15) / 16) * ((height + 15) / 16) * 192;
+                } else {
+                    // For all other formats there is no minimum compression
+                    // ratio. Use compression ratio of 1.
+                    max_size = width * height * 3 / 2;
+                }
+                mLastTrack->meta->setInt32(kKeyMaxInputSize, max_size);
+            }
+
+            // NOTE: setting another piece of metadata invalidates any pointers (such as the
+            // mimetype) previously obtained, so don't cache them.
             const char *mime;
             CHECK(mLastTrack->meta->findCString(kKeyMIMEType, &mime));
+            // Calculate average frame rate.
             if (!strncasecmp("video/", mime, 6)) {
                 size_t nSamples = mLastTrack->sampleTable->countSamples();
                 int64_t durationUs;
@@ -1409,6 +1441,8 @@
 
         case FOURCC('s', 't', 't', 's'):
         {
+            *offset += chunk_size;
+
             status_t err =
                 mLastTrack->sampleTable->setTimeToSampleParams(
                         data_offset, chunk_data_size);
@@ -1417,12 +1451,13 @@
                 return err;
             }
 
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('c', 't', 't', 's'):
         {
+            *offset += chunk_size;
+
             status_t err =
                 mLastTrack->sampleTable->setCompositionTimeToSampleParams(
                         data_offset, chunk_data_size);
@@ -1431,12 +1466,13 @@
                 return err;
             }
 
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('s', 't', 's', 's'):
         {
+            *offset += chunk_size;
+
             status_t err =
                 mLastTrack->sampleTable->setSyncSampleParams(
                         data_offset, chunk_data_size);
@@ -1445,13 +1481,14 @@
                 return err;
             }
 
-            *offset += chunk_size;
             break;
         }
 
         // @xyz
         case FOURCC('\xA9', 'x', 'y', 'z'):
         {
+            *offset += chunk_size;
+
             // Best case the total data length inside "@xyz" box
             // would be 8, for instance "@xyz" + "\x00\x04\x15\xc7" + "0+0/",
             // where "\x00\x04" is the text string length with value = 4,
@@ -1480,12 +1517,13 @@
 
             buffer[location_length] = '\0';
             mFileMetaData->setCString(kKeyLocation, buffer);
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('e', 's', 'd', 's'):
         {
+            *offset += chunk_size;
+
             if (chunk_data_size < 4) {
                 return ERROR_MALFORMED;
             }
@@ -1523,12 +1561,13 @@
                 }
             }
 
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('a', 'v', 'c', 'C'):
         {
+            *offset += chunk_size;
+
             sp<ABuffer> buffer = new ABuffer(chunk_data_size);
 
             if (mDataSource->readAt(
@@ -1539,12 +1578,12 @@
             mLastTrack->meta->setData(
                     kKeyAVCC, kTypeAVCC, buffer->data(), chunk_data_size);
 
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('d', '2', '6', '3'):
         {
+            *offset += chunk_size;
             /*
              * d263 contains a fixed 7 bytes part:
              *   vendor - 4 bytes
@@ -1570,7 +1609,6 @@
 
             mLastTrack->meta->setData(kKeyD263, kTypeD263, buffer, chunk_data_size);
 
-            *offset += chunk_size;
             break;
         }
 
@@ -1578,11 +1616,13 @@
         {
             uint8_t buffer[4];
             if (chunk_data_size < (off64_t)sizeof(buffer)) {
+                *offset += chunk_size;
                 return ERROR_MALFORMED;
             }
 
             if (mDataSource->readAt(
                         data_offset, buffer, 4) < 4) {
+                *offset += chunk_size;
                 return ERROR_IO;
             }
 
@@ -1616,6 +1656,8 @@
         case FOURCC('n', 'a', 'm', 'e'):
         case FOURCC('d', 'a', 't', 'a'):
         {
+            *offset += chunk_size;
+
             if (mPath.size() == 6 && underMetaDataPath(mPath)) {
                 status_t err = parseITunesMetaData(data_offset, chunk_data_size);
 
@@ -1624,12 +1666,13 @@
                 }
             }
 
-            *offset += chunk_size;
             break;
         }
 
         case FOURCC('m', 'v', 'h', 'd'):
         {
+            *offset += chunk_size;
+
             if (chunk_data_size < 24) {
                 return ERROR_MALFORMED;
             }
@@ -1657,7 +1700,6 @@
 
             mFileMetaData->setCString(kKeyDate, s.string());
 
-            *offset += chunk_size;
             break;
         }
 
@@ -1678,6 +1720,8 @@
 
         case FOURCC('h', 'd', 'l', 'r'):
         {
+            *offset += chunk_size;
+
             uint32_t buffer;
             if (mDataSource->readAt(
                         data_offset + 8, &buffer, 4) < 4) {
@@ -1692,7 +1736,6 @@
                 mLastTrack->meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_TEXT_3GPP);
             }
 
-            *offset += chunk_size;
             break;
         }
 
@@ -1717,6 +1760,8 @@
                 delete[] buffer;
                 buffer = NULL;
 
+                // advance read pointer so we don't end up reading this again
+                *offset += chunk_size;
                 return ERROR_IO;
             }
 
@@ -1731,6 +1776,8 @@
 
         case FOURCC('c', 'o', 'v', 'r'):
         {
+            *offset += chunk_size;
+
             if (mFileMetaData != NULL) {
                 ALOGV("chunk_data_size = %lld and data_offset = %lld",
                         chunk_data_size, data_offset);
@@ -1745,7 +1792,6 @@
                     buffer->data() + kSkipBytesOfDataBox, chunk_data_size - kSkipBytesOfDataBox);
             }
 
-            *offset += chunk_size;
             break;
         }
 
@@ -1756,13 +1802,27 @@
         case FOURCC('a', 'l', 'b', 'm'):
         case FOURCC('y', 'r', 'r', 'c'):
         {
+            *offset += chunk_size;
+
             status_t err = parse3GPPMetaData(data_offset, chunk_data_size, depth);
 
             if (err != OK) {
                 return err;
             }
 
+            break;
+        }
+
+        case FOURCC('I', 'D', '3', '2'):
+        {
             *offset += chunk_size;
+
+            if (chunk_data_size < 6) {
+                return ERROR_MALFORMED;
+            }
+
+            parseID3v2MetaData(data_offset + 6);
+
             break;
         }
 
@@ -1886,9 +1946,10 @@
             ALOGW("sub-sidx boxes not supported yet");
         }
         bool sap = d3 & 0x80000000;
-        bool saptype = d3 >> 28;
-        if (!sap || saptype > 2) {
-            ALOGW("not a stream access point, or unsupported type");
+        uint32_t saptype = (d3 >> 28) & 7;
+        if (!sap || (saptype != 1 && saptype != 2)) {
+            // type 1 and 2 are sync samples
+            ALOGW("not a stream access point, or unsupported type: %08x", d3);
         }
         total_duration += d2;
         offset += 12;
@@ -2146,7 +2207,7 @@
             break;
     }
 
-    if (size >= 8 && metadataKey) {
+    if (size >= 8 && metadataKey && !mFileMetaData->hasData(metadataKey)) {
         if (metadataKey == kKeyAlbumArt) {
             mFileMetaData->setData(
                     kKeyAlbumArt, MetaData::TYPE_NONE,
@@ -2295,6 +2356,62 @@
     return OK;
 }
 
+void MPEG4Extractor::parseID3v2MetaData(off64_t offset) {
+    ID3 id3(mDataSource, true /* ignorev1 */, offset);
+
+    if (id3.isValid()) {
+        struct Map {
+            int key;
+            const char *tag1;
+            const char *tag2;
+        };
+        static const Map kMap[] = {
+            { kKeyAlbum, "TALB", "TAL" },
+            { kKeyArtist, "TPE1", "TP1" },
+            { kKeyAlbumArtist, "TPE2", "TP2" },
+            { kKeyComposer, "TCOM", "TCM" },
+            { kKeyGenre, "TCON", "TCO" },
+            { kKeyTitle, "TIT2", "TT2" },
+            { kKeyYear, "TYE", "TYER" },
+            { kKeyAuthor, "TXT", "TEXT" },
+            { kKeyCDTrackNumber, "TRK", "TRCK" },
+            { kKeyDiscNumber, "TPA", "TPOS" },
+            { kKeyCompilation, "TCP", "TCMP" },
+        };
+        static const size_t kNumMapEntries = sizeof(kMap) / sizeof(kMap[0]);
+
+        for (size_t i = 0; i < kNumMapEntries; ++i) {
+            if (!mFileMetaData->hasData(kMap[i].key)) {
+                ID3::Iterator *it = new ID3::Iterator(id3, kMap[i].tag1);
+                if (it->done()) {
+                    delete it;
+                    it = new ID3::Iterator(id3, kMap[i].tag2);
+                }
+
+                if (it->done()) {
+                    delete it;
+                    continue;
+                }
+
+                String8 s;
+                it->getString(&s);
+                delete it;
+
+                mFileMetaData->setCString(kMap[i].key, s);
+            }
+        }
+
+        size_t dataSize;
+        String8 mime;
+        const void *data = id3.getAlbumArt(&dataSize, &mime);
+
+        if (data) {
+            mFileMetaData->setData(kKeyAlbumArt, MetaData::TYPE_NONE, data, dataSize);
+            mFileMetaData->setCString(kKeyAlbumArtMIME, mime.string());
+        }
+    }
+}
+
 sp<MediaSource> MPEG4Extractor::getTrack(size_t index) {
     status_t err;
     if ((err = readMetaData()) != OK) {
@@ -2351,6 +2468,58 @@
     return OK;
 }
 
+typedef enum {
+    //AOT_NONE             = -1,
+    //AOT_NULL_OBJECT      = 0,
+    //AOT_AAC_MAIN         = 1, /**< Main profile                              */
+    AOT_AAC_LC           = 2,   /**< Low Complexity object                     */
+    //AOT_AAC_SSR          = 3,
+    //AOT_AAC_LTP          = 4,
+    AOT_SBR              = 5,
+    //AOT_AAC_SCAL         = 6,
+    //AOT_TWIN_VQ          = 7,
+    //AOT_CELP             = 8,
+    //AOT_HVXC             = 9,
+    //AOT_RSVD_10          = 10, /**< (reserved)                                */
+    //AOT_RSVD_11          = 11, /**< (reserved)                                */
+    //AOT_TTSI             = 12, /**< TTSI Object                               */
+    //AOT_MAIN_SYNTH       = 13, /**< Main Synthetic object                     */
+    //AOT_WAV_TAB_SYNTH    = 14, /**< Wavetable Synthesis object                */
+    //AOT_GEN_MIDI         = 15, /**< General MIDI object                       */
+    //AOT_ALG_SYNTH_AUD_FX = 16, /**< Algorithmic Synthesis and Audio FX object */
+    AOT_ER_AAC_LC        = 17,   /**< Error Resilient(ER) AAC Low Complexity    */
+    //AOT_RSVD_18          = 18, /**< (reserved)                                */
+    //AOT_ER_AAC_LTP       = 19, /**< Error Resilient(ER) AAC LTP object        */
+    AOT_ER_AAC_SCAL      = 20,   /**< Error Resilient(ER) AAC Scalable object   */
+    //AOT_ER_TWIN_VQ       = 21, /**< Error Resilient(ER) TwinVQ object         */
+    AOT_ER_BSAC          = 22,   /**< Error Resilient(ER) BSAC object           */
+    AOT_ER_AAC_LD        = 23,   /**< Error Resilient(ER) AAC LowDelay object   */
+    //AOT_ER_CELP          = 24, /**< Error Resilient(ER) CELP object           */
+    //AOT_ER_HVXC          = 25, /**< Error Resilient(ER) HVXC object           */
+    //AOT_ER_HILN          = 26, /**< Error Resilient(ER) HILN object           */
+    //AOT_ER_PARA          = 27, /**< Error Resilient(ER) Parametric object     */
+    //AOT_RSVD_28          = 28, /**< might become SSC                          */
+    AOT_PS               = 29,   /**< PS, Parametric Stereo (includes SBR)      */
+    //AOT_MPEGS            = 30, /**< MPEG Surround                             */
+
+    AOT_ESCAPE           = 31,   /**< Signal AOT uses more than 5 bits          */
+
+    //AOT_MP3ONMP4_L1      = 32, /**< MPEG-Layer1 in mp4                        */
+    //AOT_MP3ONMP4_L2      = 33, /**< MPEG-Layer2 in mp4                        */
+    //AOT_MP3ONMP4_L3      = 34, /**< MPEG-Layer3 in mp4                        */
+    //AOT_RSVD_35          = 35, /**< might become DST                          */
+    //AOT_RSVD_36          = 36, /**< might become ALS                          */
+    //AOT_AAC_SLS          = 37, /**< AAC + SLS                                 */
+    //AOT_SLS              = 38, /**< SLS                                       */
+    //AOT_ER_AAC_ELD       = 39, /**< AAC Enhanced Low Delay                    */
+
+    //AOT_USAC             = 42, /**< USAC                                      */
+    //AOT_SAOC             = 43, /**< SAOC                                      */
+    //AOT_LD_MPEGS         = 44, /**< Low Delay MPEG Surround                   */
+
+    //AOT_RSVD50           = 50,  /**< Interim AOT for Rsvd50                   */
+} AUDIO_OBJECT_TYPE;
+
 status_t MPEG4Extractor::updateAudioTrackInfoFromESDS_MPEG4Audio(
         const void *esds_data, size_t esds_size) {
     ESDS esds(esds_data, esds_size);
@@ -2398,6 +2567,11 @@
         return ERROR_MALFORMED;
     }
 
+    static uint32_t kSamplingRate[] = {
+        96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
+        16000, 12000, 11025, 8000, 7350
+    };
+
     ABitReader br(csd, csd_size);
     uint32_t objectType = br.getBits(5);
 
@@ -2405,6 +2579,9 @@
         objectType = 32 + br.getBits(6);
     }
 
+    //keep AOT type
+    mLastTrack->meta->setInt32(kKeyAACAOT, objectType);
+
     uint32_t freqIndex = br.getBits(4);
 
     int32_t sampleRate = 0;
@@ -2417,28 +2594,134 @@
         numChannels = br.getBits(4);
     } else {
         numChannels = br.getBits(4);
-        if (objectType == 5) {
-            // SBR specific config per 14496-3 table 1.13
-            freqIndex = br.getBits(4);
-            if (freqIndex == 15) {
-                if (csd_size < 8) {
-                    return ERROR_MALFORMED;
-                }
-                sampleRate = br.getBits(24);
-            }
+
+        if (freqIndex == 13 || freqIndex == 14) {
+            return ERROR_MALFORMED;
         }
 
-        if (sampleRate == 0) {
-            static uint32_t kSamplingRate[] = {
-                96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
-                16000, 12000, 11025, 8000, 7350
-            };
+        sampleRate = kSamplingRate[freqIndex];
+    }
 
-            if (freqIndex == 13 || freqIndex == 14) {
+    if (objectType == AOT_SBR || objectType == AOT_PS) {//SBR specific config per 14496-3 table 1.13
+        uint32_t extFreqIndex = br.getBits(4);
+        int32_t extSampleRate;
+        if (extFreqIndex == 15) {
+            if (csd_size < 8) {
                 return ERROR_MALFORMED;
             }
+            extSampleRate = br.getBits(24);
+        } else {
+            if (extFreqIndex == 13 || extFreqIndex == 14) {
+                return ERROR_MALFORMED;
+            }
+            extSampleRate = kSamplingRate[extFreqIndex];
+        }
+        //TODO: save the extension sampling rate value in meta data =>
+        //      mLastTrack->meta->setInt32(kKeyExtSampleRate, extSampleRate);
+    }
 
-            sampleRate = kSamplingRate[freqIndex];
+    switch (numChannels) {
+        // values defined in 14496-3_2009 amendment-4 Table 1.19 - Channel Configuration
+        case 0:
+        case 1:// FC
+        case 2:// FL FR
+        case 3:// FC, FL FR
+        case 4:// FC, FL FR, RC
+        case 5:// FC, FL FR, SL SR
+        case 6:// FC, FL FR, SL SR, LFE
+            //numChannels already contains the right value
+            break;
+        case 11:// FC, FL FR, SL SR, RC, LFE
+            numChannels = 7;
+            break;
+        case 7: // FC, FCL FCR, FL FR, SL SR, LFE
+        case 12:// FC, FL  FR,  SL SR, RL RR, LFE
+        case 14:// FC, FL  FR,  SL SR, LFE, FHL FHR
+            numChannels = 8;
+            break;
+        default:
+            return ERROR_UNSUPPORTED;
+    }
+
+    {
+        if (objectType == AOT_SBR || objectType == AOT_PS) {
+            const int32_t extensionSamplingFrequency = br.getBits(4);
+            objectType = br.getBits(5);
+
+            if (objectType == AOT_ESCAPE) {
+                objectType = 32 + br.getBits(6);
+            }
+        }
+        if (objectType == AOT_AAC_LC || objectType == AOT_ER_AAC_LC ||
+                objectType == AOT_ER_AAC_LD || objectType == AOT_ER_AAC_SCAL ||
+                objectType == AOT_ER_BSAC) {
+            const int32_t frameLengthFlag = br.getBits(1);
+
+            const int32_t dependsOnCoreCoder = br.getBits(1);
+
+            if (dependsOnCoreCoder ) {
+                const int32_t coreCoderDelay = br.getBits(14);
+            }
+
+            const int32_t extensionFlag = br.getBits(1);
+
+            if (numChannels == 0 ) {
+                int32_t channelsEffectiveNum = 0;
+                int32_t channelsNum = 0;
+                const int32_t ElementInstanceTag = br.getBits(4);
+                const int32_t Profile = br.getBits(2);
+                const int32_t SamplingFrequencyIndex = br.getBits(4);
+                const int32_t NumFrontChannelElements = br.getBits(4);
+                const int32_t NumSideChannelElements = br.getBits(4);
+                const int32_t NumBackChannelElements = br.getBits(4);
+                const int32_t NumLfeChannelElements = br.getBits(2);
+                const int32_t NumAssocDataElements = br.getBits(3);
+                const int32_t NumValidCcElements = br.getBits(4);
+
+                const int32_t MonoMixdownPresent = br.getBits(1);
+                if (MonoMixdownPresent != 0) {
+                    const int32_t MonoMixdownElementNumber = br.getBits(4);
+                }
+
+                const int32_t StereoMixdownPresent = br.getBits(1);
+                if (StereoMixdownPresent != 0) {
+                    const int32_t StereoMixdownElementNumber = br.getBits(4);
+                }
+
+                const int32_t MatrixMixdownIndexPresent = br.getBits(1);
+                if (MatrixMixdownIndexPresent != 0) {
+                    const int32_t MatrixMixdownIndex = br.getBits(2);
+                    const int32_t PseudoSurroundEnable = br.getBits(1);
+                }
+
+                int i;
+                for (i=0; i < NumFrontChannelElements; i++) {
+                    const int32_t FrontElementIsCpe = br.getBits(1);
+                    const int32_t FrontElementTagSelect = br.getBits(4);
+                    channelsNum += FrontElementIsCpe ? 2 : 1;
+                }
+
+                for (i=0; i < NumSideChannelElements; i++) {
+                    const int32_t SideElementIsCpe = br.getBits(1);
+                    const int32_t SideElementTagSelect = br.getBits(4);
+                    channelsNum += SideElementIsCpe ? 2 : 1;
+                }
+
+                for (i=0; i < NumBackChannelElements; i++) {
+                    const int32_t BackElementIsCpe = br.getBits(1);
+                    const int32_t BackElementTagSelect = br.getBits(4);
+                    channelsNum += BackElementIsCpe ? 2 : 1;
+                }
+                channelsEffectiveNum = channelsNum;
+
+                for (i=0; i < NumLfeChannelElements; i++) {
+                    const int32_t LfeElementTagSelect = br.getBits(4);
+                    channelsNum += 1;
+                }
+                ALOGV("mpeg4 audio channelsNum = %d", channelsNum);
+                ALOGV("mpeg4 audio channelsEffectiveNum = %d", channelsEffectiveNum);
+                numChannels = channelsNum;
+            }
         }
     }
 
@@ -2642,9 +2925,20 @@
                 }
             }
             if (chunk_type == FOURCC('m', 'o', 'o', 'f')) {
-                // *offset points to the mdat box following this moof
-                parseChunk(offset); // doesn't actually parse it, just updates offset
-                mNextMoofOffset = *offset;
+                // *offset points to the box following this moof. Find the next moof from there.
+
+                while (true) {
+                    if (mDataSource->readAt(*offset, hdr, 8) < 8) {
+                        return ERROR_END_OF_STREAM;
+                    }
+                    chunk_size = ntohl(hdr[0]);
+                    chunk_type = ntohl(hdr[1]);
+                    if (chunk_type == FOURCC('m', 'o', 'o', 'f')) {
+                        mNextMoofOffset = *offset;
+                        break;
+                    }
+                    *offset += chunk_size;
+                }
             }
             break;
         }
@@ -2703,7 +2997,8 @@
     return OK;
 }
 
-status_t MPEG4Source::parseSampleAuxiliaryInformationSizes(off64_t offset, off64_t size) {
+status_t MPEG4Source::parseSampleAuxiliaryInformationSizes(
+        off64_t offset, off64_t /* size */) {
     ALOGV("parseSampleAuxiliaryInformationSizes");
     // 14496-12 8.7.12
     uint8_t version;
@@ -2765,7 +3060,8 @@
     return OK;
 }
 
-status_t MPEG4Source::parseSampleAuxiliaryInformationOffsets(off64_t offset, off64_t size) {
+status_t MPEG4Source::parseSampleAuxiliaryInformationOffsets(
+        off64_t offset, off64_t /* size */) {
     ALOGV("parseSampleAuxiliaryInformationOffsets");
     // 14496-12 8.7.13
     uint8_t version;
@@ -3447,7 +3743,7 @@
                 const SidxEntry *se = &mSegments[i];
                 if (totalTime + se->mDurationUs > seekTimeUs) {
                     // The requested time is somewhere in this segment
-                    if ((mode == ReadOptions::SEEK_NEXT_SYNC) ||
+                    if ((mode == ReadOptions::SEEK_NEXT_SYNC && seekTimeUs > totalTime) ||
                         (mode == ReadOptions::SEEK_CLOSEST_SYNC &&
                         (seekTimeUs - totalTime) > (totalTime + se->mDurationUs - seekTimeUs))) {
                         // requested next sync, or closest sync and it was closer to the end of
@@ -3460,11 +3756,19 @@
                 totalTime += se->mDurationUs;
                 totalOffset += se->mSize;
             }
-        mCurrentMoofOffset = totalOffset;
-        mCurrentSamples.clear();
-        mCurrentSampleIndex = 0;
-        parseChunk(&totalOffset);
-        mCurrentTime = totalTime * mTimescale / 1000000ll;
+            mCurrentMoofOffset = totalOffset;
+            mCurrentSamples.clear();
+            mCurrentSampleIndex = 0;
+            parseChunk(&totalOffset);
+            mCurrentTime = totalTime * mTimescale / 1000000ll;
+        } else {
+            // without sidx boxes, we can only seek to 0
+            mCurrentMoofOffset = mFirstMoofOffset;
+            mCurrentSamples.clear();
+            mCurrentSampleIndex = 0;
+            off64_t tmp = mCurrentMoofOffset;
+            parseChunk(&tmp);
+            mCurrentTime = 0;
         }
 
         if (mBuffer != NULL) {
@@ -3476,7 +3780,7 @@
     }
 
     off64_t offset = 0;
-    size_t size;
+    size_t size = 0;
     uint32_t cts = 0;
     bool isSyncSample = false;
     bool newBuffer = false;
@@ -3484,16 +3788,18 @@
         newBuffer = true;
 
         if (mCurrentSampleIndex >= mCurrentSamples.size()) {
-            // move to next fragment
-            Sample lastSample = mCurrentSamples[mCurrentSamples.size() - 1];
-            off64_t nextMoof = mNextMoofOffset; // lastSample.offset + lastSample.size;
+            // move to next fragment if there is one
+            if (mNextMoofOffset <= mCurrentMoofOffset) {
+                return ERROR_END_OF_STREAM;
+            }
+            off64_t nextMoof = mNextMoofOffset;
             mCurrentMoofOffset = nextMoof;
             mCurrentSamples.clear();
             mCurrentSampleIndex = 0;
             parseChunk(&nextMoof);
-                if (mCurrentSampleIndex >= mCurrentSamples.size()) {
-                    return ERROR_END_OF_STREAM;
-                }
+            if (mCurrentSampleIndex >= mCurrentSamples.size()) {
+                return ERROR_END_OF_STREAM;
+            }
         }
 
         const Sample *smpl = &mCurrentSamples[mCurrentSampleIndex];

media/libstagefright/MPEG4Writer.cpp

diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index a0f17b5..c839560 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -16,6 +16,7 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "MPEG4Writer"
+#include <inttypes.h>
 #include <utils/Log.h>
 
 #include <arpa/inet.h>
@@ -43,7 +44,9 @@
 namespace android {
 
 static const int64_t kMinStreamableFileSizeInBytes = 5 * 1024 * 1024;
-static const int64_t kMax32BitFileSize = 0x007fffffffLL;
+static const int64_t kMax32BitFileSize = 0x00ffffffffLL; // 2^32-1 : max FAT32
+                                                         // filesystem file size
+                                                         // used by most SD cards
 static const uint8_t kNalUnitTypeSeqParamSet = 0x07;
 static const uint8_t kNalUnitTypePicParamSet = 0x08;
 static const int64_t kInitialDelayTimeUs     = 700000LL;
@@ -406,7 +409,7 @@
 }
 
 status_t MPEG4Writer::Track::dump(
-        int fd, const Vector<String16>& args) const {
+        int fd, const Vector<String16>& /* args */) const {
     const size_t SIZE = 256;
     char buffer[SIZE];
     String8 result;
@@ -417,7 +420,7 @@
     result.append(buffer);
     snprintf(buffer, SIZE, "       frames encoded : %d\n", mStszTableEntries->count());
     result.append(buffer);
-    snprintf(buffer, SIZE, "       duration encoded : %lld us\n", mTrackDurationUs);
+    snprintf(buffer, SIZE, "       duration encoded : %" PRId64 " us\n", mTrackDurationUs);
     result.append(buffer);
     ::write(fd, result.string(), result.size());
     return OK;
@@ -860,11 +863,11 @@
     // Fix up the size of the 'mdat' chunk.
     if (mUse32BitOffset) {
         lseek64(mFd, mMdatOffset, SEEK_SET);
-        int32_t size = htonl(static_cast<int32_t>(mOffset - mMdatOffset));
+        uint32_t size = htonl(static_cast<uint32_t>(mOffset - mMdatOffset));
         ::write(mFd, &size, 4);
     } else {
         lseek64(mFd, mMdatOffset + 8, SEEK_SET);
-        int64_t size = mOffset - mMdatOffset;
+        uint64_t size = mOffset - mMdatOffset;
         size = hton64(size);
         ::write(mFd, &size, 8);
     }
@@ -972,13 +975,16 @@
     if (param && param->findInt32(kKeyFileType, &fileType) &&
         fileType != OUTPUT_FORMAT_MPEG_4) {
         writeFourcc("3gp4");
-    } else {
+        writeInt32(0);
         writeFourcc("isom");
+        writeFourcc("3gp4");
+    } else {
+        writeFourcc("mp42");
+        writeInt32(0);
+        writeFourcc("isom");
+        writeFourcc("mp42");
     }
 
-    writeInt32(0);
-    writeFourcc("isom");
-    writeFourcc("3gp4");
     endBox();
 }
 
@@ -1404,7 +1410,7 @@
         size_t sampleCount, int32_t duration) {
 
     if (duration == 0) {
-        ALOGW("0-duration samples found: %d", sampleCount);
+        ALOGW("0-duration samples found: %zu", sampleCount);
     }
     mSttsTableEntries->add(htonl(sampleCount));
     mSttsTableEntries->add(htonl(duration));
@@ -1584,7 +1590,7 @@
     sendSessionSummary();
 
     mChunkInfos.clear();
-    ALOGD("%d chunks are written in the last batch", outstandingChunks);
+    ALOGD("%zu chunks are written in the last batch", outstandingChunks);
 }
 
 bool MPEG4Writer::findChunkToWrite(Chunk *chunk) {
@@ -1760,7 +1766,7 @@
 }
 
 status_t MPEG4Writer::Track::stop() {
-    ALOGD("Stopping %s track", mIsAudio? "Audio": "Video");
+    ALOGD("%s track stopping", mIsAudio? "Audio": "Video");
     if (!mStarted) {
         ALOGE("Stop() called but track is not started");
         return ERROR_END_OF_STREAM;
@@ -1771,18 +1777,13 @@
     }
     mDone = true;
 
+    ALOGD("%s track source stopping", mIsAudio? "Audio": "Video");
+    mSource->stop();
+    ALOGD("%s track source stopped", mIsAudio? "Audio": "Video");
+
     void *dummy;
     pthread_join(mThread, &dummy);
-
-    status_t err = (status_t) dummy;
-
-    ALOGD("Stopping %s track source", mIsAudio? "Audio": "Video");
-    {
-        status_t status = mSource->stop();
-        if (err == OK && status != OK && status != ERROR_END_OF_STREAM) {
-            err = status;
-        }
-    }
+    status_t err = static_cast<status_t>(reinterpret_cast<uintptr_t>(dummy));
 
     ALOGD("%s track stopped", mIsAudio? "Audio": "Video");
     return err;
@@ -1797,7 +1798,7 @@
     Track *track = static_cast<Track *>(me);
 
     status_t err = track->threadEntry();
-    return (void *) err;
+    return (void *)(uintptr_t)err;
 }
 
 static void getNalUnitType(uint8_t byte, uint8_t* type) {
@@ -1869,7 +1870,7 @@
     // 2 bytes for each of the parameter set length field
     // plus the 7 bytes for the header
     if (size < 4 + 7) {
-        ALOGE("Codec specific data length too short: %d", size);
+        ALOGE("Codec specific data length too short: %zu", size);
         return ERROR_MALFORMED;
     }
 
@@ -1938,7 +1939,7 @@
         }
 
         if (nSeqParamSets > 0x1F) {
-            ALOGE("Too many seq parameter sets (%d) found", nSeqParamSets);
+            ALOGE("Too many seq parameter sets (%zu) found", nSeqParamSets);
             return ERROR_MALFORMED;
         }
     }
@@ -1951,7 +1952,7 @@
             return ERROR_MALFORMED;
         }
         if (nPicParamSets > 0xFF) {
-            ALOGE("Too many pic parameter sets (%d) found", nPicParamSets);
+            ALOGE("Too many pic parameter sets (%zd) found", nPicParamSets);
             return ERROR_MALFORMED;
         }
     }
@@ -1981,7 +1982,7 @@
     }
 
     if (size < 4) {
-        ALOGE("Codec specific data length too short: %d", size);
+        ALOGE("Codec specific data length too short: %zu", size);
         return ERROR_MALFORMED;
     }
 
@@ -2279,6 +2280,22 @@
             return UNKNOWN_ERROR;
         }
 
+        // if the duration is different for this sample, see if it is close enough to the previous
+        // duration that we can fudge it and use the same value, to avoid filling the stts table
+        // with lots of near-identical entries.
+        // "close enough" here means that the current duration needs to be adjusted by less
+        // than 0.1 milliseconds
+        if (lastDurationTicks && (currDurationTicks != lastDurationTicks)) {
+            int64_t deltaUs = ((lastDurationTicks - currDurationTicks) * 1000000LL
+                    + (mTimeScale / 2)) / mTimeScale;
+            if (deltaUs > -100 && deltaUs < 100) {
+                // use previous ticks, and adjust timestamp as if it was actually that number
+                // of ticks
+                currDurationTicks = lastDurationTicks;
+                timestampUs += deltaUs;
+            }
+        }
+
         mStszTableEntries->add(htonl(sampleSize));
         if (mStszTableEntries->count() > 2) {
 

media/libstagefright/MediaAdapter.cpp

diff --git a/media/libstagefright/MediaAdapter.cpp b/media/libstagefright/MediaAdapter.cpp
index 2484212..d680e0c 100644
--- a/media/libstagefright/MediaAdapter.cpp
+++ b/media/libstagefright/MediaAdapter.cpp
@@ -36,7 +36,7 @@
     CHECK(mCurrentMediaBuffer == NULL);
 }
 
-status_t MediaAdapter::start(MetaData *params) {
+status_t MediaAdapter::start(MetaData * /* params */) {
     Mutex::Autolock autoLock(mAdapterLock);
     if (!mStarted) {
         mStarted = true;
@@ -75,7 +75,7 @@
 }
 
 status_t MediaAdapter::read(
-            MediaBuffer **buffer, const ReadOptions *options) {
+            MediaBuffer **buffer, const ReadOptions * /* options */) {
     Mutex::Autolock autoLock(mAdapterLock);
     if (!mStarted) {
         ALOGV("Read before even started!");

media/libstagefright/MediaCodec.cpp

diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index e299caf..fe21296 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -115,7 +115,7 @@
         if (codecIdx >= 0) {
             Vector<AString> types;
             if (mcl->getSupportedTypes(codecIdx, &types) == OK) {
-                for (int i = 0; i < types.size(); i++) {
+                for (size_t i = 0; i < types.size(); i++) {
                     if (types[i].startsWith("video/")) {
                         needDedicatedLooper = true;
                         break;
@@ -1506,7 +1506,8 @@
             info->mOwnedByClient = false;
 
             if (portIndex == kPortIndexInput) {
-                msg->setInt32("err", ERROR_END_OF_STREAM);
+                /* no error, just returning buffers */
+                msg->setInt32("err", OK);
             }
             msg->post();
         }
@@ -1679,7 +1680,7 @@
         return -EACCES;
     }
 
-    if (render && (info->mData == NULL || info->mData->size() != 0)) {
+    if (render && info->mData != NULL && info->mData->size() != 0) {
         info->mNotify->setInt32("render", true);
 
         if (mSoftRenderer != NULL) {

media/libstagefright/MediaCodecSource.cpp

diff --git a/media/libstagefright/MediaCodecSource.cpp b/media/libstagefright/MediaCodecSource.cpp
new file mode 100644
index 0000000..924173c
--- /dev/null
+++ b/media/libstagefright/MediaCodecSource.cpp
@@ -0,0 +1,881 @@
+/*
+ * Copyright 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 "MediaCodecSource"
+#define DEBUG_DRIFT_TIME 0
+#include <gui/IGraphicBufferProducer.h>
+#include <gui/Surface.h>
+#include <media/ICrypto.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/MediaBuffer.h>
+#include <media/stagefright/MediaCodec.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/MediaCodecSource.h>
+#include <media/stagefright/Utils.h>
+
+namespace android {
+
+static void ReleaseMediaBufferReference(const sp<ABuffer> &accessUnit) {
+    void *mbuf;
+    if (accessUnit->meta()->findPointer("mediaBuffer", &mbuf)
+            && mbuf != NULL) {
+        ALOGV("releasing mbuf %p", mbuf);
+
+        accessUnit->meta()->setPointer("mediaBuffer", NULL);
+
+        static_cast<MediaBuffer *>(mbuf)->release();
+        mbuf = NULL;
+    }
+}
+
+struct MediaCodecSource::Puller : public AHandler {
+    Puller(const sp<MediaSource> &source);
+
+    status_t start(const sp<MetaData> &meta, const sp<AMessage> &notify);
+    void stopAsync();
+
+    void pause();
+    void resume();
+
+protected:
+    virtual void onMessageReceived(const sp<AMessage> &msg);
+    virtual ~Puller();
+
+private:
+    enum {
+        kWhatStart = 'msta',
+        kWhatStop,
+        kWhatPull,
+        kWhatPause,
+        kWhatResume,
+    };
+
+    sp<MediaSource> mSource;
+    sp<AMessage> mNotify;
+    sp<ALooper> mLooper;
+    int32_t mPullGeneration;
+    bool mIsAudio;
+    bool mPaused;
+    bool mReachedEOS;
+
+    status_t postSynchronouslyAndReturnError(const sp<AMessage> &msg);
+    void schedulePull();
+    void handleEOS();
+
+    DISALLOW_EVIL_CONSTRUCTORS(Puller);
+};
+
+MediaCodecSource::Puller::Puller(const sp<MediaSource> &source)
+    : mSource(source),
+      mLooper(new ALooper()),
+      mPullGeneration(0),
+      mIsAudio(false),
+      mPaused(false),
+      mReachedEOS(false) {
+    sp<MetaData> meta = source->getFormat();
+    const char *mime;
+    CHECK(meta->findCString(kKeyMIMEType, &mime));
+
+    mIsAudio = !strncasecmp(mime, "audio/", 6);
+
+    mLooper->setName("pull_looper");
+}
+
+MediaCodecSource::Puller::~Puller() {
+    mLooper->unregisterHandler(id());
+    mLooper->stop();
+}
+
+status_t MediaCodecSource::Puller::postSynchronouslyAndReturnError(
+        const sp<AMessage> &msg) {
+    sp<AMessage> response;
+    status_t err = msg->postAndAwaitResponse(&response);
+
+    if (err != OK) {
+        return err;
+    }
+
+    if (!response->findInt32("err", &err)) {
+        err = OK;
+    }
+
+    return err;
+}
+
+status_t MediaCodecSource::Puller::start(const sp<MetaData> &meta,
+        const sp<AMessage> &notify) {
+    ALOGV("puller (%s) start", mIsAudio ? "audio" : "video");
+    mLooper->start(
+            false /* runOnCallingThread */,
+            false /* canCallJava */,
+            PRIORITY_AUDIO);
+    mLooper->registerHandler(this);
+    mNotify = notify;
+
+    sp<AMessage> msg = new AMessage(kWhatStart, id());
+    msg->setObject("meta", meta);
+    return postSynchronouslyAndReturnError(msg);
+}
+
+void MediaCodecSource::Puller::stopAsync() {
+    ALOGV("puller (%s) stopAsync", mIsAudio ? "audio" : "video");
+    (new AMessage(kWhatStop, id()))->post();
+}
+
+void MediaCodecSource::Puller::pause() {
+    (new AMessage(kWhatPause, id()))->post();
+}
+
+void MediaCodecSource::Puller::resume() {
+    (new AMessage(kWhatResume, id()))->post();
+}
+
+void MediaCodecSource::Puller::schedulePull() {
+    sp<AMessage> msg = new AMessage(kWhatPull, id());
+    msg->setInt32("generation", mPullGeneration);
+    msg->post();
+}
+
+void MediaCodecSource::Puller::handleEOS() {
+    if (!mReachedEOS) {
+        ALOGV("puller (%s) posting EOS", mIsAudio ? "audio" : "video");
+        mReachedEOS = true;
+        sp<AMessage> notify = mNotify->dup();
+        notify->setPointer("accessUnit", NULL);
+        notify->post();
+    }
+}
+
+void MediaCodecSource::Puller::onMessageReceived(const sp<AMessage> &msg) {
+    switch (msg->what()) {
+        case kWhatStart:
+        {
+            sp<RefBase> obj;
+            CHECK(msg->findObject("meta", &obj));
+
+            mReachedEOS = false;
+
+            status_t err = mSource->start(static_cast<MetaData *>(obj.get()));
+
+            if (err == OK) {
+                schedulePull();
+            }
+
+            sp<AMessage> response = new AMessage;
+            response->setInt32("err", err);
+
+            uint32_t replyID;
+            CHECK(msg->senderAwaitsResponse(&replyID));
+            response->postReply(replyID);
+            break;
+        }
+
+        case kWhatStop:
+        {
+            ALOGV("source (%s) stopping", mIsAudio ? "audio" : "video");
+            mSource->stop();
+            ALOGV("source (%s) stopped", mIsAudio ? "audio" : "video");
+            ++mPullGeneration;
+
+            handleEOS();
+            break;
+        }
+
+        case kWhatPull:
+        {
+            int32_t generation;
+            CHECK(msg->findInt32("generation", &generation));
+
+            if (generation != mPullGeneration) {
+                break;
+            }
+
+            MediaBuffer *mbuf;
+            status_t err = mSource->read(&mbuf);
+
+            if (mPaused) {
+                if (err == OK) {
+                    mbuf->release();
+                    mbuf = NULL;
+                }
+
+                msg->post();
+                break;
+            }
+
+            if (err != OK) {
+                if (err == ERROR_END_OF_STREAM) {
+                    ALOGV("stream ended, mbuf %p", mbuf);
+                } else {
+                    ALOGE("error %d reading stream.", err);
+                }
+                handleEOS();
+            } else {
+                sp<AMessage> notify = mNotify->dup();
+
+                notify->setPointer("accessUnit", mbuf);
+                notify->post();
+
+                msg->post();
+            }
+            break;
+        }
+
+        case kWhatPause:
+        {
+            mPaused = true;
+            break;
+        }
+
+        case kWhatResume:
+        {
+            mPaused = false;
+            break;
+        }
+
+        default:
+            TRESPASS();
+    }
+}
+
+// static
+sp<MediaCodecSource> MediaCodecSource::Create(
+        const sp<ALooper> &looper,
+        const sp<AMessage> &format,
+        const sp<MediaSource> &source,
+        uint32_t flags) {
+    sp<MediaCodecSource> mediaSource =
+            new MediaCodecSource(looper, format, source, flags);
+
+    if (mediaSource->init() == OK) {
+        return mediaSource;
+    }
+    return NULL;
+}
+
+status_t MediaCodecSource::start(MetaData* params) {
+    sp<AMessage> msg = new AMessage(kWhatStart, mReflector->id());
+    msg->setObject("meta", params);
+    return postSynchronouslyAndReturnError(msg);
+}
+
+status_t MediaCodecSource::stop() {
+    sp<AMessage> msg = new AMessage(kWhatStop, mReflector->id());
+    return postSynchronouslyAndReturnError(msg);
+}
+
+status_t MediaCodecSource::pause() {
+    (new AMessage(kWhatPause, mReflector->id()))->post();
+    return OK;
+}
+
+sp<IGraphicBufferProducer> MediaCodecSource::getGraphicBufferProducer() {
+    CHECK(mFlags & FLAG_USE_SURFACE_INPUT);
+    return mGraphicBufferProducer;
+}
+
+status_t MediaCodecSource::read(
+        MediaBuffer** buffer, const ReadOptions* /* options */) {
+    Mutex::Autolock autolock(mOutputBufferLock);
+
+    *buffer = NULL;
+    while (mOutputBufferQueue.size() == 0 && !mEncodedReachedEOS) {
+        mOutputBufferCond.wait(mOutputBufferLock);
+    }
+    if (!mEncodedReachedEOS) {
+        *buffer = *mOutputBufferQueue.begin();
+        mOutputBufferQueue.erase(mOutputBufferQueue.begin());
+        return OK;
+    }
+    return mErrorCode;
+}
+
+void MediaCodecSource::signalBufferReturned(MediaBuffer *buffer) {
+    buffer->setObserver(0);
+    buffer->release();
+}
+
+MediaCodecSource::MediaCodecSource(
+        const sp<ALooper> &looper,
+        const sp<AMessage> &outputFormat,
+        const sp<MediaSource> &source,
+        uint32_t flags)
+    : mLooper(looper),
+      mOutputFormat(outputFormat),
+      mMeta(new MetaData),
+      mFlags(flags),
+      mIsVideo(false),
+      mStarted(false),
+      mStopping(false),
+      mDoMoreWorkPending(false),
+      mPullerReachedEOS(false),
+      mFirstSampleTimeUs(-1ll),
+      mEncodedReachedEOS(false),
+      mErrorCode(OK) {
+    CHECK(mLooper != NULL);
+
+    AString mime;
+    CHECK(mOutputFormat->findString("mime", &mime));
+
+    if (!strncasecmp("video/", mime.c_str(), 6)) {
+        mIsVideo = true;
+    }
+
+    if (!(mFlags & FLAG_USE_SURFACE_INPUT)) {
+        mPuller = new Puller(source);
+    }
+}
+
+MediaCodecSource::~MediaCodecSource() {
+    releaseEncoder();
+
+    mCodecLooper->stop();
+    mLooper->unregisterHandler(mReflector->id());
+}
+
+status_t MediaCodecSource::init() {
+    status_t err = initEncoder();
+
+    if (err != OK) {
+        releaseEncoder();
+    }
+
+    return err;
+}
+
+status_t MediaCodecSource::initEncoder() {
+    mReflector = new AHandlerReflector<MediaCodecSource>(this);
+    mLooper->registerHandler(mReflector);
+
+    mCodecLooper = new ALooper;
+    mCodecLooper->setName("codec_looper");
+    mCodecLooper->start();
+
+    if (mFlags & FLAG_USE_METADATA_INPUT) {
+        mOutputFormat->setInt32("store-metadata-in-buffers", 1);
+    }
+
+    if (mFlags & FLAG_USE_SURFACE_INPUT) {
+        mOutputFormat->setInt32("create-input-buffers-suspended", 1);
+    }
+
+    AString outputMIME;
+    CHECK(mOutputFormat->findString("mime", &outputMIME));
+
+    mEncoder = MediaCodec::CreateByType(
+            mCodecLooper, outputMIME.c_str(), true /* encoder */);
+
+    if (mEncoder == NULL) {
+        return NO_INIT;
+    }
+
+    ALOGV("output format is '%s'", mOutputFormat->debugString(0).c_str());
+
+    status_t err = mEncoder->configure(
+                mOutputFormat,
+                NULL /* nativeWindow */,
+                NULL /* crypto */,
+                MediaCodec::CONFIGURE_FLAG_ENCODE);
+
+    if (err != OK) {
+        return err;
+    }
+
+    mEncoder->getOutputFormat(&mOutputFormat);
+    convertMessageToMetaData(mOutputFormat, mMeta);
+
+    if (mFlags & FLAG_USE_SURFACE_INPUT) {
+        CHECK(mIsVideo);
+
+        err = mEncoder->createInputSurface(&mGraphicBufferProducer);
+
+        if (err != OK) {
+            return err;
+        }
+    }
+
+    err = mEncoder->start();
+
+    if (err != OK) {
+        return err;
+    }
+
+    err = mEncoder->getInputBuffers(&mEncoderInputBuffers);
+
+    if (err != OK) {
+        return err;
+    }
+
+    err = mEncoder->getOutputBuffers(&mEncoderOutputBuffers);
+
+    if (err != OK) {
+        return err;
+    }
+
+    mEncodedReachedEOS = false;
+    mErrorCode = OK;
+
+    return OK;
+}
+
+void MediaCodecSource::releaseEncoder() {
+    if (mEncoder == NULL) {
+        return;
+    }
+
+    mEncoder->release();
+    mEncoder.clear();
+
+    while (!mInputBufferQueue.empty()) {
+        MediaBuffer *mbuf = *mInputBufferQueue.begin();
+        mInputBufferQueue.erase(mInputBufferQueue.begin());
+        if (mbuf != NULL) {
+            mbuf->release();
+        }
+    }
+
+    for (size_t i = 0; i < mEncoderInputBuffers.size(); ++i) {
+        sp<ABuffer> accessUnit = mEncoderInputBuffers.itemAt(i);
+        ReleaseMediaBufferReference(accessUnit);
+    }
+
+    mEncoderInputBuffers.clear();
+    mEncoderOutputBuffers.clear();
+}
+
+bool MediaCodecSource::reachedEOS() {
+    return mEncodedReachedEOS && ((mPuller == NULL) || mPullerReachedEOS);
+}
+
+status_t MediaCodecSource::postSynchronouslyAndReturnError(
+        const sp<AMessage> &msg) {
+    sp<AMessage> response;
+    status_t err = msg->postAndAwaitResponse(&response);
+
+    if (err != OK) {
+        return err;
+    }
+
+    if (!response->findInt32("err", &err)) {
+        err = OK;
+    }
+
+    return err;
+}
+
+void MediaCodecSource::signalEOS(status_t err) {
+    if (!mEncodedReachedEOS) {
+        ALOGI("encoder (%s) reached EOS", mIsVideo ? "video" : "audio");
+        {
+            Mutex::Autolock autoLock(mOutputBufferLock);
+            // release all unread media buffers
+            for (List<MediaBuffer*>::iterator it = mOutputBufferQueue.begin();
+                    it != mOutputBufferQueue.end(); it++) {
+                (*it)->release();
+            }
+            mOutputBufferQueue.clear();
+            mEncodedReachedEOS = true;
+            mErrorCode = err;
+            mOutputBufferCond.signal();
+        }
+
+        releaseEncoder();
+    }
+    if (mStopping && reachedEOS()) {
+        ALOGI("MediaCodecSource (%s) fully stopped",
+                mIsVideo ? "video" : "audio");
+        // posting reply to everyone that's waiting
+        List<uint32_t>::iterator it;
+        for (it = mStopReplyIDQueue.begin();
+                it != mStopReplyIDQueue.end(); it++) {
+            (new AMessage)->postReply(*it);
+        }
+        mStopReplyIDQueue.clear();
+        mStopping = false;
+    }
+}
+
+void MediaCodecSource::suspend() {
+    CHECK(mFlags & FLAG_USE_SURFACE_INPUT);
+    if (mEncoder != NULL) {
+        sp<AMessage> params = new AMessage;
+        params->setInt32("drop-input-frames", true);
+        mEncoder->setParameters(params);
+    }
+}
+
+void MediaCodecSource::resume(int64_t skipFramesBeforeUs) {
+    CHECK(mFlags & FLAG_USE_SURFACE_INPUT);
+    if (mEncoder != NULL) {
+        sp<AMessage> params = new AMessage;
+        params->setInt32("drop-input-frames", false);
+        if (skipFramesBeforeUs > 0) {
+            params->setInt64("skip-frames-before", skipFramesBeforeUs);
+        }
+        mEncoder->setParameters(params);
+    }
+}
+
+void MediaCodecSource::scheduleDoMoreWork() {
+    if (mDoMoreWorkPending) {
+        return;
+    }
+
+    mDoMoreWorkPending = true;
+
+    if (mEncoderActivityNotify == NULL) {
+        mEncoderActivityNotify = new AMessage(
+                kWhatEncoderActivity, mReflector->id());
+    }
+    mEncoder->requestActivityNotification(mEncoderActivityNotify);
+}
+
+status_t MediaCodecSource::feedEncoderInputBuffers() {
+    while (!mInputBufferQueue.empty()
+            && !mAvailEncoderInputIndices.empty()) {
+        MediaBuffer* mbuf = *mInputBufferQueue.begin();
+        mInputBufferQueue.erase(mInputBufferQueue.begin());
+
+        size_t bufferIndex = *mAvailEncoderInputIndices.begin();
+        mAvailEncoderInputIndices.erase(mAvailEncoderInputIndices.begin());
+
+        int64_t timeUs = 0ll;
+        uint32_t flags = 0;
+        size_t size = 0;
+
+        if (mbuf != NULL) {
+            CHECK(mbuf->meta_data()->findInt64(kKeyTime, &timeUs));
+
+            // push decoding time for video, or drift time for audio
+            if (mIsVideo) {
+                mDecodingTimeQueue.push_back(timeUs);
+            } else {
+#if DEBUG_DRIFT_TIME
+                if (mFirstSampleTimeUs < 0ll) {
+                    mFirstSampleTimeUs = timeUs;
+                }
+
+                int64_t driftTimeUs = 0;
+                if (mbuf->meta_data()->findInt64(kKeyDriftTime, &driftTimeUs)
+                        && driftTimeUs) {
+                    driftTimeUs = timeUs - mFirstSampleTimeUs - driftTimeUs;
+                }
+                mDriftTimeQueue.push_back(driftTimeUs);
+#endif // DEBUG_DRIFT_TIME
+            }
+
+            size = mbuf->size();
+
+            memcpy(mEncoderInputBuffers.itemAt(bufferIndex)->data(),
+                   mbuf->data(), size);
+
+            if (mIsVideo) {
+                // video encoder will release MediaBuffer when done
+                // with underlying data.
+                mEncoderInputBuffers.itemAt(bufferIndex)->meta()
+                        ->setPointer("mediaBuffer", mbuf);
+            } else {
+                mbuf->release();
+            }
+        } else {
+            flags = MediaCodec::BUFFER_FLAG_EOS;
+        }
+
+        status_t err = mEncoder->queueInputBuffer(
+                bufferIndex, 0, size, timeUs, flags);
+
+        if (err != OK) {
+            return err;
+        }
+    }
+
+    return OK;
+}
+
+status_t MediaCodecSource::doMoreWork() {
+    status_t err;
+
+    if (!(mFlags & FLAG_USE_SURFACE_INPUT)) {
+        for (;;) {
+            size_t bufferIndex;
+            err = mEncoder->dequeueInputBuffer(&bufferIndex);
+
+            if (err != OK) {
+                break;
+            }
+
+            mAvailEncoderInputIndices.push_back(bufferIndex);
+        }
+
+        feedEncoderInputBuffers();
+    }
+
+    for (;;) {
+        size_t bufferIndex;
+        size_t offset;
+        size_t size;
+        int64_t timeUs;
+        uint32_t flags;
+        native_handle_t* handle = NULL;
+        err = mEncoder->dequeueOutputBuffer(
+                &bufferIndex, &offset, &size, &timeUs, &flags);
+
+        if (err != OK) {
+            if (err == INFO_FORMAT_CHANGED) {
+                continue;
+            } else if (err == INFO_OUTPUT_BUFFERS_CHANGED) {
+                mEncoder->getOutputBuffers(&mEncoderOutputBuffers);
+                continue;
+            }
+
+            if (err == -EAGAIN) {
+                err = OK;
+            }
+            break;
+        }
+        if (!(flags & MediaCodec::BUFFER_FLAG_EOS)) {
+            sp<ABuffer> outbuf = mEncoderOutputBuffers.itemAt(bufferIndex);
+
+            MediaBuffer *mbuf = new MediaBuffer(outbuf->size());
+            memcpy(mbuf->data(), outbuf->data(), outbuf->size());
+
+            if (!(flags & MediaCodec::BUFFER_FLAG_CODECCONFIG)) {
+                if (mIsVideo) {
+                    int64_t decodingTimeUs;
+                    if (mFlags & FLAG_USE_SURFACE_INPUT) {
+                        // GraphicBufferSource is supposed to discard samples
+                        // queued before start, and offset timeUs by start time
+                        CHECK_GE(timeUs, 0ll);
+                        // TODO:
+                        // Decoding time for surface source is unavailable,
+                        // use presentation time for now. May need to move
+                        // this logic into MediaCodec.
+                        decodingTimeUs = timeUs;
+                    } else {
+                        CHECK(!mDecodingTimeQueue.empty());
+                        decodingTimeUs = *(mDecodingTimeQueue.begin());
+                        mDecodingTimeQueue.erase(mDecodingTimeQueue.begin());
+                    }
+                    mbuf->meta_data()->setInt64(kKeyDecodingTime, decodingTimeUs);
+
+                    ALOGV("[video] time %lld us (%.2f secs), dts/pts diff %lld",
+                            timeUs, timeUs / 1E6, decodingTimeUs - timeUs);
+                } else {
+                    int64_t driftTimeUs = 0;
+#if DEBUG_DRIFT_TIME
+                    CHECK(!mDriftTimeQueue.empty());
+                    driftTimeUs = *(mDriftTimeQueue.begin());
+                    mDriftTimeQueue.erase(mDriftTimeQueue.begin());
+                    mbuf->meta_data()->setInt64(kKeyDriftTime, driftTimeUs);
+#endif // DEBUG_DRIFT_TIME
+                    ALOGV("[audio] time %lld us (%.2f secs), drift %lld",
+                            timeUs, timeUs / 1E6, driftTimeUs);
+                }
+                mbuf->meta_data()->setInt64(kKeyTime, timeUs);
+            } else {
+                mbuf->meta_data()->setInt32(kKeyIsCodecConfig, true);
+            }
+            if (flags & MediaCodec::BUFFER_FLAG_SYNCFRAME) {
+                mbuf->meta_data()->setInt32(kKeyIsSyncFrame, true);
+            }
+            mbuf->setObserver(this);
+            mbuf->add_ref();
+
+            {
+                Mutex::Autolock autoLock(mOutputBufferLock);
+                mOutputBufferQueue.push_back(mbuf);
+                mOutputBufferCond.signal();
+            }
+        }
+
+        mEncoder->releaseOutputBuffer(bufferIndex);
+
+        if (flags & MediaCodec::BUFFER_FLAG_EOS) {
+            err = ERROR_END_OF_STREAM;
+            break;
+        }
+    }
+
+    return err;
+}
+
+status_t MediaCodecSource::onStart(MetaData *params) {
+    if (mStopping) {
+        ALOGE("Failed to start while we're stopping");
+        return INVALID_OPERATION;
+    }
+
+    if (mStarted) {
+        ALOGI("MediaCodecSource (%s) resuming", mIsVideo ? "video" : "audio");
+        if (mFlags & FLAG_USE_SURFACE_INPUT) {
+            resume();
+        } else {
+            CHECK(mPuller != NULL);
+            mPuller->resume();
+        }
+        return OK;
+    }
+
+    ALOGI("MediaCodecSource (%s) starting", mIsVideo ? "video" : "audio");
+
+    status_t err = OK;
+
+    if (mFlags & FLAG_USE_SURFACE_INPUT) {
+        int64_t startTimeUs;
+        if (!params || !params->findInt64(kKeyTime, &startTimeUs)) {
+            startTimeUs = -1ll;
+        }
+        resume(startTimeUs);
+        scheduleDoMoreWork();
+    } else {
+        CHECK(mPuller != NULL);
+        sp<AMessage> notify = new AMessage(
+                kWhatPullerNotify, mReflector->id());
+        err = mPuller->start(params, notify);
+        if (err != OK) {
+            mPullerReachedEOS = true;
+            return err;
+        }
+    }
+
+    ALOGI("MediaCodecSource (%s) started", mIsVideo ? "video" : "audio");
+
+    mStarted = true;
+    return OK;
+}
+
+void MediaCodecSource::onMessageReceived(const sp<AMessage> &msg) {
+    switch (msg->what()) {
+    case kWhatPullerNotify:
+    {
+        MediaBuffer *mbuf;
+        CHECK(msg->findPointer("accessUnit", (void**)&mbuf));
+
+        if (mbuf == NULL) {
+            ALOGI("puller (%s) reached EOS",
+                    mIsVideo ? "video" : "audio");
+            mPullerReachedEOS = true;
+        }
+
+        if (mEncoder == NULL) {
+            ALOGV("got msg '%s' after encoder shutdown.",
+                  msg->debugString().c_str());
+
+            if (mbuf != NULL) {
+                mbuf->release();
+            } else {
+                signalEOS();
+            }
+            break;
+        }
+
+        mInputBufferQueue.push_back(mbuf);
+
+        feedEncoderInputBuffers();
+        scheduleDoMoreWork();
+
+        break;
+    }
+    case kWhatEncoderActivity:
+    {
+        mDoMoreWorkPending = false;
+
+        if (mEncoder == NULL) {
+            break;
+        }
+
+        status_t err = doMoreWork();
+
+        if (err == OK) {
+            scheduleDoMoreWork();
+        } else {
+            // reached EOS, or error
+            signalEOS(err);
+        }
+
+        break;
+    }
+    case kWhatStart:
+    {
+        uint32_t replyID;
+        CHECK(msg->senderAwaitsResponse(&replyID));
+
+        sp<RefBase> obj;
+        CHECK(msg->findObject("meta", &obj));
+        MetaData *params = static_cast<MetaData *>(obj.get());
+
+        sp<AMessage> response = new AMessage;
+        response->setInt32("err", onStart(params));
+        response->postReply(replyID);
+        break;
+    }
+    case kWhatStop:
+    {
+        ALOGI("MediaCodecSource (%s) stopping", mIsVideo ? "video" : "audio");
+
+        uint32_t replyID;
+        CHECK(msg->senderAwaitsResponse(&replyID));
+
+        if (reachedEOS()) {
+            // if we already reached EOS, reply and return now
+            ALOGI("MediaCodecSource (%s) already stopped",
+                    mIsVideo ? "video" : "audio");
+            (new AMessage)->postReply(replyID);
+            break;
+        }
+
+        mStopReplyIDQueue.push_back(replyID);
+        if (mStopping) {
+            // nothing to do if we're already stopping, reply will be posted
+            // to all when we're stopped.
+            break;
+        }
+
+        mStopping = true;
+
+        // if using surface, signal source EOS and wait for EOS to come back.
+        // otherwise, release encoder and post EOS if haven't done already
+        if (mFlags & FLAG_USE_SURFACE_INPUT) {
+            mEncoder->signalEndOfInputStream();
+        } else {
+            CHECK(mPuller != NULL);
+            mPuller->stopAsync();
+            signalEOS();
+        }
+        break;
+    }
+    case kWhatPause:
+    {
+        if (mFlags && FLAG_USE_SURFACE_INPUT) {
+            suspend();
+        } else {
+            CHECK(mPuller != NULL);
+            mPuller->pause();
+        }
+        break;
+    }
+    default:
+        TRESPASS();
+    }
+}
+
+} // namespace android

media/libstagefright/MediaDefs.cpp

diff --git a/media/libstagefright/MediaDefs.cpp b/media/libstagefright/MediaDefs.cpp
index b5d4e44..c670bb4 100644
--- a/media/libstagefright/MediaDefs.cpp
+++ b/media/libstagefright/MediaDefs.cpp
@@ -36,12 +36,14 @@
 const char *MEDIA_MIMETYPE_AUDIO_AAC = "audio/mp4a-latm";
 const char *MEDIA_MIMETYPE_AUDIO_QCELP = "audio/qcelp";
 const char *MEDIA_MIMETYPE_AUDIO_VORBIS = "audio/vorbis";
+const char *MEDIA_MIMETYPE_AUDIO_OPUS = "audio/opus";
 const char *MEDIA_MIMETYPE_AUDIO_G711_ALAW = "audio/g711-alaw";
 const char *MEDIA_MIMETYPE_AUDIO_G711_MLAW = "audio/g711-mlaw";
 const char *MEDIA_MIMETYPE_AUDIO_RAW = "audio/raw";
 const char *MEDIA_MIMETYPE_AUDIO_FLAC = "audio/flac";
 const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS = "audio/aac-adts";
 const char *MEDIA_MIMETYPE_AUDIO_MSGSM = "audio/gsm";
+const char *MEDIA_MIMETYPE_AUDIO_AC3 = "audio/ac3";
 
 const char *MEDIA_MIMETYPE_CONTAINER_MPEG4 = "video/mp4";
 const char *MEDIA_MIMETYPE_CONTAINER_WAV = "audio/x-wav";

media/libstagefright/MetaData.cpp

diff --git a/media/libstagefright/MetaData.cpp b/media/libstagefright/MetaData.cpp
index ae6ae2d..74234a6 100644
--- a/media/libstagefright/MetaData.cpp
+++ b/media/libstagefright/MetaData.cpp
@@ -16,6 +16,7 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "MetaData"
+#include <inttypes.h>
 #include <utils/Log.h>
 
 #include <stdlib.h>
@@ -89,6 +90,9 @@
     return setData(key, TYPE_RECT, &r, sizeof(r));
 }
 
+/**
+ * Note that the returned pointer becomes invalid when additional metadata is set.
+ */
 bool MetaData::findCString(uint32_t key, const char **value) {
     uint32_t type;
     const void *data;
@@ -218,6 +222,16 @@
     return true;
 }
 
+bool MetaData::hasData(uint32_t key) const {
+    ssize_t i = mItems.indexOfKey(key);
+
+    if (i < 0) {
+        return false;
+    }
+
+    return true;
+}
+
 MetaData::typed_data::typed_data()
     : mType(0),
       mSize(0) {
@@ -294,7 +308,7 @@
     const void *data = storage();
     switch(mType) {
         case TYPE_NONE:
-            out = String8::format("no type, size %d)", mSize);
+            out = String8::format("no type, size %zu)", mSize);
             break;
         case TYPE_C_STRING:
             out = String8::format("(char*) %s", (const char *)data);
@@ -303,7 +317,7 @@
             out = String8::format("(int32_t) %d", *(int32_t *)data);
             break;
         case TYPE_INT64:
-            out = String8::format("(int64_t) %lld", *(int64_t *)data);
+            out = String8::format("(int64_t) %" PRId64, *(int64_t *)data);
             break;
         case TYPE_FLOAT:
             out = String8::format("(float) %f", *(float *)data);
@@ -320,7 +334,7 @@
         }
 
         default:
-            out = String8::format("(unknown type %d, size %d)", mType, mSize);
+            out = String8::format("(unknown type %d, size %zu)", mType, mSize);
             if (mSize <= 48) { // if it's less than three lines of hex data, dump it
                 AString foo;
                 hexdump(data, mSize, 0, &foo);

media/libstagefright/NuCachedSource2.cpp

diff --git a/media/libstagefright/NuCachedSource2.cpp b/media/libstagefright/NuCachedSource2.cpp
index 05e599b..72ea32d 100644
--- a/media/libstagefright/NuCachedSource2.cpp
+++ b/media/libstagefright/NuCachedSource2.cpp
@@ -213,7 +213,14 @@
 
     mLooper->setName("NuCachedSource2");
     mLooper->registerHandler(mReflector);
-    mLooper->start();
+
+    // Since it may not be obvious why our looper thread needs to be
+    // able to call into java since it doesn't appear to do so at all...
+    // IMediaHTTPConnection may be (and most likely is) implemented in JAVA
+    // and a local JAVA IBinder will call directly into JNI methods.
+    // So whenever we call DataSource::readAt it may end up in a call to
+    // IMediaHTTPConnection::readAt and therefore call back into JAVA.
+    mLooper->start(false /* runOnCallingThread */, true /* canCallJava */);
 
     Mutex::Autolock autoLock(mLock);
     (new AMessage(kWhatFetchMore, mReflector->id()))->post();
@@ -326,7 +333,7 @@
             mNumRetriesLeft = 0;
         }
 
-        ALOGE("source returned error %ld, %d retries left", n, mNumRetriesLeft);
+        ALOGE("source returned error %d, %d retries left", n, mNumRetriesLeft);
         mCache->releasePage(page);
     } else if (n == 0) {
         ALOGI("ERROR_END_OF_STREAM");
@@ -641,7 +648,7 @@
     ssize_t lowwaterMarkKb, highwaterMarkKb;
     int keepAliveSecs;
 
-    if (sscanf(s, "%ld/%ld/%d",
+    if (sscanf(s, "%d/%d/%d",
                &lowwaterMarkKb, &highwaterMarkKb, &keepAliveSecs) != 3) {
         ALOGE("Failed to parse cache parameters from '%s'.", s);
         return;

media/libstagefright/NuMediaExtractor.cpp

diff --git a/media/libstagefright/NuMediaExtractor.cpp b/media/libstagefright/NuMediaExtractor.cpp
index 7bc7da2..64f56e9 100644
--- a/media/libstagefright/NuMediaExtractor.cpp
+++ b/media/libstagefright/NuMediaExtractor.cpp
@@ -58,7 +58,9 @@
 }
 
 status_t NuMediaExtractor::setDataSource(
-        const char *path, const KeyedVector<String8, String8> *headers) {
+        const sp<IMediaHTTPService> &httpService,
+        const char *path,
+        const KeyedVector<String8, String8> *headers) {
     Mutex::Autolock autoLock(mLock);
 
     if (mImpl != NULL) {
@@ -66,7 +68,7 @@
     }
 
     sp<DataSource> dataSource =
-        DataSource::CreateFromURI(path, headers);
+        DataSource::CreateFromURI(httpService, path, headers);
 
     if (dataSource == NULL) {
         return -ENOENT;

media/libstagefright/OMXClient.cpp

diff --git a/media/libstagefright/OMXClient.cpp b/media/libstagefright/OMXClient.cpp
index 9820ef5..9f9352d 100644
--- a/media/libstagefright/OMXClient.cpp
+++ b/media/libstagefright/OMXClient.cpp
@@ -69,6 +69,10 @@
     virtual status_t storeMetaDataInBuffers(
             node_id node, OMX_U32 port_index, OMX_BOOL enable);
 
+    virtual status_t prepareForAdaptivePlayback(
+            node_id node, OMX_U32 port_index, OMX_BOOL enable,
+            OMX_U32 maxFrameWidth, OMX_U32 maxFrameHeight);
+
     virtual status_t enableGraphicBuffers(
             node_id node, OMX_U32 port_index, OMX_BOOL enable);
 
@@ -268,6 +272,13 @@
     return getOMX(node)->storeMetaDataInBuffers(node, port_index, enable);
 }
 
+status_t MuxOMX::prepareForAdaptivePlayback(
+        node_id node, OMX_U32 port_index, OMX_BOOL enable,
+        OMX_U32 maxFrameWidth, OMX_U32 maxFrameHeight) {
+    return getOMX(node)->prepareForAdaptivePlayback(
+            node, port_index, enable, maxFrameWidth, maxFrameHeight);
+}
+
 status_t MuxOMX::enableGraphicBuffers(
         node_id node, OMX_U32 port_index, OMX_BOOL enable) {
     return getOMX(node)->enableGraphicBuffers(node, port_index, enable);

media/libstagefright/OMXCodec.cpp

diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp
index 7b37365..4d3b5bd 100644
--- a/media/libstagefright/OMXCodec.cpp
+++ b/media/libstagefright/OMXCodec.cpp
@@ -40,7 +40,9 @@
 #include <utils/Vector.h>
 
 #include <OMX_Audio.h>
+#include <OMX_AudioExt.h>
 #include <OMX_Component.h>
+#include <OMX_IndexExt.h>
 
 #include "include/avc_utils.h"
 
@@ -359,12 +361,7 @@
             observer->setCodec(codec);
 
             err = codec->configureCodec(meta);
-
             if (err == OK) {
-                if (!strcmp("OMX.Nvidia.mpeg2v.decode", componentName)) {
-                    codec->mFlags |= kOnlySubmitOneInputBufferAtOneTime;
-                }
-
                 return codec;
             }
 
@@ -492,6 +489,13 @@
 
             CHECK(meta->findData(kKeyVorbisBooks, &type, &data, &size));
             addCodecSpecificData(data, size);
+        } else if (meta->findData(kKeyOpusHeader, &type, &data, &size)) {
+            addCodecSpecificData(data, size);
+
+            CHECK(meta->findData(kKeyOpusCodecDelay, &type, &data, &size));
+            addCodecSpecificData(data, size);
+            CHECK(meta->findData(kKeyOpusSeekPreRoll, &type, &data, &size));
+            addCodecSpecificData(data, size);
         }
     }
 
@@ -533,6 +537,17 @@
                     sampleRate,
                     numChannels);
         }
+    } else if (!strcasecmp(MEDIA_MIMETYPE_AUDIO_AC3, mMIME)) {
+        int32_t numChannels;
+        int32_t sampleRate;
+        CHECK(meta->findInt32(kKeyChannelCount, &numChannels));
+        CHECK(meta->findInt32(kKeySampleRate, &sampleRate));
+
+        status_t err = setAC3Format(numChannels, sampleRate);
+        if (err != OK) {
+            CODEC_LOGE("setAC3Format() failed (err = %d)", err);
+            return err;
+        }
     } else if (!strcasecmp(MEDIA_MIMETYPE_AUDIO_G711_ALAW, mMIME)
             || !strcasecmp(MEDIA_MIMETYPE_AUDIO_G711_MLAW, mMIME)) {
         // These are PCM-like formats with a fixed sample rate but
@@ -1346,8 +1361,7 @@
       mLeftOverBuffer(NULL),
       mPaused(false),
       mNativeWindow(
-              (!strncmp(componentName, "OMX.google.", 11)
-              || !strcmp(componentName, "OMX.Nvidia.mpeg2v.decode"))
+              (!strncmp(componentName, "OMX.google.", 11))
                         ? NULL : nativeWindow) {
     mPortStatus[kPortIndexInput] = ENABLED;
     mPortStatus[kPortIndexOutput] = ENABLED;
@@ -1380,6 +1394,8 @@
             "audio_decoder.aac", "audio_encoder.aac" },
         { MEDIA_MIMETYPE_AUDIO_VORBIS,
             "audio_decoder.vorbis", "audio_encoder.vorbis" },
+        { MEDIA_MIMETYPE_AUDIO_OPUS,
+            "audio_decoder.opus", "audio_encoder.opus" },
         { MEDIA_MIMETYPE_AUDIO_G711_MLAW,
             "audio_decoder.g711mlaw", "audio_encoder.g711mlaw" },
         { MEDIA_MIMETYPE_AUDIO_G711_ALAW,
@@ -1400,6 +1416,10 @@
             "audio_decoder.flac", "audio_encoder.flac" },
         { MEDIA_MIMETYPE_AUDIO_MSGSM,
             "audio_decoder.gsm", "audio_encoder.gsm" },
+        { MEDIA_MIMETYPE_VIDEO_MPEG2,
+            "video_decoder.mpeg2", "video_encoder.mpeg2" },
+        { MEDIA_MIMETYPE_AUDIO_AC3,
+            "audio_decoder.ac3", "audio_encoder.ac3" },
     };
 
     static const size_t kNumMimeToRole =
@@ -3495,6 +3515,31 @@
     return OK;
 }
 
+status_t OMXCodec::setAC3Format(int32_t numChannels, int32_t sampleRate) {
+    OMX_AUDIO_PARAM_ANDROID_AC3TYPE def;
+    InitOMXParams(&def);
+    def.nPortIndex = kPortIndexInput;
+
+    status_t err = mOMX->getParameter(
+            mNode,
+            (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3,
+            &def,
+            sizeof(def));
+
+    if (err != OK) {
+        return err;
+    }
+
+    def.nChannels = numChannels;
+    def.nSampleRate = sampleRate;
+
+    return mOMX->setParameter(
+            mNode,
+            (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3,
+            &def,
+            sizeof(def));
+}
+
 void OMXCodec::setG711Format(int32_t numChannels) {
     CHECK(!mIsEncoder);
     setRawAudioFormat(kPortIndexInput, 8000, numChannels);
@@ -4089,6 +4134,7 @@
         "OMX_AUDIO_CodingMP3",
         "OMX_AUDIO_CodingSBC",
         "OMX_AUDIO_CodingVORBIS",
+        "OMX_AUDIO_CodingOPUS",
         "OMX_AUDIO_CodingWMA",
         "OMX_AUDIO_CodingRA",
         "OMX_AUDIO_CodingMIDI",
@@ -4428,6 +4474,17 @@
                 mOutputFormat->setInt32(kKeyChannelCount, numChannels);
                 mOutputFormat->setInt32(kKeySampleRate, sampleRate);
                 mOutputFormat->setInt32(kKeyBitRate, bitRate);
+            } else if (audio_def->eEncoding ==
+                    (OMX_AUDIO_CODINGTYPE)OMX_AUDIO_CodingAndroidAC3) {
+                mOutputFormat->setCString(
+                        kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AC3);
+                int32_t numChannels, sampleRate, bitRate;
+                inputFormat->findInt32(kKeyChannelCount, &numChannels);
+                inputFormat->findInt32(kKeySampleRate, &sampleRate);
+                inputFormat->findInt32(kKeyBitRate, &bitRate);
+                mOutputFormat->setInt32(kKeyChannelCount, numChannels);
+                mOutputFormat->setInt32(kKeySampleRate, sampleRate);
+                mOutputFormat->setInt32(kKeyBitRate, bitRate);
             } else {
                 CHECK(!"Should not be here. Unknown audio encoding.");
             }
@@ -4585,12 +4642,6 @@
     caps->mFlags = 0;
     caps->mComponentName = componentName;
 
-    if (!isEncoder && !strncmp(mime, "video/", 6) &&
-            omx->storeMetaDataInBuffers(
-                    node, 1 /* port index */, OMX_TRUE) == OK) {
-        caps->mFlags |= CodecCapabilities::kFlagSupportsAdaptivePlayback;
-    }
-
     OMX_VIDEO_PARAM_PROFILELEVELTYPE param;
     InitOMXParams(&param);
 
@@ -4626,6 +4677,16 @@
         caps->mColorFormats.push(portFormat.eColorFormat);
     }
 
+    if (!isEncoder && !strncmp(mime, "video/", 6)) {
+        if (omx->storeMetaDataInBuffers(
+                    node, 1 /* port index */, OMX_TRUE) == OK ||
+            omx->prepareForAdaptivePlayback(
+                    node, 1 /* port index */, OMX_TRUE,
+                    1280 /* width */, 720 /* height */) == OK) {
+            caps->mFlags |= CodecCapabilities::kFlagSupportsAdaptivePlayback;
+        }
+    }
+
     CHECK_EQ(omx->freeNode(node), (status_t)OK);
 
     return OK;

media/libstagefright/OggExtractor.cpp

diff --git a/media/libstagefright/OggExtractor.cpp b/media/libstagefright/OggExtractor.cpp
index 5e79e78..f3eeb03 100644
--- a/media/libstagefright/OggExtractor.cpp
+++ b/media/libstagefright/OggExtractor.cpp
@@ -151,7 +151,7 @@
     return mExtractor->mImpl->getFormat();
 }
 
-status_t OggSource::start(MetaData *params) {
+status_t OggSource::start(MetaData * /* params */) {
     if (mStarted) {
         return INVALID_OPERATION;
     }
@@ -381,7 +381,7 @@
     ssize_t n;
     if ((n = mSource->readAt(offset, header, sizeof(header)))
             < (ssize_t)sizeof(header)) {
-        ALOGV("failed to read %d bytes at offset 0x%016llx, got %ld bytes",
+        ALOGV("failed to read %zu bytes at offset 0x%016llx, got %d bytes",
              sizeof(header), offset, n);
 
         if (n < 0) {
@@ -505,7 +505,7 @@
                     packetSize);
 
             if (n < (ssize_t)packetSize) {
-                ALOGV("failed to read %d bytes at 0x%016llx, got %ld bytes",
+                ALOGV("failed to read %zu bytes at 0x%016llx, got %d bytes",
                      packetSize, dataOffset, n);
                 return ERROR_IO;
             }
@@ -546,7 +546,7 @@
                 buffer = NULL;
             }
 
-            ALOGV("readPage returned %ld", n);
+            ALOGV("readPage returned %d", n);
 
             return n < 0 ? n : (status_t)ERROR_END_OF_STREAM;
         }
@@ -998,7 +998,7 @@
 }
 
 sp<MetaData> OggExtractor::getTrackMetaData(
-        size_t index, uint32_t flags) {
+        size_t index, uint32_t /* flags */) {
     if (index >= 1) {
         return NULL;
     }

media/libstagefright/StagefrightMediaScanner.cpp

diff --git a/media/libstagefright/StagefrightMediaScanner.cpp b/media/libstagefright/StagefrightMediaScanner.cpp
index af8186c..fe20835 100644
--- a/media/libstagefright/StagefrightMediaScanner.cpp
+++ b/media/libstagefright/StagefrightMediaScanner.cpp
@@ -24,6 +24,7 @@
 
 #include <media/stagefright/StagefrightMediaScanner.h>
 
+#include <media/IMediaHTTPService.h>
 #include <media/mediametadataretriever.h>
 #include <private/media/VideoFrame.h>
 
@@ -117,7 +118,7 @@
 }
 
 MediaScanResult StagefrightMediaScanner::processFileInternal(
-        const char *path, const char *mimeType,
+        const char *path, const char * /* mimeType */,
         MediaScannerClient &client) {
     const char *extension = strrchr(path, '.');
 
@@ -147,7 +148,7 @@
     status_t status;
     if (fd < 0) {
         // couldn't open it locally, maybe the media server can?
-        status = mRetriever->setDataSource(path);
+        status = mRetriever->setDataSource(NULL /* httpService */, path);
     } else {
         status = mRetriever->setDataSource(fd, 0, 0x7ffffffffffffffL);
         close(fd);

media/libstagefright/StagefrightMetadataRetriever.cpp

diff --git a/media/libstagefright/StagefrightMetadataRetriever.cpp b/media/libstagefright/StagefrightMetadataRetriever.cpp
index 19af4fb..9475d05 100644
--- a/media/libstagefright/StagefrightMetadataRetriever.cpp
+++ b/media/libstagefright/StagefrightMetadataRetriever.cpp
@@ -16,10 +16,12 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "StagefrightMetadataRetriever"
+#include <inttypes.h>
 #include <utils/Log.h>
 
 #include "include/StagefrightMetadataRetriever.h"
 
+#include <media/IMediaHTTPService.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/ColorConverter.h>
 #include <media/stagefright/DataSource.h>
@@ -50,7 +52,9 @@
 }
 
 status_t StagefrightMetadataRetriever::setDataSource(
-        const char *uri, const KeyedVector<String8, String8> *headers) {
+        const sp<IMediaHTTPService> &httpService,
+        const char *uri,
+        const KeyedVector<String8, String8> *headers) {
     ALOGV("setDataSource(%s)", uri);
 
     mParsedMetaData = false;
@@ -58,7 +62,7 @@
     delete mAlbumArt;
     mAlbumArt = NULL;
 
-    mSource = DataSource::CreateFromURI(uri, headers);
+    mSource = DataSource::CreateFromURI(httpService, uri, headers);
 
     if (mSource == NULL) {
         ALOGE("Unable to create data source for '%s'.", uri);
@@ -488,7 +492,7 @@
     size_t numTracks = mExtractor->countTracks();
 
     char tmp[32];
-    sprintf(tmp, "%d", numTracks);
+    sprintf(tmp, "%zu", numTracks);
 
     mMetaData.add(METADATA_KEY_NUM_TRACKS, String8(tmp));
 
@@ -545,7 +549,7 @@
     }
 
     // The duration value is a string representing the duration in ms.
-    sprintf(tmp, "%lld", (maxDurationUs + 500) / 1000);
+    sprintf(tmp, "%" PRId64, (maxDurationUs + 500) / 1000);
     mMetaData.add(METADATA_KEY_DURATION, String8(tmp));
 
     if (hasAudio) {
@@ -573,7 +577,7 @@
         if (mSource->getSize(&sourceSize) == OK) {
             int64_t avgBitRate = (int64_t)(sourceSize * 8E6 / maxDurationUs);
 
-            sprintf(tmp, "%lld", avgBitRate);
+            sprintf(tmp, "%" PRId64, avgBitRate);
             mMetaData.add(METADATA_KEY_BITRATE, String8(tmp));
         }
     }

media/libstagefright/SurfaceMediaSource.cpp

diff --git a/media/libstagefright/SurfaceMediaSource.cpp b/media/libstagefright/SurfaceMediaSource.cpp
index 6b934d4..10c00f4 100644
--- a/media/libstagefright/SurfaceMediaSource.cpp
+++ b/media/libstagefright/SurfaceMediaSource.cpp
@@ -99,8 +99,11 @@
     dump(result, "", buffer, 1024);
 }
 
-void SurfaceMediaSource::dump(String8& result, const char* prefix,
-        char* buffer, size_t SIZE) const
+void SurfaceMediaSource::dump(
+        String8& result,
+        const char* /* prefix */,
+        char* buffer,
+        size_t /* SIZE */) const
 {
     Mutex::Autolock lock(mMutex);
 
@@ -202,6 +205,9 @@
         return OK;
     }
 
+    mStarted = false;
+    mFrameAvailableCondition.signal();
+
     while (mNumPendingBuffers > 0) {
         ALOGI("Still waiting for %d buffers to be returned.",
                 mNumPendingBuffers);
@@ -215,8 +221,6 @@
         mMediaBuffersAvailableCondition.wait(mMutex);
     }
 
-    mStarted = false;
-    mFrameAvailableCondition.signal();
     mMediaBuffersAvailableCondition.signal();
 
     return mBufferQueue->consumerDisconnect();
@@ -269,9 +273,8 @@
             bufferHandle, (*buffer)->range_length(), (*buffer)->range_offset());
 }
 
-status_t SurfaceMediaSource::read( MediaBuffer **buffer,
-                                    const ReadOptions *options)
-{
+status_t SurfaceMediaSource::read(
+        MediaBuffer **buffer, const ReadOptions * /* options */) {
     ALOGV("read");
     Mutex::Autolock lock(mMutex);
 

media/libstagefright/TimedEventQueue.cpp

diff --git a/media/libstagefright/TimedEventQueue.cpp b/media/libstagefright/TimedEventQueue.cpp
index 7e9c4bf..3d2eb1f 100644
--- a/media/libstagefright/TimedEventQueue.cpp
+++ b/media/libstagefright/TimedEventQueue.cpp
@@ -31,17 +31,29 @@
 
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/ALooper.h>
+#include <binder/IServiceManager.h>
+#include <powermanager/PowerManager.h>
+#include <binder/IPCThreadState.h>
+#include <utils/CallStack.h>
 
 namespace android {
 
+static int64_t kWakelockMinDelay = 100000ll;  // 100ms
+
 TimedEventQueue::TimedEventQueue()
     : mNextEventID(1),
       mRunning(false),
-      mStopped(false) {
+      mStopped(false),
+      mDeathRecipient(new PMDeathRecipient(this)),
+      mWakeLockCount(0) {
 }
 
 TimedEventQueue::~TimedEventQueue() {
     stop();
+    if (mPowerManager != 0) {
+        sp<IBinder> binder = mPowerManager->asBinder();
+        binder->unlinkToDeath(mDeathRecipient);
+    }
 }
 
 void TimedEventQueue::start() {
@@ -76,6 +88,9 @@
     void *dummy;
     pthread_join(mThread, &dummy);
 
+    // some events may be left in the queue if we did not flush and the wake lock
+    // must be released.
+    releaseWakeLock_l(true /*force*/);
     mQueue.clear();
 
     mRunning = false;
@@ -112,11 +127,16 @@
     QueueItem item;
     item.event = event;
     item.realtime_us = realtime_us;
+    item.has_wakelock = false;
 
     if (it == mQueue.begin()) {
         mQueueHeadChangedCondition.signal();
     }
 
+    if (realtime_us > ALooper::GetNowUs() + kWakelockMinDelay) {
+        acquireWakeLock_l();
+        item.has_wakelock = true;
+    }
     mQueue.insert(it, item);
 
     mQueueNotEmptyCondition.signal();
@@ -171,8 +191,10 @@
         ALOGV("cancelling event %d", (*it).event->eventID());
 
         (*it).event->setEventID(0);
+        if ((*it).has_wakelock) {
+            releaseWakeLock_l();
+        }
         it = mQueue.erase(it);
-
         if (stopAfterFirstMatch) {
             return;
         }
@@ -195,6 +217,7 @@
     for (;;) {
         int64_t now_us = 0;
         sp<Event> event;
+        bool wakeLocked = false;
 
         {
             Mutex::Autolock autoLock(mLock);
@@ -261,26 +284,29 @@
             // removeEventFromQueue_l will return NULL.
             // Otherwise, the QueueItem will be removed
             // from the queue and the referenced event returned.
-            event = removeEventFromQueue_l(eventID);
+            event = removeEventFromQueue_l(eventID, &wakeLocked);
         }
 
         if (event != NULL) {
             // Fire event with the lock NOT held.
             event->fire(this, now_us);
+            if (wakeLocked) {
+                Mutex::Autolock autoLock(mLock);
+                releaseWakeLock_l();
+            }
         }
     }
 }
 
 sp<TimedEventQueue::Event> TimedEventQueue::removeEventFromQueue_l(
-        event_id id) {
+        event_id id, bool *wakeLocked) {
     for (List<QueueItem>::iterator it = mQueue.begin();
          it != mQueue.end(); ++it) {
         if ((*it).event->eventID() == id) {
             sp<Event> event = (*it).event;
             event->setEventID(0);
-
+            *wakeLocked = (*it).has_wakelock;
             mQueue.erase(it);
-
             return event;
         }
     }
@@ -290,5 +316,70 @@
     return NULL;
 }
 
+void TimedEventQueue::acquireWakeLock_l()
+{
+    if (mWakeLockCount == 0) {
+        CHECK(mWakeLockToken == 0);
+        if (mPowerManager == 0) {
+            // use checkService() to avoid blocking if power service is not up yet
+            sp<IBinder> binder =
+                defaultServiceManager()->checkService(String16("power"));
+            if (binder == 0) {
+                ALOGW("cannot connect to the power manager service");
+            } else {
+                mPowerManager = interface_cast<IPowerManager>(binder);
+                binder->linkToDeath(mDeathRecipient);
+            }
+        }
+        if (mPowerManager != 0) {
+            sp<IBinder> binder = new BBinder();
+            int64_t token = IPCThreadState::self()->clearCallingIdentity();
+            status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
+                                                             binder,
+                                                             String16("TimedEventQueue"),
+                                                             String16("media"));
+            IPCThreadState::self()->restoreCallingIdentity(token);
+            if (status == NO_ERROR) {
+                mWakeLockToken = binder;
+                mWakeLockCount++;
+            }
+        }
+    } else {
+        mWakeLockCount++;
+    }
+}
+
+void TimedEventQueue::releaseWakeLock_l(bool force)
+{
+    if (mWakeLockCount == 0) {
+        return;
+    }
+    if (force) {
+        // Force wakelock release below by setting reference count to 1.
+        mWakeLockCount = 1;
+    }
+    if (--mWakeLockCount == 0) {
+        CHECK(mWakeLockToken != 0);
+        if (mPowerManager != 0) {
+            int64_t token = IPCThreadState::self()->clearCallingIdentity();
+            mPowerManager->releaseWakeLock(mWakeLockToken, 0);
+            IPCThreadState::self()->restoreCallingIdentity(token);
+        }
+        mWakeLockToken.clear();
+    }
+}
+
+void TimedEventQueue::clearPowerManager()
+{
+    Mutex::Autolock _l(mLock);
+    releaseWakeLock_l(true /*force*/);
+    mPowerManager.clear();
+}
+
+void TimedEventQueue::PMDeathRecipient::binderDied(
+        const wp<IBinder>& /* who */) {
+    mQueue->clearPowerManager();
+}
+
 }  // namespace android
 

media/libstagefright/Utils.cpp

diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index 4db8e80..4ff805f 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -251,6 +251,13 @@
         buffer->meta()->setInt32("csd", true);
         buffer->meta()->setInt64("timeUs", 0);
         msg->setBuffer("csd-1", buffer);
+    } else if (meta->findData(kKeyOpusHeader, &type, &data, &size)) {
+        sp<ABuffer> buffer = new ABuffer(size);
+        memcpy(buffer->data(), data, size);
+
+        buffer->meta()->setInt32("csd", true);
+        buffer->meta()->setInt64("timeUs", 0);
+        msg->setBuffer("csd-0", buffer);
     }
 
     *format = msg;
@@ -452,6 +459,11 @@
         }
     }
 
+    int32_t timeScale;
+    if (msg->findInt32("time-scale", &timeScale)) {
+        meta->setInt32(kKeyTimeScale, timeScale);
+    }
+
     // XXX TODO add whatever other keys there are
 
 #if 0
@@ -523,6 +535,7 @@
     { MEDIA_MIMETYPE_AUDIO_AMR_WB,      AUDIO_FORMAT_AMR_WB },
     { MEDIA_MIMETYPE_AUDIO_AAC,         AUDIO_FORMAT_AAC },
     { MEDIA_MIMETYPE_AUDIO_VORBIS,      AUDIO_FORMAT_VORBIS },
+    { MEDIA_MIMETYPE_AUDIO_OPUS,        AUDIO_FORMAT_OPUS},
     { 0, AUDIO_FORMAT_INVALID }
 };
 
@@ -540,7 +553,8 @@
     return BAD_VALUE;
 }
 
-bool canOffloadStream(const sp<MetaData>& meta, bool hasVideo, bool isStreaming)
+bool canOffloadStream(const sp<MetaData>& meta, bool hasVideo,
+                      bool isStreaming, audio_stream_type_t streamType)
 {
     const char *mime;
     CHECK(meta->findCString(kKeyMIMEType, &mime));
@@ -561,6 +575,17 @@
         return false;
     }
 
+    // check whether it is ELD/LD content -> no offloading
+    // FIXME: this should depend on audio DSP capabilities. mapMimeToAudioFormat() should use the
+    // metadata to refine the AAC format and the audio HAL should only list supported profiles.
+    int32_t aacaot = -1;
+    if (meta->findInt32(kKeyAACAOT, &aacaot)) {
+        if (aacaot == 23 || aacaot == 39 ) {
+            ALOGV("track of type '%s' is ELD/LD content", mime);
+            return false;
+        }
+    }
+
     int32_t srate = -1;
     if (!meta->findInt32(kKeySampleRate, &srate)) {
         ALOGV("track of type '%s' does not publish sample rate", mime);
@@ -594,7 +619,7 @@
     info.bit_rate = brate;
 
 
-    info.stream_type = AUDIO_STREAM_MUSIC;
+    info.stream_type = streamType;
     info.has_video = hasVideo;
     info.is_streaming = isStreaming;
 

media/libstagefright/VBRISeeker.cpp

diff --git a/media/libstagefright/VBRISeeker.cpp b/media/libstagefright/VBRISeeker.cpp
index a245f2c..af858b9 100644
--- a/media/libstagefright/VBRISeeker.cpp
+++ b/media/libstagefright/VBRISeeker.cpp
@@ -119,7 +119,7 @@
 
         seeker->mSegments.push(numBytes);
 
-        ALOGV("entry #%d: %d offset 0x%08lx", i, numBytes, offset);
+        ALOGV("entry #%d: %u offset 0x%016llx", i, numBytes, offset);
         offset += numBytes;
     }
 
@@ -160,7 +160,7 @@
         *pos += mSegments.itemAt(segmentIndex++);
     }
 
-    ALOGV("getOffsetForTime %lld us => 0x%08lx", *timeUs, *pos);
+    ALOGV("getOffsetForTime %lld us => 0x%016llx", *timeUs, *pos);
 
     *timeUs = nowUs;
 

media/libstagefright/WAVExtractor.cpp

diff --git a/media/libstagefright/WAVExtractor.cpp b/media/libstagefright/WAVExtractor.cpp
index 22af6fb..fe9058b 100644
--- a/media/libstagefright/WAVExtractor.cpp
+++ b/media/libstagefright/WAVExtractor.cpp
@@ -127,7 +127,7 @@
 }
 
 sp<MetaData> WAVExtractor::getTrackMetaData(
-        size_t index, uint32_t flags) {
+        size_t index, uint32_t /* flags */) {
     if (mInitCheck != OK || index > 0) {
         return NULL;
     }
@@ -358,7 +358,7 @@
     }
 }
 
-status_t WAVSource::start(MetaData *params) {
+status_t WAVSource::start(MetaData * /* params */) {
     ALOGV("WAVSource::start");
 
     CHECK(!mStarted);

media/libstagefright/WVMExtractor.cpp

diff --git a/media/libstagefright/WVMExtractor.cpp b/media/libstagefright/WVMExtractor.cpp
index 5ae80cc..bc48272 100644
--- a/media/libstagefright/WVMExtractor.cpp
+++ b/media/libstagefright/WVMExtractor.cpp
@@ -76,7 +76,7 @@
 {
     gVendorLibHandle = dlopen("libwvm.so", RTLD_NOW);
     if (gVendorLibHandle == NULL) {
-        ALOGE("Failed to open libwvm.so");
+        ALOGE("Failed to open libwvm.so: %s", dlerror());
     }
 }
 

media/libstagefright/avc_utils.cpp

diff --git a/media/libstagefright/avc_utils.cpp b/media/libstagefright/avc_utils.cpp
index b822868..38a1f6b 100644
--- a/media/libstagefright/avc_utils.cpp
+++ b/media/libstagefright/avc_utils.cpp
@@ -40,6 +40,25 @@
     return x + (1u << numZeroes) - 1;
 }
 
+signed parseSE(ABitReader *br) {
+    unsigned codeNum = parseUE(br);
+
+    return (codeNum & 1) ? (codeNum + 1) / 2 : -(codeNum / 2);
+}
+
+static void skipScalingList(ABitReader *br, size_t sizeOfScalingList) {
+    size_t lastScale = 8;
+    size_t nextScale = 8;
+    for (size_t j = 0; j < sizeOfScalingList; ++j) {
+        if (nextScale != 0) {
+            signed delta_scale = parseSE(br);
+            nextScale = (lastScale + delta_scale + 256) % 256;
+        }
+
+        lastScale = (nextScale == 0) ? lastScale : nextScale;
+    }
+}
+
 // Determine video dimensions from the sequence parameterset.
 void FindAVCDimensions(
         const sp<ABuffer> &seqParamSet,
@@ -63,7 +82,24 @@
         parseUE(&br);  // bit_depth_luma_minus8
         parseUE(&br);  // bit_depth_chroma_minus8
         br.skipBits(1);  // qpprime_y_zero_transform_bypass_flag
-        CHECK_EQ(br.getBits(1), 0u);  // seq_scaling_matrix_present_flag
+
+        if (br.getBits(1)) {  // seq_scaling_matrix_present_flag
+            for (size_t i = 0; i < 8; ++i) {
+                if (br.getBits(1)) {  // seq_scaling_list_present_flag[i]
+
+                    // WARNING: the code below has not ever been exercised...
+                    // need a real-world example.
+
+                    if (i < 6) {
+                        // ScalingList4x4[i],16,...
+                        skipScalingList(&br, 16);
+                    } else {
+                        // ScalingList8x8[i-6],64,...
+                        skipScalingList(&br, 64);
+                    }
+                }
+            }
+        }
     }
 
     parseUE(&br);  // log2_max_frame_num_minus4
@@ -251,9 +287,7 @@
     return OK;
 }
 
-static sp<ABuffer> FindNAL(
-        const uint8_t *data, size_t size, unsigned nalType,
-        size_t *stopOffset) {
+static sp<ABuffer> FindNAL(const uint8_t *data, size_t size, unsigned nalType) {
     const uint8_t *nalStart;
     size_t nalSize;
     while (getNextNALUnit(&data, &size, &nalStart, &nalSize, true) == OK) {
@@ -293,7 +327,7 @@
     const uint8_t *data = accessUnit->data();
     size_t size = accessUnit->size();
 
-    sp<ABuffer> seqParamSet = FindNAL(data, size, 7, NULL);
+    sp<ABuffer> seqParamSet = FindNAL(data, size, 7);
     if (seqParamSet == NULL) {
         return NULL;
     }
@@ -303,8 +337,7 @@
     FindAVCDimensions(
             seqParamSet, &width, &height, &sarWidth, &sarHeight);
 
-    size_t stopOffset;
-    sp<ABuffer> picParamSet = FindNAL(data, size, 8, &stopOffset);
+    sp<ABuffer> picParamSet = FindNAL(data, size, 8);
     CHECK(picParamSet != NULL);
 
     size_t csdSize =

media/libstagefright/chromium_http/Android.mk

diff --git a/media/libstagefright/chromium_http/Android.mk b/media/libstagefright/chromium_http/Android.mk
deleted file mode 100644
index f26f386..0000000
--- a/media/libstagefright/chromium_http/Android.mk
+++ /dev/null
@@ -1,37 +0,0 @@
-LOCAL_PATH:= $(call my-dir)
-
-ifneq ($(TARGET_BUILD_PDK), true)
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES:=       \
-        DataUriSource.cpp \
-        ChromiumHTTPDataSource.cpp \
-        support.cpp \
-        chromium_http_stub.cpp
-
-LOCAL_C_INCLUDES:= \
-        $(TOP)/frameworks/av/media/libstagefright \
-        $(TOP)/frameworks/native/include/media/openmax \
-        external/chromium \
-        external/chromium/android
-
-LOCAL_CFLAGS += -Wno-multichar
-
-LOCAL_SHARED_LIBRARIES += \
-        libstlport \
-        libchromium_net \
-        libutils \
-        libcutils \
-        liblog \
-        libstagefright_foundation \
-        libstagefright \
-        libdrmframework
-
-include external/stlport/libstlport.mk
-
-LOCAL_MODULE:= libstagefright_chromium_http
-
-LOCAL_MODULE_TAGS := optional
-
-include $(BUILD_SHARED_LIBRARY)
-endif

media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp

diff --git a/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp b/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp
deleted file mode 100644
index a862d8b..0000000
--- a/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp
+++ /dev/null
@@ -1,350 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "ChromiumHTTPDataSource"
-#include <media/stagefright/foundation/ADebug.h>
-
-#include "include/ChromiumHTTPDataSource.h"
-
-#include <media/stagefright/foundation/ALooper.h>
-#include <media/stagefright/MediaErrors.h>
-
-#include "support.h"
-
-#include <cutils/properties.h> // for property_get
-
-namespace android {
-
-ChromiumHTTPDataSource::ChromiumHTTPDataSource(uint32_t flags)
-    : mFlags(flags),
-      mState(DISCONNECTED),
-      mDelegate(new SfDelegate),
-      mCurrentOffset(0),
-      mIOResult(OK),
-      mContentSize(-1),
-      mDecryptHandle(NULL),
-      mDrmManagerClient(NULL) {
-    mDelegate->setOwner(this);
-}
-
-ChromiumHTTPDataSource::~ChromiumHTTPDataSource() {
-    disconnect();
-
-    delete mDelegate;
-    mDelegate = NULL;
-
-    clearDRMState_l();
-
-    if (mDrmManagerClient != NULL) {
-        delete mDrmManagerClient;
-        mDrmManagerClient = NULL;
-    }
-}
-
-status_t ChromiumHTTPDataSource::connect(
-        const char *uri,
-        const KeyedVector<String8, String8> *headers,
-        off64_t offset) {
-    Mutex::Autolock autoLock(mLock);
-
-    uid_t uid;
-    if (getUID(&uid)) {
-        mDelegate->setUID(uid);
-    }
-
-#if defined(LOG_NDEBUG) && !LOG_NDEBUG
-    LOG_PRI(ANDROID_LOG_VERBOSE, LOG_TAG, "connect on behalf of uid %d", uid);
-#endif
-
-    return connect_l(uri, headers, offset);
-}
-
-status_t ChromiumHTTPDataSource::connect_l(
-        const char *uri,
-        const KeyedVector<String8, String8> *headers,
-        off64_t offset) {
-    if (mState != DISCONNECTED) {
-        disconnect_l();
-    }
-
-#if defined(LOG_NDEBUG) && !LOG_NDEBUG
-    LOG_PRI(ANDROID_LOG_VERBOSE, LOG_TAG,
-                "connect to <URL suppressed> @%lld", offset);
-#endif
-
-    mURI = uri;
-    mContentType = String8("application/octet-stream");
-
-    if (headers != NULL) {
-        mHeaders = *headers;
-    } else {
-        mHeaders.clear();
-    }
-
-    mState = CONNECTING;
-    mContentSize = -1;
-    mCurrentOffset = offset;
-
-    mDelegate->initiateConnection(mURI.c_str(), &mHeaders, offset);
-
-    while (mState == CONNECTING || mState == DISCONNECTING) {
-        mCondition.wait(mLock);
-    }
-
-    return mState == CONNECTED ? OK : mIOResult;
-}
-
-void ChromiumHTTPDataSource::onConnectionEstablished(
-        int64_t contentSize, const char *contentType) {
-    Mutex::Autolock autoLock(mLock);
-
-    if (mState != CONNECTING) {
-        // We may have initiated disconnection.
-        CHECK_EQ(mState, DISCONNECTING);
-        return;
-    }
-
-    mState = CONNECTED;
-    mContentSize = (contentSize < 0) ? -1 : contentSize + mCurrentOffset;
-    mContentType = String8(contentType);
-    mCondition.broadcast();
-}
-
-void ChromiumHTTPDataSource::onConnectionFailed(status_t err) {
-    Mutex::Autolock autoLock(mLock);
-    mState = DISCONNECTED;
-    mCondition.broadcast();
-
-    // mURI.clear();
-
-    mIOResult = err;
-}
-
-void ChromiumHTTPDataSource::disconnect() {
-    Mutex::Autolock autoLock(mLock);
-    disconnect_l();
-}
-
-void ChromiumHTTPDataSource::disconnect_l() {
-    if (mState == DISCONNECTED) {
-        return;
-    }
-
-    mState = DISCONNECTING;
-    mIOResult = -EINTR;
-
-    mDelegate->initiateDisconnect();
-
-    while (mState == DISCONNECTING) {
-        mCondition.wait(mLock);
-    }
-
-    CHECK_EQ((int)mState, (int)DISCONNECTED);
-}
-
-status_t ChromiumHTTPDataSource::initCheck() const {
-    Mutex::Autolock autoLock(mLock);
-
-    return mState == CONNECTED ? OK : NO_INIT;
-}
-
-ssize_t ChromiumHTTPDataSource::readAt(off64_t offset, void *data, size_t size) {
-    Mutex::Autolock autoLock(mLock);
-
-    if (mState != CONNECTED) {
-        return INVALID_OPERATION;
-    }
-
-#if 0
-    char value[PROPERTY_VALUE_MAX];
-    if (property_get("media.stagefright.disable-net", value, 0)
-            && (!strcasecmp(value, "true") || !strcmp(value, "1"))) {
-        LOG_PRI(ANDROID_LOG_INFO, LOG_TAG, "Simulating that the network is down.");
-        disconnect_l();
-        return ERROR_IO;
-    }
-#endif
-
-    if (offset != mCurrentOffset) {
-        AString tmp = mURI;
-        KeyedVector<String8, String8> tmpHeaders = mHeaders;
-
-        disconnect_l();
-
-        status_t err = connect_l(tmp.c_str(), &tmpHeaders, offset);
-
-        if (err != OK) {
-            return err;
-        }
-    }
-
-    mState = READING;
-
-    int64_t startTimeUs = ALooper::GetNowUs();
-
-    mDelegate->initiateRead(data, size);
-
-    while (mState == READING) {
-        mCondition.wait(mLock);
-    }
-
-    if (mIOResult < OK) {
-        return mIOResult;
-    }
-
-    if (mState == CONNECTED) {
-        int64_t delayUs = ALooper::GetNowUs() - startTimeUs;
-
-        // The read operation was successful, mIOResult contains
-        // the number of bytes read.
-        addBandwidthMeasurement(mIOResult, delayUs);
-
-        mCurrentOffset += mIOResult;
-        return mIOResult;
-    }
-
-    return ERROR_IO;
-}
-
-void ChromiumHTTPDataSource::onReadCompleted(ssize_t size) {
-    Mutex::Autolock autoLock(mLock);
-
-    mIOResult = size;
-
-    if (mState == READING) {
-        mState = CONNECTED;
-        mCondition.broadcast();
-    }
-}
-
-status_t ChromiumHTTPDataSource::getSize(off64_t *size) {
-    Mutex::Autolock autoLock(mLock);
-
-    if (mContentSize < 0) {
-        return ERROR_UNSUPPORTED;
-    }
-
-    *size = mContentSize;
-
-    return OK;
-}
-
-uint32_t ChromiumHTTPDataSource::flags() {
-    return kWantsPrefetching | kIsHTTPBasedSource;
-}
-
-// static
-void ChromiumHTTPDataSource::InitiateRead(
-        ChromiumHTTPDataSource *me, void *data, size_t size) {
-    me->initiateRead(data, size);
-}
-
-void ChromiumHTTPDataSource::initiateRead(void *data, size_t size) {
-    mDelegate->initiateRead(data, size);
-}
-
-void ChromiumHTTPDataSource::onDisconnectComplete() {
-    Mutex::Autolock autoLock(mLock);
-    CHECK_EQ((int)mState, (int)DISCONNECTING);
-
-    mState = DISCONNECTED;
-    // mURI.clear();
-    mIOResult = -ENOTCONN;
-
-    mCondition.broadcast();
-}
-
-sp<DecryptHandle> ChromiumHTTPDataSource::DrmInitialization(const char* mime) {
-    Mutex::Autolock autoLock(mLock);
-
-    if (mDrmManagerClient == NULL) {
-        mDrmManagerClient = new DrmManagerClient();
-    }
-
-    if (mDrmManagerClient == NULL) {
-        return NULL;
-    }
-
-    if (mDecryptHandle == NULL) {
-        /* Note if redirect occurs, mUri is the redirect uri instead of the
-         * original one
-         */
-        mDecryptHandle = mDrmManagerClient->openDecryptSession(
-                String8(mURI.c_str()), mime);
-    }
-
-    if (mDecryptHandle == NULL) {
-        delete mDrmManagerClient;
-        mDrmManagerClient = NULL;
-    }
-
-    return mDecryptHandle;
-}
-
-void ChromiumHTTPDataSource::getDrmInfo(
-        sp<DecryptHandle> &handle, DrmManagerClient **client) {
-    Mutex::Autolock autoLock(mLock);
-
-    handle = mDecryptHandle;
-    *client = mDrmManagerClient;
-}
-
-String8 ChromiumHTTPDataSource::getUri() {
-    Mutex::Autolock autoLock(mLock);
-
-    return String8(mURI.c_str());
-}
-
-String8 ChromiumHTTPDataSource::getMIMEType() const {
-    Mutex::Autolock autoLock(mLock);
-
-    return mContentType;
-}
-
-void ChromiumHTTPDataSource::clearDRMState_l() {
-    if (mDecryptHandle != NULL) {
-        // To release mDecryptHandle
-        CHECK(mDrmManagerClient);
-        mDrmManagerClient->closeDecryptSession(mDecryptHandle);
-        mDecryptHandle = NULL;
-    }
-}
-
-status_t ChromiumHTTPDataSource::reconnectAtOffset(off64_t offset) {
-    Mutex::Autolock autoLock(mLock);
-
-    if (mURI.empty()) {
-        return INVALID_OPERATION;
-    }
-
-    LOG_PRI(ANDROID_LOG_INFO, LOG_TAG, "Reconnecting...");
-    status_t err = connect_l(mURI.c_str(), &mHeaders, offset);
-    if (err != OK) {
-        LOG_PRI(ANDROID_LOG_INFO, LOG_TAG, "Reconnect failed w/ err 0x%08x", err);
-    }
-
-    return err;
-}
-
-// static
-status_t ChromiumHTTPDataSource::UpdateProxyConfig(
-        const char *host, int32_t port, const char *exclusionList) {
-    return SfDelegate::UpdateProxyConfig(host, port, exclusionList);
-}
-
-}  // namespace android
-

media/libstagefright/chromium_http/DataUriSource.cpp

diff --git a/media/libstagefright/chromium_http/DataUriSource.cpp b/media/libstagefright/chromium_http/DataUriSource.cpp
deleted file mode 100644
index ecf3fa1..0000000
--- a/media/libstagefright/chromium_http/DataUriSource.cpp
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <include/DataUriSource.h>
-
-#include <net/base/data_url.h>
-#include <googleurl/src/gurl.h>
-
-
-namespace android {
-
-DataUriSource::DataUriSource(const char *uri) :
-    mDataUri(uri),
-    mInited(NO_INIT) {
-
-    // Copy1: const char *uri -> String8 mDataUri.
-    std::string mimeTypeStr, unusedCharsetStr, dataStr;
-    // Copy2: String8 mDataUri -> std::string
-    const bool ret = net::DataURL::Parse(
-            GURL(std::string(mDataUri.string())),
-            &mimeTypeStr, &unusedCharsetStr, &dataStr);
-    // Copy3: std::string dataStr -> AString mData
-    mData.setTo(dataStr.data(), dataStr.length());
-    mInited = ret ? OK : UNKNOWN_ERROR;
-
-    // The chromium data url implementation defaults to using "text/plain"
-    // if no mime type is specified. We prefer to leave this unspecified
-    // instead, since the mime type is sniffed in most cases.
-    if (mimeTypeStr != "text/plain") {
-        mMimeType = mimeTypeStr.c_str();
-    }
-}
-
-ssize_t DataUriSource::readAt(off64_t offset, void *out, size_t size) {
-    if (mInited != OK) {
-        return mInited;
-    }
-
-    const off64_t length = mData.size();
-    if (offset >= length) {
-        return UNKNOWN_ERROR;
-    }
-
-    const char *dataBuf = mData.c_str();
-    const size_t bytesToCopy =
-            offset + size >= length ? (length - offset) : size;
-
-    if (bytesToCopy > 0) {
-        memcpy(out, dataBuf + offset, bytesToCopy);
-    }
-
-    return bytesToCopy;
-}
-
-}  // namespace android

media/libstagefright/chromium_http/chromium_http_stub.cpp

diff --git a/media/libstagefright/chromium_http/chromium_http_stub.cpp b/media/libstagefright/chromium_http/chromium_http_stub.cpp
deleted file mode 100644
index 289f6de..0000000
--- a/media/libstagefright/chromium_http/chromium_http_stub.cpp
+++ /dev/null
@@ -1,38 +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 <dlfcn.h>
-
-#include <include/chromium_http_stub.h>
-#include <include/ChromiumHTTPDataSource.h>
-#include <include/DataUriSource.h>
-
-namespace android {
-
-HTTPBase *createChromiumHTTPDataSource(uint32_t flags) {
-    return new ChromiumHTTPDataSource(flags);
-}
-
-status_t UpdateChromiumHTTPDataSourceProxyConfig(
-        const char *host, int32_t port, const char *exclusionList) {
-    return ChromiumHTTPDataSource::UpdateProxyConfig(host, port, exclusionList);
-}
-
-DataSource *createDataUriSource(const char *uri) {
-    return new DataUriSource(uri);
-}
-
-}

media/libstagefright/chromium_http/support.cpp

diff --git a/media/libstagefright/chromium_http/support.cpp b/media/libstagefright/chromium_http/support.cpp
deleted file mode 100644
index 0a8e3e3..0000000
--- a/media/libstagefright/chromium_http/support.cpp
+++ /dev/null
@@ -1,558 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "ChromiumHTTPDataSourceSupport"
-#include <utils/Log.h>
-
-#include <media/stagefright/foundation/AString.h>
-
-#include "support.h"
-
-#include "android/net/android_network_library_impl.h"
-#include "base/logging.h"
-#include "base/threading/thread.h"
-#include "net/base/cert_verifier.h"
-#include "net/base/cookie_monster.h"
-#include "net/base/host_resolver.h"
-#include "net/base/ssl_config_service.h"
-#include "net/http/http_auth_handler_factory.h"
-#include "net/http/http_cache.h"
-#include "net/proxy/proxy_config_service_android.h"
-
-#include "include/ChromiumHTTPDataSource.h"
-
-#include <cutils/log.h>
-#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/Utils.h>
-#include <string>
-
-namespace android {
-
-static Mutex gNetworkThreadLock;
-static base::Thread *gNetworkThread = NULL;
-static scoped_refptr<SfRequestContext> gReqContext;
-static scoped_ptr<net::NetworkChangeNotifier> gNetworkChangeNotifier;
-
-bool logMessageHandler(
-        int severity,
-        const char* file,
-        int line,
-        size_t message_start,
-        const std::string& str) {
-    int androidSeverity = ANDROID_LOG_VERBOSE;
-    switch(severity) {
-    case logging::LOG_FATAL:
-        androidSeverity = ANDROID_LOG_FATAL;
-        break;
-    case logging::LOG_ERROR_REPORT:
-    case logging::LOG_ERROR:
-        androidSeverity = ANDROID_LOG_ERROR;
-        break;
-    case logging::LOG_WARNING:
-        androidSeverity = ANDROID_LOG_WARN;
-        break;
-    default:
-        androidSeverity = ANDROID_LOG_VERBOSE;
-        break;
-    }
-    android_printLog(androidSeverity, "chromium-libstagefright",
-                    "%s:%d: %s", file, line, str.c_str());
-    return false;
-}
-
-struct AutoPrioritySaver {
-    AutoPrioritySaver()
-        : mTID(androidGetTid()),
-          mPrevPriority(androidGetThreadPriority(mTID)) {
-        androidSetThreadPriority(mTID, ANDROID_PRIORITY_NORMAL);
-    }
-
-    ~AutoPrioritySaver() {
-        androidSetThreadPriority(mTID, mPrevPriority);
-    }
-
-private:
-    pid_t mTID;
-    int mPrevPriority;
-
-    DISALLOW_EVIL_CONSTRUCTORS(AutoPrioritySaver);
-};
-
-static void InitializeNetworkThreadIfNecessary() {
-    Mutex::Autolock autoLock(gNetworkThreadLock);
-
-    if (gNetworkThread == NULL) {
-        // Make sure any threads spawned by the chromium framework are
-        // running at normal priority instead of inheriting this thread's.
-        AutoPrioritySaver saver;
-
-        gNetworkThread = new base::Thread("network");
-        base::Thread::Options options;
-        options.message_loop_type = MessageLoop::TYPE_IO;
-        CHECK(gNetworkThread->StartWithOptions(options));
-
-        gReqContext = new SfRequestContext;
-
-        gNetworkChangeNotifier.reset(net::NetworkChangeNotifier::Create());
-
-        net::AndroidNetworkLibrary::RegisterSharedInstance(
-                new SfNetworkLibrary);
-        logging::SetLogMessageHandler(logMessageHandler);
-    }
-}
-
-static void MY_LOGI(const char *s) {
-    LOG_PRI(ANDROID_LOG_INFO, LOG_TAG, "%s", s);
-}
-
-static void MY_LOGV(const char *s) {
-#if !defined(LOG_NDEBUG) || LOG_NDEBUG == 0
-    LOG_PRI(ANDROID_LOG_VERBOSE, LOG_TAG, "%s", s);
-#endif
-}
-
-SfNetLog::SfNetLog()
-    : mNextID(1) {
-}
-
-void SfNetLog::AddEntry(
-        EventType type,
-        const base::TimeTicks &time,
-        const Source &source,
-        EventPhase phase,
-        EventParameters *params) {
-#if 0
-    MY_LOGI(StringPrintf(
-                "AddEntry time=%s type=%s source=%s phase=%s\n",
-                TickCountToString(time).c_str(),
-                EventTypeToString(type),
-                SourceTypeToString(source.type),
-                EventPhaseToString(phase)).c_str());
-#endif
-}
-
-uint32 SfNetLog::NextID() {
-    return mNextID++;
-}
-
-net::NetLog::LogLevel SfNetLog::GetLogLevel() const {
-    return LOG_BASIC;
-}
-
-////////////////////////////////////////////////////////////////////////////////
-
-SfRequestContext::SfRequestContext() {
-    mUserAgent = MakeUserAgent().c_str();
-
-    set_net_log(new SfNetLog());
-
-    set_host_resolver(
-        net::CreateSystemHostResolver(
-                net::HostResolver::kDefaultParallelism,
-                NULL /* resolver_proc */,
-                net_log()));
-
-    set_ssl_config_service(
-        net::SSLConfigService::CreateSystemSSLConfigService());
-
-    mProxyConfigService = new net::ProxyConfigServiceAndroid;
-
-    set_proxy_service(net::ProxyService::CreateWithoutProxyResolver(
-        mProxyConfigService, net_log()));
-
-    set_http_transaction_factory(new net::HttpCache(
-            host_resolver(),
-            new net::CertVerifier(),
-            dnsrr_resolver(),
-            dns_cert_checker(),
-            proxy_service(),
-            ssl_config_service(),
-            net::HttpAuthHandlerFactory::CreateDefault(host_resolver()),
-            network_delegate(),
-            net_log(),
-            NULL));  // backend_factory
-
-    set_cookie_store(new net::CookieMonster(NULL, NULL));
-}
-
-const std::string &SfRequestContext::GetUserAgent(const GURL &url) const {
-    return mUserAgent;
-}
-
-status_t SfRequestContext::updateProxyConfig(
-        const char *host, int32_t port, const char *exclusionList) {
-    Mutex::Autolock autoLock(mProxyConfigLock);
-
-    if (host == NULL || *host == '\0') {
-        MY_LOGV("updateProxyConfig NULL");
-
-        std::string proxy;
-        std::string exList;
-        mProxyConfigService->UpdateProxySettings(proxy, exList);
-    } else {
-#if !defined(LOG_NDEBUG) || LOG_NDEBUG == 0
-        LOG_PRI(ANDROID_LOG_VERBOSE, LOG_TAG,
-                "updateProxyConfig %s:%d, exclude '%s'",
-                host, port, exclusionList);
-#endif
-
-        std::string proxy = StringPrintf("%s:%d", host, port).c_str();
-        std::string exList = exclusionList;
-        mProxyConfigService->UpdateProxySettings(proxy, exList);
-    }
-
-    return OK;
-}
-
-////////////////////////////////////////////////////////////////////////////////
-
-SfNetworkLibrary::SfNetworkLibrary() {}
-
-SfNetworkLibrary::VerifyResult SfNetworkLibrary::VerifyX509CertChain(
-        const std::vector<std::string>& cert_chain,
-        const std::string& hostname,
-        const std::string& auth_type) {
-    return VERIFY_OK;
-}
-
-////////////////////////////////////////////////////////////////////////////////
-
-SfDelegate::SfDelegate()
-    : mOwner(NULL),
-      mURLRequest(NULL),
-      mReadBuffer(new net::IOBufferWithSize(8192)),
-      mNumBytesRead(0),
-      mNumBytesTotal(0),
-      mDataDestination(NULL),
-      mAtEOS(false) {
-    InitializeNetworkThreadIfNecessary();
-}
-
-SfDelegate::~SfDelegate() {
-    CHECK(mURLRequest == NULL);
-}
-
-// static
-status_t SfDelegate::UpdateProxyConfig(
-        const char *host, int32_t port, const char *exclusionList) {
-    InitializeNetworkThreadIfNecessary();
-
-    return gReqContext->updateProxyConfig(host, port, exclusionList);
-}
-
-void SfDelegate::setOwner(ChromiumHTTPDataSource *owner) {
-    mOwner = owner;
-}
-
-void SfDelegate::setUID(uid_t uid) {
-    gReqContext->setUID(uid);
-}
-
-bool SfDelegate::getUID(uid_t *uid) const {
-    return gReqContext->getUID(uid);
-}
-
-void SfDelegate::OnReceivedRedirect(
-            net::URLRequest *request, const GURL &new_url, bool *defer_redirect) {
-    MY_LOGV("OnReceivedRedirect");
-}
-
-void SfDelegate::OnAuthRequired(
-            net::URLRequest *request, net::AuthChallengeInfo *auth_info) {
-    MY_LOGV("OnAuthRequired");
-
-    inherited::OnAuthRequired(request, auth_info);
-}
-
-void SfDelegate::OnCertificateRequested(
-            net::URLRequest *request, net::SSLCertRequestInfo *cert_request_info) {
-    MY_LOGV("OnCertificateRequested");
-
-    inherited::OnCertificateRequested(request, cert_request_info);
-}
-
-void SfDelegate::OnSSLCertificateError(
-            net::URLRequest *request, int cert_error, net::X509Certificate *cert) {
-    fprintf(stderr, "OnSSLCertificateError cert_error=%d\n", cert_error);
-
-    inherited::OnSSLCertificateError(request, cert_error, cert);
-}
-
-void SfDelegate::OnGetCookies(net::URLRequest *request, bool blocked_by_policy) {
-    MY_LOGV("OnGetCookies");
-}
-
-void SfDelegate::OnSetCookie(
-        net::URLRequest *request,
-        const std::string &cookie_line,
-        const net::CookieOptions &options,
-        bool blocked_by_policy) {
-    MY_LOGV("OnSetCookie");
-}
-
-void SfDelegate::OnResponseStarted(net::URLRequest *request) {
-    if (request->status().status() != net::URLRequestStatus::SUCCESS) {
-        MY_LOGI(StringPrintf(
-                    "Request failed with status %d and os_error %d",
-                    request->status().status(),
-                    request->status().os_error()).c_str());
-
-        delete mURLRequest;
-        mURLRequest = NULL;
-
-        mOwner->onConnectionFailed(ERROR_IO);
-        return;
-    } else if (mRangeRequested && request->GetResponseCode() != 206) {
-        MY_LOGI(StringPrintf(
-                    "We requested a content range, but server didn't "
-                    "support that. (responded with %d)",
-                    request->GetResponseCode()).c_str());
-
-        delete mURLRequest;
-        mURLRequest = NULL;
-
-        mOwner->onConnectionFailed(-EPIPE);
-        return;
-    } else if ((request->GetResponseCode() / 100) != 2) {
-        MY_LOGI(StringPrintf(
-                    "Server responded with http status %d",
-                    request->GetResponseCode()).c_str());
-
-        delete mURLRequest;
-        mURLRequest = NULL;
-
-        mOwner->onConnectionFailed(ERROR_IO);
-        return;
-    }
-
-    MY_LOGV("OnResponseStarted");
-
-    std::string headers;
-    request->GetAllResponseHeaders(&headers);
-
-    MY_LOGV(StringPrintf("response headers: %s", headers.c_str()).c_str());
-
-    std::string contentType;
-    request->GetResponseHeaderByName("Content-Type", &contentType);
-
-    mOwner->onConnectionEstablished(
-            request->GetExpectedContentSize(), contentType.c_str());
-}
-
-void SfDelegate::OnReadCompleted(net::URLRequest *request, int bytes_read) {
-    if (bytes_read == -1) {
-        MY_LOGI(StringPrintf(
-                    "OnReadCompleted, read failed, status %d",
-                    request->status().status()).c_str());
-
-        mOwner->onReadCompleted(ERROR_IO);
-        return;
-    }
-
-    MY_LOGV(StringPrintf("OnReadCompleted, read %d bytes", bytes_read).c_str());
-
-    if (bytes_read < 0) {
-        MY_LOGI(StringPrintf(
-                    "Read failed w/ status %d\n",
-                    request->status().status()).c_str());
-
-        mOwner->onReadCompleted(ERROR_IO);
-        return;
-    } else if (bytes_read == 0) {
-        mAtEOS = true;
-        mOwner->onReadCompleted(mNumBytesRead);
-        return;
-    }
-
-    CHECK_GT(bytes_read, 0);
-    CHECK_LE(mNumBytesRead + bytes_read, mNumBytesTotal);
-
-    memcpy((uint8_t *)mDataDestination + mNumBytesRead,
-           mReadBuffer->data(),
-           bytes_read);
-
-    mNumBytesRead += bytes_read;
-
-    readMore(request);
-}
-
-void SfDelegate::readMore(net::URLRequest *request) {
-    while (mNumBytesRead < mNumBytesTotal) {
-        size_t copy = mNumBytesTotal - mNumBytesRead;
-        if (copy > mReadBuffer->size()) {
-            copy = mReadBuffer->size();
-        }
-
-        int n;
-        if (request->Read(mReadBuffer, copy, &n)) {
-            MY_LOGV(StringPrintf("Read %d bytes directly.", n).c_str());
-
-            CHECK_LE((size_t)n, copy);
-
-            memcpy((uint8_t *)mDataDestination + mNumBytesRead,
-                   mReadBuffer->data(),
-                   n);
-
-            mNumBytesRead += n;
-
-            if (n == 0) {
-                mAtEOS = true;
-                break;
-            }
-        } else {
-            MY_LOGV("readMore pending read");
-
-            if (request->status().status() != net::URLRequestStatus::IO_PENDING) {
-                MY_LOGI(StringPrintf(
-                            "Direct read failed w/ status %d\n",
-                            request->status().status()).c_str());
-
-                mOwner->onReadCompleted(ERROR_IO);
-                return;
-            }
-
-            return;
-        }
-    }
-
-    mOwner->onReadCompleted(mNumBytesRead);
-}
-
-void SfDelegate::initiateConnection(
-        const char *uri,
-        const KeyedVector<String8, String8> *headers,
-        off64_t offset) {
-    GURL url(uri);
-
-    MessageLoop *loop = gNetworkThread->message_loop();
-    loop->PostTask(
-            FROM_HERE,
-            NewRunnableFunction(
-                &SfDelegate::OnInitiateConnectionWrapper,
-                this,
-                url,
-                headers,
-                offset));
-
-}
-
-// static
-void SfDelegate::OnInitiateConnectionWrapper(
-        SfDelegate *me, GURL url,
-        const KeyedVector<String8, String8> *headers,
-        off64_t offset) {
-    me->onInitiateConnection(url, headers, offset);
-}
-
-void SfDelegate::onInitiateConnection(
-        const GURL &url,
-        const KeyedVector<String8, String8> *extra,
-        off64_t offset) {
-    CHECK(mURLRequest == NULL);
-
-    mURLRequest = new net::URLRequest(url, this);
-    mAtEOS = false;
-
-    mRangeRequested = false;
-
-    if (offset != 0 || extra != NULL) {
-        net::HttpRequestHeaders headers =
-            mURLRequest->extra_request_headers();
-
-        if (offset != 0) {
-            headers.AddHeaderFromString(
-                    StringPrintf("Range: bytes=%lld-", offset).c_str());
-
-            mRangeRequested = true;
-        }
-
-        if (extra != NULL) {
-            for (size_t i = 0; i < extra->size(); ++i) {
-                AString s;
-                s.append(extra->keyAt(i).string());
-                s.append(": ");
-                s.append(extra->valueAt(i).string());
-
-                headers.AddHeaderFromString(s.c_str());
-            }
-        }
-
-        mURLRequest->SetExtraRequestHeaders(headers);
-    }
-
-    mURLRequest->set_context(gReqContext);
-
-    mURLRequest->Start();
-}
-
-void SfDelegate::initiateDisconnect() {
-    MessageLoop *loop = gNetworkThread->message_loop();
-    loop->PostTask(
-            FROM_HERE,
-            NewRunnableFunction(
-                &SfDelegate::OnInitiateDisconnectWrapper, this));
-}
-
-// static
-void SfDelegate::OnInitiateDisconnectWrapper(SfDelegate *me) {
-    me->onInitiateDisconnect();
-}
-
-void SfDelegate::onInitiateDisconnect() {
-    if (mURLRequest == NULL) {
-        return;
-    }
-
-    mURLRequest->Cancel();
-
-    delete mURLRequest;
-    mURLRequest = NULL;
-
-    mOwner->onDisconnectComplete();
-}
-
-void SfDelegate::initiateRead(void *data, size_t size) {
-    MessageLoop *loop = gNetworkThread->message_loop();
-    loop->PostTask(
-            FROM_HERE,
-            NewRunnableFunction(
-                &SfDelegate::OnInitiateReadWrapper, this, data, size));
-}
-
-// static
-void SfDelegate::OnInitiateReadWrapper(
-        SfDelegate *me, void *data, size_t size) {
-    me->onInitiateRead(data, size);
-}
-
-void SfDelegate::onInitiateRead(void *data, size_t size) {
-    CHECK(mURLRequest != NULL);
-
-    mNumBytesRead = 0;
-    mNumBytesTotal = size;
-    mDataDestination = data;
-
-    if (mAtEOS) {
-        mOwner->onReadCompleted(0);
-        return;
-    }
-
-    readMore(mURLRequest);
-}
-
-}  // namespace android
-

media/libstagefright/chromium_http/support.h

diff --git a/media/libstagefright/chromium_http/support.h b/media/libstagefright/chromium_http/support.h
deleted file mode 100644
index 975a1d3..0000000
--- a/media/libstagefright/chromium_http/support.h
+++ /dev/null
@@ -1,178 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef SUPPORT_H_
-
-#define SUPPORT_H_
-
-#include <assert.h>
-
-#include "net/base/net_log.h"
-#include "net/url_request/url_request.h"
-#include "net/url_request/url_request_context.h"
-#include "net/base/android_network_library.h"
-#include "net/base/io_buffer.h"
-
-#include <utils/KeyedVector.h>
-#include <utils/Mutex.h>
-#include <utils/String8.h>
-
-namespace net {
-    struct ProxyConfigServiceAndroid;
-};
-
-namespace android {
-
-struct SfNetLog : public net::NetLog {
-    SfNetLog();
-
-    virtual void AddEntry(
-            EventType type,
-            const base::TimeTicks &time,
-            const Source &source,
-            EventPhase phase,
-            EventParameters *params);
-
-    virtual uint32 NextID();
-    virtual LogLevel GetLogLevel() const;
-
-private:
-    uint32 mNextID;
-
-    DISALLOW_EVIL_CONSTRUCTORS(SfNetLog);
-};
-
-struct SfRequestContext : public net::URLRequestContext {
-    SfRequestContext();
-
-    virtual const std::string &GetUserAgent(const GURL &url) const;
-
-    status_t updateProxyConfig(
-            const char *host, int32_t port, const char *exclusionList);
-
-private:
-    Mutex mProxyConfigLock;
-
-    std::string mUserAgent;
-    net::ProxyConfigServiceAndroid *mProxyConfigService;
-
-    DISALLOW_EVIL_CONSTRUCTORS(SfRequestContext);
-};
-
-// This is required for https support, we don't really verify certificates,
-// we accept anything...
-struct SfNetworkLibrary : public net::AndroidNetworkLibrary {
-    SfNetworkLibrary();
-
-    virtual VerifyResult VerifyX509CertChain(
-            const std::vector<std::string>& cert_chain,
-            const std::string& hostname,
-            const std::string& auth_type);
-
-private:
-    DISALLOW_EVIL_CONSTRUCTORS(SfNetworkLibrary);
-};
-
-struct ChromiumHTTPDataSource;
-
-struct SfDelegate : public net::URLRequest::Delegate {
-    SfDelegate();
-    virtual ~SfDelegate();
-
-    void initiateConnection(
-            const char *uri,
-            const KeyedVector<String8, String8> *headers,
-            off64_t offset);
-
-    void initiateDisconnect();
-    void initiateRead(void *data, size_t size);
-
-    void setOwner(ChromiumHTTPDataSource *mOwner);
-
-    // Gets the UID of the calling process
-    bool getUID(uid_t *uid) const;
-
-    void setUID(uid_t uid);
-
-    virtual void OnReceivedRedirect(
-            net::URLRequest *request, const GURL &new_url, bool *defer_redirect);
-
-    virtual void OnAuthRequired(
-            net::URLRequest *request, net::AuthChallengeInfo *auth_info);
-
-    virtual void OnCertificateRequested(
-            net::URLRequest *request, net::SSLCertRequestInfo *cert_request_info);
-
-    virtual void OnSSLCertificateError(
-            net::URLRequest *request, int cert_error, net::X509Certificate *cert);
-
-    virtual void OnGetCookies(net::URLRequest *request, bool blocked_by_policy);
-
-    virtual void OnSetCookie(
-            net::URLRequest *request,
-            const std::string &cookie_line,
-            const net::CookieOptions &options,
-            bool blocked_by_policy);
-
-    virtual void OnResponseStarted(net::URLRequest *request);
-
-    virtual void OnReadCompleted(net::URLRequest *request, int bytes_read);
-
-    static status_t UpdateProxyConfig(
-            const char *host, int32_t port, const char *exclusionList);
-
-private:
-    typedef Delegate inherited;
-
-    ChromiumHTTPDataSource *mOwner;
-
-    net::URLRequest *mURLRequest;
-    scoped_refptr<net::IOBufferWithSize> mReadBuffer;
-
-    size_t mNumBytesRead;
-    size_t mNumBytesTotal;
-    void *mDataDestination;
-
-    bool mRangeRequested;
-    bool mAtEOS;
-
-    void readMore(net::URLRequest *request);
-
-    static void OnInitiateConnectionWrapper(
-            SfDelegate *me,
-            GURL url,
-            const KeyedVector<String8, String8> *headers,
-            off64_t offset);
-
-    static void OnInitiateDisconnectWrapper(SfDelegate *me);
-
-    static void OnInitiateReadWrapper(
-            SfDelegate *me, void *data, size_t size);
-
-    void onInitiateConnection(
-            const GURL &url,
-            const KeyedVector<String8, String8> *headers,
-            off64_t offset);
-
-    void onInitiateDisconnect();
-    void onInitiateRead(void *data, size_t size);
-
-    DISALLOW_EVIL_CONSTRUCTORS(SfDelegate);
-};
-
-}  // namespace android
-
-#endif  // SUPPORT_H_

media/libstagefright/chromium_http_stub.cpp

diff --git a/media/libstagefright/chromium_http_stub.cpp b/media/libstagefright/chromium_http_stub.cpp
deleted file mode 100644
index ed8a878..0000000
--- a/media/libstagefright/chromium_http_stub.cpp
+++ /dev/null
@@ -1,102 +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 <dlfcn.h>
-
-#include <media/stagefright/DataSource.h>
-
-#include "include/chromium_http_stub.h"
-#include "include/HTTPBase.h"
-
-namespace android {
-
-static bool gFirst = true;
-static void *gHandle;
-static Mutex gLibMutex;
-
-HTTPBase *(*gLib_createChromiumHTTPDataSource)(uint32_t flags);
-DataSource *(*gLib_createDataUriSource)(const char *uri);
-
-status_t (*gLib_UpdateChromiumHTTPDataSourceProxyConfig)(
-        const char *host, int32_t port, const char *exclusionList);
-
-static bool load_libstagefright_chromium_http() {
-    Mutex::Autolock autoLock(gLibMutex);
-    void *sym;
-
-    if (!gFirst) {
-        return (gHandle != NULL);
-    }
-
-    gFirst = false;
-
-    gHandle = dlopen("libstagefright_chromium_http.so", RTLD_NOW);
-    if (gHandle == NULL) {
-        return false;
-    }
-
-    sym = dlsym(gHandle, "createChromiumHTTPDataSource");
-    if (sym == NULL) {
-        gHandle = NULL;
-        return false;
-    }
-    gLib_createChromiumHTTPDataSource = (HTTPBase *(*)(uint32_t))sym;
-
-    sym = dlsym(gHandle, "createDataUriSource");
-    if (sym == NULL) {
-        gHandle = NULL;
-        return false;
-    }
-    gLib_createDataUriSource = (DataSource *(*)(const char *))sym;
-
-    sym = dlsym(gHandle, "UpdateChromiumHTTPDataSourceProxyConfig");
-    if (sym == NULL) {
-        gHandle = NULL;
-        return false;
-    }
-    gLib_UpdateChromiumHTTPDataSourceProxyConfig =
-        (status_t (*)(const char *, int32_t, const char *))sym;
-
-    return true;
-}
-
-HTTPBase *createChromiumHTTPDataSource(uint32_t flags) {
-    if (!load_libstagefright_chromium_http()) {
-        return NULL;
-    }
-
-    return gLib_createChromiumHTTPDataSource(flags);
-}
-
-status_t UpdateChromiumHTTPDataSourceProxyConfig(
-        const char *host, int32_t port, const char *exclusionList) {
-    if (!load_libstagefright_chromium_http()) {
-        return INVALID_OPERATION;
-    }
-
-    return gLib_UpdateChromiumHTTPDataSourceProxyConfig(
-            host, port, exclusionList);
-}
-
-DataSource *createDataUriSource(const char *uri) {
-    if (!load_libstagefright_chromium_http()) {
-        return NULL;
-    }
-
-    return gLib_createDataUriSource(uri);
-}
-
-}

media/libstagefright/codecs/aacdec/Android.mk

diff --git a/media/libstagefright/codecs/aacdec/Android.mk b/media/libstagefright/codecs/aacdec/Android.mk
index ffa64f9..49ff238 100644
--- a/media/libstagefright/codecs/aacdec/Android.mk
+++ b/media/libstagefright/codecs/aacdec/Android.mk
@@ -17,6 +17,8 @@
 
 LOCAL_CFLAGS :=
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_STATIC_LIBRARIES := libFraunhoferAAC
 
 LOCAL_SHARED_LIBRARIES := \

media/libstagefright/codecs/aacdec/SoftAAC2.cpp

diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
index f842e27..4ac8999 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
@@ -30,7 +30,7 @@
 #define DRC_DEFAULT_MOBILE_REF_LEVEL 64  /* 64*-0.25dB = -16 dB below full scale for mobile conf */
 #define DRC_DEFAULT_MOBILE_DRC_CUT   127 /* maximum compression of dynamic range for mobile conf */
 #define DRC_DEFAULT_MOBILE_DRC_BOOST 127 /* maximum compression of dynamic range for mobile conf */
-#define MAX_CHANNEL_COUNT            6  /* maximum number of audio channels that can be decoded */
+#define MAX_CHANNEL_COUNT            8  /* maximum number of audio channels that can be decoded */
 // names of properties that can be used to override the default DRC settings
 #define PROP_DRC_OVERRIDE_REF_LEVEL  "aac_drc_reference_level"
 #define PROP_DRC_OVERRIDE_CUT        "aac_drc_cut"
@@ -296,8 +296,11 @@
         if (!(property_get("media.aac_51_output_enabled", value, NULL) &&
                 (!strcmp(value, "1") || !strcasecmp(value, "true")))) {
             ALOGI("Downmixing multichannel AAC to stereo");
-            aacDecoder_SetParam(mAACDecoder, AAC_PCM_OUTPUT_CHANNELS, 2);
+            aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, 2);
             mStreamInfo->numChannels = 2;
+            // By default, the decoder creates a 5.1 channel downmix signal
+            // for seven and eight channel input streams. To enable 6.1 and 7.1 channel output
+            // use aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, -1)
         }
     }
 }
@@ -374,7 +377,7 @@
                 mNumSamplesOutput = 0;
             }
 
-            if (mIsADTS) {
+            if (mIsADTS && inHeader->nFilledLen) {
                 size_t adtsHeaderSize = 0;
                 // skip 30 bits, aac_frame_length follows.
                 // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll?????

media/libstagefright/codecs/aacenc/Android.mk

diff --git a/media/libstagefright/codecs/aacenc/Android.mk b/media/libstagefright/codecs/aacenc/Android.mk
index 057c69b..58ec3ba 100644
--- a/media/libstagefright/codecs/aacenc/Android.mk
+++ b/media/libstagefright/codecs/aacenc/Android.mk
@@ -82,6 +82,8 @@
 LOCAL_C_INCLUDES += $(LOCAL_PATH)/src/asm/ARMV7
 endif
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_STATIC_LIBRARY)
 
 ################################################################################
@@ -106,6 +108,8 @@
 
   LOCAL_CFLAGS :=
 
+  LOCAL_CFLAGS += -Werror
+
   LOCAL_STATIC_LIBRARIES := libFraunhoferAAC
 
   LOCAL_SHARED_LIBRARIES := \
@@ -128,6 +132,8 @@
 
   LOCAL_CFLAGS := -DOSCL_IMPORT_REF=
 
+  LOCAL_CFLAGS += -Werror
+
   LOCAL_STATIC_LIBRARIES := \
           libstagefright_aacenc
 

media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp

diff --git a/media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp b/media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp
index ff2b503..9a91579 100644
--- a/media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp
+++ b/media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp
@@ -338,7 +338,7 @@
     return OK;
 }
 
-void SoftAACEncoder2::onQueueFilled(OMX_U32 portIndex) {
+void SoftAACEncoder2::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError) {
         return;
     }

media/libstagefright/codecs/aacenc/basic_op/oper_32b.c

diff --git a/media/libstagefright/codecs/aacenc/basic_op/oper_32b.c b/media/libstagefright/codecs/aacenc/basic_op/oper_32b.c
index cc01927..1d029fc 100644
--- a/media/libstagefright/codecs/aacenc/basic_op/oper_32b.c
+++ b/media/libstagefright/codecs/aacenc/basic_op/oper_32b.c
@@ -24,6 +24,8 @@
 #include "basic_op.h"
 #include "oper_32b.h"
 
+#define UNUSED(x) (void)(x)
+
 /*****************************************************************************
  *                                                                           *
  *  Function L_Extract()                                                     *
@@ -243,6 +245,8 @@
 Word32 rsqrt(Word32 value,     /*!< Operand to square root (0.0 ... 1) */
              Word32 accuracy)  /*!< Number of valid bits that will be calculated */
 {
+    UNUSED(accuracy);
+
     Word32 root = 0;
 	Word32 scale;
 

media/libstagefright/codecs/aacenc/src/aacenc.c

diff --git a/media/libstagefright/codecs/aacenc/src/aacenc.c b/media/libstagefright/codecs/aacenc/src/aacenc.c
index d1c8621..40db92c 100644
--- a/media/libstagefright/codecs/aacenc/src/aacenc.c
+++ b/media/libstagefright/codecs/aacenc/src/aacenc.c
@@ -27,6 +27,8 @@
 #include "cmnMemory.h"
 #include "memalign.h"
 
+#define UNUSED(x) (void)(x)
+
 /**
 * Init the audio codec module and return codec handle
 * \param phCodec [OUT] Return the video codec handle
@@ -46,6 +48,8 @@
 	VO_MEM_OPERATOR *pMemOP;
 	int interMem;
 
+        UNUSED(vType);
+
 	interMem = 0;
 	error = 0;
 
@@ -471,6 +475,10 @@
 */
 VO_U32 VO_API voAACEncGetParam(VO_HANDLE hCodec, VO_S32 uParamID, VO_PTR pData)
 {
+        UNUSED(hCodec);
+        UNUSED(uParamID);
+        UNUSED(pData);
+
 	return VO_ERR_NONE;
 }
 

media/libstagefright/codecs/aacenc/src/adj_thr.c

diff --git a/media/libstagefright/codecs/aacenc/src/adj_thr.c b/media/libstagefright/codecs/aacenc/src/adj_thr.c
index ccfe883..471631c 100644
--- a/media/libstagefright/codecs/aacenc/src/adj_thr.c
+++ b/media/libstagefright/codecs/aacenc/src/adj_thr.c
@@ -72,7 +72,7 @@
                           const Word16 nChannels)
 {
   Word16 ch, sfb, sfbGrp;
-  Word32 *pthrExp, *psfbThre;
+  Word32 *pthrExp = NULL, *psfbThre;
   for (ch=0; ch<nChannels; ch++) {
     PSY_OUT_CHANNEL *psyOutChan = &psyOutChannel[ch];
 	 for(sfbGrp = 0; sfbGrp < psyOutChan->sfbCnt; sfbGrp+= psyOutChan->sfbPerGroup)

media/libstagefright/codecs/aacenc/src/bitenc.c

diff --git a/media/libstagefright/codecs/aacenc/src/bitenc.c b/media/libstagefright/codecs/aacenc/src/bitenc.c
index fcc12dd..d1fd647 100644
--- a/media/libstagefright/codecs/aacenc/src/bitenc.c
+++ b/media/libstagefright/codecs/aacenc/src/bitenc.c
@@ -26,6 +26,7 @@
 #include "qc_data.h"
 #include "interface.h"
 
+#define UNUSED(x) (void)(x)
 
 static const  Word16 globalGainOffset = 100;
 static const  Word16 icsReservedBit   = 0;
@@ -585,6 +586,8 @@
   Word16 elementUsedBits;
   Word16 frameBits=0;
 
+  UNUSED(ancBytes);
+
   /*   struct bitbuffer bsWriteCopy; */
   bitMarkUp = GetBitsAvail(hBitStream);
   if(qcOut->qcElement.adtsUsed)  /*  write adts header*/

media/libstagefright/codecs/aacenc/src/dyn_bits.c

diff --git a/media/libstagefright/codecs/aacenc/src/dyn_bits.c b/media/libstagefright/codecs/aacenc/src/dyn_bits.c
index 7769188..4d763d0 100644
--- a/media/libstagefright/codecs/aacenc/src/dyn_bits.c
+++ b/media/libstagefright/codecs/aacenc/src/dyn_bits.c
@@ -25,7 +25,6 @@
 #include "bit_cnt.h"
 #include "psy_const.h"
 
-
 /*****************************************************************************
 *
 * function name: buildBitLookUp
@@ -226,7 +225,7 @@
   }
 
   while (TRUE) {
-    Word16 maxMergeGain, maxNdx, maxNdxNext, maxNdxLast;
+    Word16 maxMergeGain, maxNdx = 0, maxNdxNext, maxNdxLast;
 
     maxMergeGain = findMaxMerge(mergeGainLookUp, sectionInfo, maxSfb, &maxNdx);
 

media/libstagefright/codecs/aacenc/src/psy_main.c

diff --git a/media/libstagefright/codecs/aacenc/src/psy_main.c b/media/libstagefright/codecs/aacenc/src/psy_main.c
index 4e9218c..6f0679c 100644
--- a/media/libstagefright/codecs/aacenc/src/psy_main.c
+++ b/media/libstagefright/codecs/aacenc/src/psy_main.c
@@ -38,6 +38,8 @@
 #include "tns_func.h"
 #include "memalign.h"
 
+#define UNUSED(x) (void)(x)
+
 /*                                    long       start       short       stop */
 static Word16 blockType2windowShape[] = {KBD_WINDOW,SINE_WINDOW,SINE_WINDOW,KBD_WINDOW};
 
@@ -170,7 +172,9 @@
 *****************************************************************************/
 Word16 PsyOutDelete(PSY_OUT *hPsyOut, VO_MEM_OPERATOR *pMemOP)
 {
-  hPsyOut=NULL;
+  UNUSED(hPsyOut);
+  UNUSED(pMemOP);
+
   return 0;
 }
 

media/libstagefright/codecs/aacenc/src/qc_main.c

diff --git a/media/libstagefright/codecs/aacenc/src/qc_main.c b/media/libstagefright/codecs/aacenc/src/qc_main.c
index 48ff300..e5d78aa 100644
--- a/media/libstagefright/codecs/aacenc/src/qc_main.c
+++ b/media/libstagefright/codecs/aacenc/src/qc_main.c
@@ -33,6 +33,7 @@
 #include "channel_map.h"
 #include "memalign.h"
 
+#define UNUSED(x) (void)(x)
 
 typedef enum{
   FRAME_LEN_BYTES_MODULO =  1,
@@ -204,11 +205,8 @@
 **********************************************************************************/
 void QCDelete(QC_STATE *hQC, VO_MEM_OPERATOR *pMemOP)
 {
-
-  /*
-     nothing to do
-  */
-  hQC=NULL;
+  UNUSED(hQC);
+  UNUSED(pMemOP);
 }
 
 /*********************************************************************************

media/libstagefright/codecs/aacenc/src/tns.c

diff --git a/media/libstagefright/codecs/aacenc/src/tns.c b/media/libstagefright/codecs/aacenc/src/tns.c
index 455a864..5172612 100644
--- a/media/libstagefright/codecs/aacenc/src/tns.c
+++ b/media/libstagefright/codecs/aacenc/src/tns.c
@@ -30,6 +30,8 @@
 #include "psy_configuration.h"
 #include "tns_func.h"
 
+#define UNUSED(x) (void)(x)
+
 #define TNS_MODIFY_BEGIN         2600  /* Hz */
 #define RATIO_PATCH_LOWER_BORDER 380   /* Hz */
 #define TNS_GAIN_THRESH			 141   /* 1.41*100 */
@@ -643,6 +645,8 @@
   Word32 i;
   Word32 tnsOrderPlus1 = tnsOrder + 1;
 
+  UNUSED(window);
+
   assert(tnsOrder <= TNS_MAX_ORDER);      /* remove asserts later? (btg) */
 
   for(i=0;i<tnsOrder;i++) {

media/libstagefright/codecs/amrnb/common/Android.mk

diff --git a/media/libstagefright/codecs/amrnb/common/Android.mk b/media/libstagefright/codecs/amrnb/common/Android.mk
index 30ce29c..a2b3c8f 100644
--- a/media/libstagefright/codecs/amrnb/common/Android.mk
+++ b/media/libstagefright/codecs/amrnb/common/Android.mk
@@ -69,6 +69,8 @@
 LOCAL_CFLAGS := \
         -DOSCL_UNUSED_ARG= -DOSCL_IMPORT_REF= -DOSCL_EXPORT_REF=
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_MODULE := libstagefright_amrnb_common
 
 include $(BUILD_SHARED_LIBRARY)

media/libstagefright/codecs/amrnb/dec/Android.mk

diff --git a/media/libstagefright/codecs/amrnb/dec/Android.mk b/media/libstagefright/codecs/amrnb/dec/Android.mk
index 8d6c6f8..b067456 100644
--- a/media/libstagefright/codecs/amrnb/dec/Android.mk
+++ b/media/libstagefright/codecs/amrnb/dec/Android.mk
@@ -47,6 +47,8 @@
 LOCAL_CFLAGS := \
         -DOSCL_UNUSED_ARG= -DOSCL_IMPORT_REF=
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_MODULE := libstagefright_amrnbdec
 
 include $(BUILD_STATIC_LIBRARY)
@@ -68,6 +70,8 @@
 
 LOCAL_CFLAGS := -DOSCL_IMPORT_REF=
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_STATIC_LIBRARIES := \
         libstagefright_amrnbdec libstagefright_amrwbdec
 

media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp

diff --git a/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp b/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp
index 3320688..d1b0f76 100644
--- a/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp
+++ b/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp
@@ -274,7 +274,7 @@
     return frameSize;
 }
 
-void SoftAMR::onQueueFilled(OMX_U32 portIndex) {
+void SoftAMR::onQueueFilled(OMX_U32 /* portIndex */) {
     List<BufferInfo *> &inQueue = getPortQueue(0);
     List<BufferInfo *> &outQueue = getPortQueue(1);
 
@@ -428,7 +428,7 @@
     }
 }
 
-void SoftAMR::onPortFlushCompleted(OMX_U32 portIndex) {
+void SoftAMR::onPortFlushCompleted(OMX_U32 /* portIndex */) {
 }
 
 void SoftAMR::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) {

media/libstagefright/codecs/amrnb/enc/Android.mk

diff --git a/media/libstagefright/codecs/amrnb/enc/Android.mk b/media/libstagefright/codecs/amrnb/enc/Android.mk
index f4e467a..afc0b89 100644
--- a/media/libstagefright/codecs/amrnb/enc/Android.mk
+++ b/media/libstagefright/codecs/amrnb/enc/Android.mk
@@ -69,6 +69,8 @@
 LOCAL_CFLAGS := \
         -DOSCL_UNUSED_ARG=
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_MODULE := libstagefright_amrnbenc
 
 include $(BUILD_STATIC_LIBRARY)
@@ -88,6 +90,8 @@
         $(LOCAL_PATH)/../common/include \
         $(LOCAL_PATH)/../common
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_STATIC_LIBRARIES := \
         libstagefright_amrnbenc
 

media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.cpp

diff --git a/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.cpp b/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.cpp
index 50b739c..9489457 100644
--- a/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.cpp
+++ b/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.cpp
@@ -270,7 +270,7 @@
     }
 }
 
-void SoftAMRNBEncoder::onQueueFilled(OMX_U32 portIndex) {
+void SoftAMRNBEncoder::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError) {
         return;
     }

media/libstagefright/codecs/amrwb/Android.mk

diff --git a/media/libstagefright/codecs/amrwb/Android.mk b/media/libstagefright/codecs/amrwb/Android.mk
index 677107f..efdf988 100644
--- a/media/libstagefright/codecs/amrwb/Android.mk
+++ b/media/libstagefright/codecs/amrwb/Android.mk
@@ -50,6 +50,8 @@
 LOCAL_CFLAGS := \
         -DOSCL_UNUSED_ARG= -DOSCL_IMPORT_REF=
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_MODULE := libstagefright_amrwbdec
 
 include $(BUILD_STATIC_LIBRARY)

media/libstagefright/codecs/amrwbenc/Android.mk

diff --git a/media/libstagefright/codecs/amrwbenc/Android.mk b/media/libstagefright/codecs/amrwbenc/Android.mk
index c5b8e0c..64fe8d1 100644
--- a/media/libstagefright/codecs/amrwbenc/Android.mk
+++ b/media/libstagefright/codecs/amrwbenc/Android.mk
@@ -112,6 +112,8 @@
 LOCAL_C_INCLUDES += $(LOCAL_PATH)/src/asm/ARMV7
 endif
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_STATIC_LIBRARY)
 
 ################################################################################
@@ -126,6 +128,8 @@
 	frameworks/av/media/libstagefright/codecs/common/include \
 	frameworks/native/include/media/openmax
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_STATIC_LIBRARIES := \
         libstagefright_amrwbenc
 

media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.cpp

diff --git a/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.cpp b/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.cpp
index 9ccb49c..91a512d 100644
--- a/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.cpp
+++ b/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.cpp
@@ -317,7 +317,7 @@
     }
 }
 
-void SoftAMRWBEncoder::onQueueFilled(OMX_U32 portIndex) {
+void SoftAMRWBEncoder::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError) {
         return;
     }

media/libstagefright/codecs/amrwbenc/src/autocorr.c

diff --git a/media/libstagefright/codecs/amrwbenc/src/autocorr.c b/media/libstagefright/codecs/amrwbenc/src/autocorr.c
index 8c477ca..0b2ea89 100644
--- a/media/libstagefright/codecs/amrwbenc/src/autocorr.c
+++ b/media/libstagefright/codecs/amrwbenc/src/autocorr.c
@@ -28,6 +28,8 @@
 #include "acelp.h"
 #include "ham_wind.tab"
 
+#define UNUSED(x) (void)(x)
+
 void Autocorr(
 		Word16 x[],                           /* (i)    : Input signal                      */
 		Word16 m,                             /* (i)    : LPC order                         */
@@ -40,6 +42,8 @@
 	Word32 L_sum, L_sum1, L_tmp, F_LEN;
 	Word16 *p1,*p2,*p3;
 	const Word16 *p4;
+        UNUSED(m);
+
 	/* Windowing of signal */
 	p1 = x;
 	p4 = vo_window;

media/libstagefright/codecs/amrwbenc/src/convolve.c

diff --git a/media/libstagefright/codecs/amrwbenc/src/convolve.c b/media/libstagefright/codecs/amrwbenc/src/convolve.c
index acba532..4c1f7d4 100644
--- a/media/libstagefright/codecs/amrwbenc/src/convolve.c
+++ b/media/libstagefright/codecs/amrwbenc/src/convolve.c
@@ -25,6 +25,8 @@
 #include "typedef.h"
 #include "basic_op.h"
 
+#define UNUSED(x) (void)(x)
+
 void Convolve (
 		Word16 x[],        /* (i)     : input vector                           */
 		Word16 h[],        /* (i)     : impulse response                       */
@@ -35,6 +37,8 @@
 	Word32  i, n;
 	Word16 *tmpH,*tmpX;
 	Word32 s;
+        UNUSED(L);
+
 	for (n = 0; n < 64;)
 	{
 		tmpH = h+n;

media/libstagefright/codecs/amrwbenc/src/pitch_f4.c

diff --git a/media/libstagefright/codecs/amrwbenc/src/pitch_f4.c b/media/libstagefright/codecs/amrwbenc/src/pitch_f4.c
index 0d66c31..b66b55e 100644
--- a/media/libstagefright/codecs/amrwbenc/src/pitch_f4.c
+++ b/media/libstagefright/codecs/amrwbenc/src/pitch_f4.c
@@ -31,6 +31,8 @@
 #define UP_SAMP      4
 #define L_INTERPOL1  4
 
+#define UNUSED(x) (void)(x)
+
 /* Local functions */
 
 #ifdef ASM_OPT
@@ -171,6 +173,7 @@
 	Word32 corr, exp_corr, norm, exp, scale;
 	Word16 exp_norm, excf[L_SUBFR], tmp;
 	Word32 L_tmp, L_tmp1, L_tmp2;
+        UNUSED(L_subfr);
 
 	/* compute the filtered excitation for the first delay t_min */
 	k = -t_min;

media/libstagefright/codecs/amrwbenc/src/syn_filt.c

diff --git a/media/libstagefright/codecs/amrwbenc/src/syn_filt.c b/media/libstagefright/codecs/amrwbenc/src/syn_filt.c
index 1bda05a..961aadc 100644
--- a/media/libstagefright/codecs/amrwbenc/src/syn_filt.c
+++ b/media/libstagefright/codecs/amrwbenc/src/syn_filt.c
@@ -26,6 +26,8 @@
 #include "math_op.h"
 #include "cnst.h"
 
+#define UNUSED(x) (void)(x)
+
 void Syn_filt(
 		Word16 a[],                           /* (i) Q12 : a[m+1] prediction coefficients           */
 		Word16 x[],                           /* (i)     : input signal                             */
@@ -95,6 +97,8 @@
 	Word32 i,a0;
 	Word32 L_tmp, L_tmp1;
 	Word16 *p1, *p2, *p3;
+        UNUSED(m);
+
 	a0 = a[0] >> (4 + Qnew);          /* input / 16 and >>Qnew */
 	/* Do the filtering. */
 	for (i = 0; i < lg; i++)

media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c

diff --git a/media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c b/media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c
index ea9da52..df7b9b3 100644
--- a/media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c
+++ b/media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c
@@ -39,6 +39,8 @@
 #include "mem_align.h"
 #include "cmnMemory.h"
 
+#define UNUSED(x) (void)(x)
+
 #ifdef __cplusplus
 extern "C" {
 #endif
@@ -1602,6 +1604,8 @@
 	VO_MEM_OPERATOR voMemoprator;
 #endif
 	VO_MEM_OPERATOR *pMemOP;
+        UNUSED(vType);
+
 	int interMem = 0;
 
 	if(pUserData == NULL || pUserData->memflag != VO_IMF_USERMEMOPERATOR || pUserData->memData == NULL )

media/libstagefright/codecs/avc/common/Android.mk

diff --git a/media/libstagefright/codecs/avc/common/Android.mk b/media/libstagefright/codecs/avc/common/Android.mk
index 22dee15..844ef0a 100644
--- a/media/libstagefright/codecs/avc/common/Android.mk
+++ b/media/libstagefright/codecs/avc/common/Android.mk
@@ -16,4 +16,6 @@
 	$(LOCAL_PATH)/src \
  	$(LOCAL_PATH)/include
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_SHARED_LIBRARY)

media/libstagefright/codecs/avc/enc/Android.mk

diff --git a/media/libstagefright/codecs/avc/enc/Android.mk b/media/libstagefright/codecs/avc/enc/Android.mk
index 7d17c2a..537ba42 100644
--- a/media/libstagefright/codecs/avc/enc/Android.mk
+++ b/media/libstagefright/codecs/avc/enc/Android.mk
@@ -30,6 +30,8 @@
 LOCAL_CFLAGS := \
     -DOSCL_IMPORT_REF= -DOSCL_UNUSED_ARG= -DOSCL_EXPORT_REF=
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_STATIC_LIBRARY)
 
 ################################################################################
@@ -69,4 +71,6 @@
 LOCAL_MODULE := libstagefright_soft_h264enc
 LOCAL_MODULE_TAGS := optional
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_SHARED_LIBRARY)

media/libstagefright/codecs/avc/enc/SoftAVCEncoder.cpp

diff --git a/media/libstagefright/codecs/avc/enc/SoftAVCEncoder.cpp b/media/libstagefright/codecs/avc/enc/SoftAVCEncoder.cpp
index 1d66120..a15b040 100644
--- a/media/libstagefright/codecs/avc/enc/SoftAVCEncoder.cpp
+++ b/media/libstagefright/codecs/avc/enc/SoftAVCEncoder.cpp
@@ -34,6 +34,12 @@
 
 #include "SoftAVCEncoder.h"
 
+#if LOG_NDEBUG
+#define UNUSED_UNLESS_VERBOSE(x) (void)(x)
+#else
+#define UNUSED_UNLESS_VERBOSE(x)
+#endif
+
 namespace android {
 
 template<class T>
@@ -136,14 +142,14 @@
 }
 
 static void* MallocWrapper(
-        void *userData, int32_t size, int32_t attrs) {
+        void * /* userData */, int32_t size, int32_t /* attrs */) {
     void *ptr = malloc(size);
     if (ptr)
         memset(ptr, 0, size);
     return ptr;
 }
 
-static void FreeWrapper(void *userData, void* ptr) {
+static void FreeWrapper(void * /* userData */, void* ptr) {
     free(ptr);
 }
 
@@ -217,7 +223,7 @@
     mHandle->CBAVC_Free = FreeWrapper;
 
     CHECK(mEncParams != NULL);
-    memset(mEncParams, 0, sizeof(mEncParams));
+    memset(mEncParams, 0, sizeof(*mEncParams));
     mEncParams->rate_control = AVC_ON;
     mEncParams->initQP = 0;
     mEncParams->init_CBP_removal_delay = 1600;
@@ -593,6 +599,17 @@
                 mVideoHeight = def->format.video.nFrameHeight;
                 mVideoFrameRate = def->format.video.xFramerate >> 16;
                 mVideoColorFormat = def->format.video.eColorFormat;
+
+                OMX_PARAM_PORTDEFINITIONTYPE *portDef =
+                    &editPortInfo(0)->mDef;
+                portDef->format.video.nFrameWidth = mVideoWidth;
+                portDef->format.video.nFrameHeight = mVideoHeight;
+                portDef->format.video.xFramerate = def->format.video.xFramerate;
+                portDef->format.video.eColorFormat =
+                    (OMX_COLOR_FORMATTYPE) mVideoColorFormat;
+                portDef = &editPortInfo(1)->mDef;
+                portDef->format.video.nFrameWidth = mVideoWidth;
+                portDef->format.video.nFrameHeight = mVideoHeight;
             } else {
                 mVideoBitRate = def->format.video.nBitrate;
             }
@@ -711,7 +728,7 @@
     }
 }
 
-void SoftAVCEncoder::onQueueFilled(OMX_U32 portIndex) {
+void SoftAVCEncoder::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError || mSawInputEOS) {
         return;
     }
@@ -784,7 +801,7 @@
             }
         }
 
-        buffer_handle_t srcBuffer; // for MetaDataMode only
+        buffer_handle_t srcBuffer = NULL; // for MetaDataMode only
 
         // Get next input video frame
         if (mReadyForNextFrame) {
@@ -871,7 +888,13 @@
         CHECK(encoderStatus == AVCENC_SUCCESS || encoderStatus == AVCENC_NEW_IDR);
         dataLength = outHeader->nAllocLen;  // Reset the output buffer length
         if (inHeader->nFilledLen > 0) {
+            if (outHeader->nAllocLen >= 4) {
+                memcpy(outPtr, "\x00\x00\x00\x01", 4);
+                outPtr += 4;
+                dataLength -= 4;
+            }
             encoderStatus = PVAVCEncodeNAL(mHandle, outPtr, &dataLength, &type);
+            dataLength = outPtr + dataLength - outHeader->pBuffer;
             if (encoderStatus == AVCENC_SUCCESS) {
                 CHECK(NULL == PVAVCEncGetOverrunBuffer(mHandle));
             } else if (encoderStatus == AVCENC_PICTURE_READY) {
@@ -947,6 +970,7 @@
 }
 
 void SoftAVCEncoder::signalBufferReturned(MediaBuffer *buffer) {
+    UNUSED_UNLESS_VERBOSE(buffer);
     ALOGV("signalBufferReturned: %p", buffer);
 }
 

media/libstagefright/codecs/avc/enc/src/bitstream_io.cpp

diff --git a/media/libstagefright/codecs/avc/enc/src/bitstream_io.cpp b/media/libstagefright/codecs/avc/enc/src/bitstream_io.cpp
index 0e3037f..d71c327 100644
--- a/media/libstagefright/codecs/avc/enc/src/bitstream_io.cpp
+++ b/media/libstagefright/codecs/avc/enc/src/bitstream_io.cpp
@@ -103,6 +103,15 @@
     {
         num_bits -= 8;
         byte = (current_word >> num_bits) & 0xFF;
+        if (stream->count_zeros == 2)
+        {   /* for num_bits = 32, this can add 2 more bytes extra for EPBS */
+            if (byte <= 3)
+            {
+                *write_pnt++ = 0x3;
+                stream->write_pos++;
+                stream->count_zeros = 0;
+            }
+        }
         if (byte != 0)
         {
             *write_pnt++ = byte;
@@ -114,12 +123,6 @@
             stream->count_zeros++;
             *write_pnt++ = byte;
             stream->write_pos++;
-            if (stream->count_zeros == 2)
-            {   /* for num_bits = 32, this can add 2 more bytes extra for EPBS */
-                *write_pnt++ = 0x3;
-                stream->write_pos++;
-                stream->count_zeros = 0;
-            }
         }
     }
 

media/libstagefright/codecs/common/Android.mk

diff --git a/media/libstagefright/codecs/common/Android.mk b/media/libstagefright/codecs/common/Android.mk
index a33cb92..b0010ff 100644
--- a/media/libstagefright/codecs/common/Android.mk
+++ b/media/libstagefright/codecs/common/Android.mk
@@ -14,6 +14,8 @@
 LOCAL_C_INCLUDES := \
 	$(LOCAL_PATH)/include
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_SHARED_LIBRARY)
 
 

media/libstagefright/codecs/common/cmnMemory.c

diff --git a/media/libstagefright/codecs/common/cmnMemory.c b/media/libstagefright/codecs/common/cmnMemory.c
index aa52bd9..5bb6cc4 100644
--- a/media/libstagefright/codecs/common/cmnMemory.c
+++ b/media/libstagefright/codecs/common/cmnMemory.c
@@ -26,8 +26,12 @@
 
 //VO_MEM_OPERATOR		g_memOP;
 
+#define UNUSED(x) (void)(x)
+
 VO_U32 cmnMemAlloc (VO_S32 uID,  VO_MEM_INFO * pMemInfo)
 {
+        UNUSED(uID);
+
 	if (!pMemInfo)
 		return VO_ERR_INVALID_ARG;
 
@@ -37,34 +41,48 @@
 
 VO_U32 cmnMemFree (VO_S32 uID, VO_PTR pMem)
 {
+        UNUSED(uID);
+
 	free (pMem);
 	return 0;
 }
 
 VO_U32	cmnMemSet (VO_S32 uID, VO_PTR pBuff, VO_U8 uValue, VO_U32 uSize)
 {
+        UNUSED(uID);
+
 	memset (pBuff, uValue, uSize);
 	return 0;
 }
 
 VO_U32	cmnMemCopy (VO_S32 uID, VO_PTR pDest, VO_PTR pSource, VO_U32 uSize)
 {
+        UNUSED(uID);
+
 	memcpy (pDest, pSource, uSize);
 	return 0;
 }
 
 VO_U32	cmnMemCheck (VO_S32 uID, VO_PTR pBuffer, VO_U32 uSize)
 {
+        UNUSED(uID);
+        UNUSED(pBuffer);
+        UNUSED(uSize);
+
 	return 0;
 }
 
 VO_S32 cmnMemCompare (VO_S32 uID, VO_PTR pBuffer1, VO_PTR pBuffer2, VO_U32 uSize)
 {
+        UNUSED(uID);
+
 	return memcmp(pBuffer1, pBuffer2, uSize);
 }
 
 VO_U32	cmnMemMove (VO_S32 uID, VO_PTR pDest, VO_PTR pSource, VO_U32 uSize)
 {
+        UNUSED(uID);
+
 	memmove (pDest, pSource, uSize);
 	return 0;
 }

media/libstagefright/codecs/flac/enc/Android.mk

diff --git a/media/libstagefright/codecs/flac/enc/Android.mk b/media/libstagefright/codecs/flac/enc/Android.mk
index f01d605..59a11de 100644
--- a/media/libstagefright/codecs/flac/enc/Android.mk
+++ b/media/libstagefright/codecs/flac/enc/Android.mk
@@ -9,6 +9,8 @@
         frameworks/native/include/media/openmax \
         external/flac/include
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_SHARED_LIBRARIES := \
         libstagefright libstagefright_omx libstagefright_foundation libutils liblog
 

media/libstagefright/codecs/flac/enc/SoftFlacEncoder.cpp

diff --git a/media/libstagefright/codecs/flac/enc/SoftFlacEncoder.cpp b/media/libstagefright/codecs/flac/enc/SoftFlacEncoder.cpp
index e64fe72..40661e7 100644
--- a/media/libstagefright/codecs/flac/enc/SoftFlacEncoder.cpp
+++ b/media/libstagefright/codecs/flac/enc/SoftFlacEncoder.cpp
@@ -27,6 +27,12 @@
 #define FLAC_COMPRESSION_LEVEL_DEFAULT 5
 #define FLAC_COMPRESSION_LEVEL_MAX     8
 
+#if LOG_NDEBUG
+#define UNUSED_UNLESS_VERBOSE(x) (void)(x)
+#else
+#define UNUSED_UNLESS_VERBOSE(x)
+#endif
+
 namespace android {
 
 template<class T>
@@ -257,7 +263,7 @@
 }
 
 void SoftFlacEncoder::onQueueFilled(OMX_U32 portIndex) {
-
+    UNUSED_UNLESS_VERBOSE(portIndex);
     ALOGV("SoftFlacEncoder::onQueueFilled(portIndex=%ld)", portIndex);
 
     if (mSignalledError) {
@@ -343,10 +349,11 @@
     }
 }
 
-
 FLAC__StreamEncoderWriteStatus SoftFlacEncoder::onEncodedFlacAvailable(
             const FLAC__byte buffer[],
-            size_t bytes, unsigned samples, unsigned current_frame) {
+            size_t bytes, unsigned samples,
+            unsigned current_frame) {
+    UNUSED_UNLESS_VERBOSE(current_frame);
     ALOGV("SoftFlacEncoder::onEncodedFlacAvailable(bytes=%d, samples=%d, curr_frame=%d)",
             bytes, samples, current_frame);
 
@@ -444,8 +451,12 @@
 
 // static
 FLAC__StreamEncoderWriteStatus SoftFlacEncoder::flacEncoderWriteCallback(
-            const FLAC__StreamEncoder *encoder, const FLAC__byte buffer[],
-            size_t bytes, unsigned samples, unsigned current_frame, void *client_data) {
+            const FLAC__StreamEncoder * /* encoder */,
+            const FLAC__byte buffer[],
+            size_t bytes,
+            unsigned samples,
+            unsigned current_frame,
+            void *client_data) {
     return ((SoftFlacEncoder*) client_data)->onEncodedFlacAvailable(
             buffer, bytes, samples, current_frame);
 }

media/libstagefright/codecs/g711/dec/Android.mk

diff --git a/media/libstagefright/codecs/g711/dec/Android.mk b/media/libstagefright/codecs/g711/dec/Android.mk
index 4c80da6..a0112e1 100644
--- a/media/libstagefright/codecs/g711/dec/Android.mk
+++ b/media/libstagefright/codecs/g711/dec/Android.mk
@@ -14,4 +14,6 @@
 LOCAL_MODULE := libstagefright_soft_g711dec
 LOCAL_MODULE_TAGS := optional
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_SHARED_LIBRARY)

media/libstagefright/codecs/g711/dec/SoftG711.cpp

diff --git a/media/libstagefright/codecs/g711/dec/SoftG711.cpp b/media/libstagefright/codecs/g711/dec/SoftG711.cpp
index bcdd3c7..160ada0 100644
--- a/media/libstagefright/codecs/g711/dec/SoftG711.cpp
+++ b/media/libstagefright/codecs/g711/dec/SoftG711.cpp
@@ -182,7 +182,7 @@
     }
 }
 
-void SoftG711::onQueueFilled(OMX_U32 portIndex) {
+void SoftG711::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError) {
         return;
     }

media/libstagefright/codecs/gsm/dec/Android.mk

diff --git a/media/libstagefright/codecs/gsm/dec/Android.mk b/media/libstagefright/codecs/gsm/dec/Android.mk
index 71613d2..30868d5 100644
--- a/media/libstagefright/codecs/gsm/dec/Android.mk
+++ b/media/libstagefright/codecs/gsm/dec/Android.mk
@@ -9,6 +9,8 @@
         frameworks/native/include/media/openmax \
         external/libgsm/inc
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_SHARED_LIBRARIES := \
         libstagefright libstagefright_omx libstagefright_foundation libutils liblog
 

media/libstagefright/codecs/gsm/dec/SoftGSM.cpp

diff --git a/media/libstagefright/codecs/gsm/dec/SoftGSM.cpp b/media/libstagefright/codecs/gsm/dec/SoftGSM.cpp
index 00e0c85..18f7d29 100644
--- a/media/libstagefright/codecs/gsm/dec/SoftGSM.cpp
+++ b/media/libstagefright/codecs/gsm/dec/SoftGSM.cpp
@@ -172,7 +172,7 @@
     }
 }
 
-void SoftGSM::onQueueFilled(OMX_U32 portIndex) {
+void SoftGSM::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError) {
         return;
     }

media/libstagefright/codecs/m4v_h263/dec/Android.mk

diff --git a/media/libstagefright/codecs/m4v_h263/dec/Android.mk b/media/libstagefright/codecs/m4v_h263/dec/Android.mk
index a3d5779..1d232c6 100644
--- a/media/libstagefright/codecs/m4v_h263/dec/Android.mk
+++ b/media/libstagefright/codecs/m4v_h263/dec/Android.mk
@@ -46,6 +46,8 @@
 
 LOCAL_CFLAGS := -DOSCL_EXPORT_REF= -DOSCL_IMPORT_REF=
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_STATIC_LIBRARY)
 
 ################################################################################
@@ -72,4 +74,6 @@
 LOCAL_MODULE := libstagefright_soft_mpeg4dec
 LOCAL_MODULE_TAGS := optional
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_SHARED_LIBRARY)

media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp

diff --git a/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp b/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp
index fb2a430..0d1ab71 100644
--- a/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp
+++ b/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp
@@ -91,7 +91,7 @@
     return OK;
 }
 
-void SoftMPEG4::onQueueFilled(OMX_U32 portIndex) {
+void SoftMPEG4::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError || mOutputPortSettingsChange != NONE) {
         return;
     }

media/libstagefright/codecs/m4v_h263/dec/src/get_pred_adv_b_add.cpp

diff --git a/media/libstagefright/codecs/m4v_h263/dec/src/get_pred_adv_b_add.cpp b/media/libstagefright/codecs/m4v_h263/dec/src/get_pred_adv_b_add.cpp
index e23f23d..fe9e7dc 100644
--- a/media/libstagefright/codecs/m4v_h263/dec/src/get_pred_adv_b_add.cpp
+++ b/media/libstagefright/codecs/m4v_h263/dec/src/get_pred_adv_b_add.cpp
@@ -96,7 +96,7 @@
     offset = width - B_SIZE; /* offset for prev */
     offset2 = (pred_width_rnd >> 1) - 4; /* offset for pred_block */
 
-    tmp = (uint32)prev & 0x3;
+    tmp = (uintptr_t)prev & 0x3;
     pred_block -= offset2; /* preset */
 
     if (tmp == 0)  /* word-aligned */
@@ -203,7 +203,7 @@
     /* Branch based on pixel location (half-pel or full-pel) for x and y */
     pred_block -= offset2; /* preset */
 
-    tmp = (uint32)prev & 3;
+    tmp = (uintptr_t)prev & 3;
     mask = 254;
     mask |= (mask << 8);
     mask |= (mask << 16); /* 0xFEFEFEFE */
@@ -532,7 +532,7 @@
     /* Branch based on pixel location (half-pel or full-pel) for x and y */
     pred_block -= offset2; /* preset */
 
-    tmp = (uint32)prev & 3;
+    tmp = (uintptr_t)prev & 3;
     mask = 254;
     mask |= (mask << 8);
     mask |= (mask << 16); /* 0xFEFEFEFE */
@@ -884,7 +884,7 @@
     mask |= (mask << 8);
     mask |= (mask << 16); /* 0x3f3f3f3f */
 
-    tmp = (uint32)prev & 3;
+    tmp = (uintptr_t)prev & 3;
 
     pred_block -= 4; /* preset */
 

media/libstagefright/codecs/m4v_h263/enc/Android.mk

diff --git a/media/libstagefright/codecs/m4v_h263/enc/Android.mk b/media/libstagefright/codecs/m4v_h263/enc/Android.mk
index 83a2dd2..c9006d9 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/Android.mk
+++ b/media/libstagefright/codecs/m4v_h263/enc/Android.mk
@@ -33,6 +33,8 @@
     $(TOP)/frameworks/av/media/libstagefright/include \
     $(TOP)/frameworks/native/include/media/openmax
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_STATIC_LIBRARY)
 
 ################################################################################
@@ -72,4 +74,6 @@
 LOCAL_MODULE := libstagefright_soft_mpeg4enc
 LOCAL_MODULE_TAGS := optional
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_SHARED_LIBRARY)

media/libstagefright/codecs/m4v_h263/enc/SoftMPEG4Encoder.cpp

diff --git a/media/libstagefright/codecs/m4v_h263/enc/SoftMPEG4Encoder.cpp b/media/libstagefright/codecs/m4v_h263/enc/SoftMPEG4Encoder.cpp
index e02af90..2c73e57 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/SoftMPEG4Encoder.cpp
+++ b/media/libstagefright/codecs/m4v_h263/enc/SoftMPEG4Encoder.cpp
@@ -620,7 +620,7 @@
     }
 }
 
-void SoftMPEG4Encoder::onQueueFilled(OMX_U32 portIndex) {
+void SoftMPEG4Encoder::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError || mSawInputEOS) {
         return;
     }
@@ -677,7 +677,7 @@
             mSawInputEOS = true;
         }
 
-        buffer_handle_t srcBuffer; // for MetaDataMode only
+        buffer_handle_t srcBuffer = NULL; // for MetaDataMode only
         if (inHeader->nFilledLen > 0) {
             uint8_t *inputData = NULL;
             if (mStoreMetaDataInBuffers) {

media/libstagefright/codecs/m4v_h263/enc/src/dct.cpp

diff --git a/media/libstagefright/codecs/m4v_h263/enc/src/dct.cpp b/media/libstagefright/codecs/m4v_h263/enc/src/dct.cpp
index fa50eeb..fa4ae23 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/src/dct.cpp
+++ b/media/libstagefright/codecs/m4v_h263/enc/src/dct.cpp
@@ -250,7 +250,7 @@
             out[40] = k4 ;   /* row 5 */
             out++;
         }
-        while ((UInt)out < (UInt)dst) ;
+        while ((uintptr_t)out < (uintptr_t)dst) ;
 
         return ;
     }
@@ -455,7 +455,7 @@
             out[8] = k5 ;       /* row 1 */
             out++;
         }
-        while ((UInt)out < (UInt)dst) ;
+        while ((uintptr_t)out < (uintptr_t)dst) ;
 
         return ;
     }
@@ -635,7 +635,7 @@
             out[8] = k5 ;       /* row 1 */
             out++;
         }
-        while ((UInt)out < (UInt)dst) ;
+        while ((uintptr_t)out < (uintptr_t)dst) ;
 
         return ;
     }
@@ -846,7 +846,7 @@
             out[40] = k4 ;   /* row 5 */
             out++;
         }
-        while ((UInt)out < (UInt)dst) ;
+        while ((uintptr_t)out < (uintptr_t)dst) ;
 
         return ;
     }
@@ -1033,7 +1033,7 @@
             out[8] = k5 ;       /* row 1 */
             out++;
         }
-        while ((UInt)out < (UInt)dst) ;
+        while ((uintptr_t)out < (uintptr_t)dst) ;
 
         return ;
     }
@@ -1195,7 +1195,7 @@
             out[8] = k5 ;       /* row 1 */
             out++;
         }
-        while ((UInt)out < (UInt)dst) ;
+        while ((uintptr_t)out < (uintptr_t)dst) ;
 
         return ;
     }

media/libstagefright/codecs/m4v_h263/enc/src/fastcodemb.cpp

diff --git a/media/libstagefright/codecs/m4v_h263/enc/src/fastcodemb.cpp b/media/libstagefright/codecs/m4v_h263/enc/src/fastcodemb.cpp
index 6fd41c3..0ad39a6 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/src/fastcodemb.cpp
+++ b/media/libstagefright/codecs/m4v_h263/enc/src/fastcodemb.cpp
@@ -572,7 +572,7 @@
         cur2    = cur2 & (mask << 8);   /* mask first and third bytes */
         sum2    = sum2 + ((UInt)cur2 >> 8);
     }
-    while ((UInt)curInt < (UInt)end);
+    while ((uintptr_t)curInt < (uintptr_t)end);
 
     cur1 = sum4 - (sum2 << 8);  /* get even-sum */
     cur1 = cur1 + sum2;         /* add 16 bit even-sum and odd-sum*/
@@ -611,7 +611,7 @@
         load2 = load2 & (mask << 8); /* even bytes */
         sum2 += ((UInt)load2 >> 8); /* sum even bytes, 16 bit */
     }
-    while ((UInt)curInt < (UInt)end);
+    while ((uintptr_t)curInt < (uintptr_t)end);
     load1 = sum4 - (sum2 << 8);     /* get even-sum */
     load1 = load1 + sum2;           /* add 16 bit even-sum and odd-sum*/
     load1 = load1 + (load1 << 16);  /* add upper and lower 16 bit sum */

media/libstagefright/codecs/m4v_h263/enc/src/motion_comp.cpp

diff --git a/media/libstagefright/codecs/m4v_h263/enc/src/motion_comp.cpp b/media/libstagefright/codecs/m4v_h263/enc/src/motion_comp.cpp
index b81d278..06e8926 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/src/motion_comp.cpp
+++ b/media/libstagefright/codecs/m4v_h263/enc/src/motion_comp.cpp
@@ -1959,7 +1959,7 @@
         dst += offset;
         src += offset;
     }
-    while ((UInt)src < (UInt)end);
+    while ((uintptr_t)src < (uintptr_t)end);
 
     return ;
 }

media/libstagefright/codecs/m4v_h263/enc/src/sad_inline.h

diff --git a/media/libstagefright/codecs/m4v_h263/enc/src/sad_inline.h b/media/libstagefright/codecs/m4v_h263/enc/src/sad_inline.h
index ba77dfd..b865f23 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/src/sad_inline.h
+++ b/media/libstagefright/codecs/m4v_h263/enc/src/sad_inline.h
@@ -85,7 +85,7 @@
 
         x9 = 0x80808080; /* const. */
 
-        x8 = (uint32)ref & 0x3;
+        x8 = (uintptr_t)ref & 0x3;
         if (x8 == 3)
             goto SadMBOffset3;
         if (x8 == 2)

media/libstagefright/codecs/mp3dec/Android.mk

diff --git a/media/libstagefright/codecs/mp3dec/Android.mk b/media/libstagefright/codecs/mp3dec/Android.mk
index 135c715..8284490 100644
--- a/media/libstagefright/codecs/mp3dec/Android.mk
+++ b/media/libstagefright/codecs/mp3dec/Android.mk
@@ -50,6 +50,8 @@
 LOCAL_CFLAGS := \
         -DOSCL_UNUSED_ARG=
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_MODULE := libstagefright_mp3dec
 
 LOCAL_ARM_MODE := arm
@@ -69,6 +71,8 @@
         $(LOCAL_PATH)/src \
         $(LOCAL_PATH)/include
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_SHARED_LIBRARIES := \
         libstagefright libstagefright_omx libstagefright_foundation libutils liblog
 

media/libstagefright/codecs/mp3dec/SoftMP3.cpp

diff --git a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
index 877e3cb..a09ab7c 100644
--- a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
+++ b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
@@ -188,7 +188,7 @@
     }
 }
 
-void SoftMP3::onQueueFilled(OMX_U32 portIndex) {
+void SoftMP3::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError || mOutputPortSettingsChange != NONE) {
         return;
     }
@@ -335,6 +335,9 @@
         // depend on fragments from the last one decoded.
         pvmp3_InitDecoder(mConfig, mDecoderBuf);
         mIsFirst = true;
+        mSignalledError = false;
+        mSawInputEos = false;
+        mSignalledOutputEos = false;
     }
 }
 

media/libstagefright/codecs/mp3dec/src/pvmp3_mpeg2_get_scale_data.cpp

diff --git a/media/libstagefright/codecs/mp3dec/src/pvmp3_mpeg2_get_scale_data.cpp b/media/libstagefright/codecs/mp3dec/src/pvmp3_mpeg2_get_scale_data.cpp
index ee42dc5..499672b 100644
--- a/media/libstagefright/codecs/mp3dec/src/pvmp3_mpeg2_get_scale_data.cpp
+++ b/media/libstagefright/codecs/mp3dec/src/pvmp3_mpeg2_get_scale_data.cpp
@@ -139,7 +139,7 @@
     int16 blocknumber = 0;
 
     granuleInfo *gr_info = &(si->ch[ch].gran[gr]);
-    uint32 scalefac_comp, int_scalefac_comp, new_slen[4];
+    uint32 scalefac_comp, int_scalefac_comp, new_slen[4] = { 0,0,0,0 };
 
     scalefac_comp =  gr_info->scalefac_compress;
 

media/libstagefright/codecs/on2/dec/Android.mk

diff --git a/media/libstagefright/codecs/on2/dec/Android.mk b/media/libstagefright/codecs/on2/dec/Android.mk
index 7f2c46d..93ff64c 100644
--- a/media/libstagefright/codecs/on2/dec/Android.mk
+++ b/media/libstagefright/codecs/on2/dec/Android.mk
@@ -20,4 +20,6 @@
 LOCAL_MODULE := libstagefright_soft_vpxdec
 LOCAL_MODULE_TAGS := optional
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_SHARED_LIBRARY)

media/libstagefright/codecs/on2/dec/SoftVPX.cpp

diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
index 476e986..423a057 100644
--- a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
+++ b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
@@ -85,7 +85,7 @@
     return OK;
 }
 
-void SoftVPX::onQueueFilled(OMX_U32 portIndex) {
+void SoftVPX::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mOutputPortSettingsChange != NONE) {
         return;
     }

media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp

diff --git a/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp b/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
index 5f2b5c8..5efe022 100644
--- a/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
+++ b/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
@@ -141,6 +141,7 @@
       mWidth(176),
       mHeight(144),
       mBitrate(192000),  // in bps
+      mBitrateUpdated(false),
       mBitrateControlMode(VPX_VBR),  // variable bitrate
       mFrameDurationUs(33333),  // Defaults to 30 fps
       mDCTPartitions(0),
@@ -536,6 +537,22 @@
             return OMX_ErrorNone;
         }
 
+        case OMX_IndexConfigVideoBitrate:
+        {
+            OMX_VIDEO_CONFIG_BITRATETYPE *params =
+                (OMX_VIDEO_CONFIG_BITRATETYPE *)_params;
+
+            if (params->nPortIndex != kOutputPortIndex) {
+                return OMX_ErrorBadPortIndex;
+            }
+
+            if (mBitrate != params->nEncodeBitrate) {
+                mBitrate = params->nEncodeBitrate;
+                mBitrateUpdated = true;
+            }
+            return OMX_ErrorNone;
+        }
+
         default:
             return SimpleSoftOMXComponent::setConfig(index, _params);
     }
@@ -660,6 +677,9 @@
         def->format.video.nFrameHeight = mHeight;
         def->format.video.xFramerate = port->format.video.xFramerate;
         def->format.video.eColorFormat = mColorFormat;
+        def = &editPortInfo(kOutputPortIndex)->mDef;
+        def->format.video.nFrameWidth = mWidth;
+        def->format.video.nFrameHeight = mHeight;
 
         return OMX_ErrorNone;
     } else if (port->nPortIndex == kOutputPortIndex) {
@@ -779,6 +799,21 @@
             mKeyFrameRequested = false;
         }
 
+        if (mBitrateUpdated) {
+            mCodecConfiguration->rc_target_bitrate = mBitrate/1000;
+            vpx_codec_err_t res = vpx_codec_enc_config_set(mCodecContext,
+                                                           mCodecConfiguration);
+            if (res != VPX_CODEC_OK) {
+                ALOGE("vp8 encoder failed to update bitrate: %s",
+                      vpx_codec_err_to_string(res));
+                notify(OMX_EventError,
+                       OMX_ErrorUndefined,
+                       0, // Extra notification data
+                       NULL); // Notification data pointer
+            }
+            mBitrateUpdated = false;
+        }
+
         codec_return = vpx_codec_encode(
                 mCodecContext,
                 &raw_frame,
@@ -803,6 +838,8 @@
             if (encoded_packet->kind == VPX_CODEC_CX_FRAME_PKT) {
                 outputBufferHeader->nTimeStamp = encoded_packet->data.frame.pts;
                 outputBufferHeader->nFlags = 0;
+                if (encoded_packet->data.frame.flags & VPX_FRAME_IS_KEY)
+                  outputBufferHeader->nFlags |= OMX_BUFFERFLAG_SYNCFRAME;
                 outputBufferHeader->nOffset = 0;
                 outputBufferHeader->nFilledLen = encoded_packet->data.frame.sz;
                 memcpy(outputBufferHeader->pBuffer,

media/libstagefright/codecs/on2/enc/SoftVPXEncoder.h

diff --git a/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.h b/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.h
index 4ee5e51..076830f 100644
--- a/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.h
+++ b/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.h
@@ -128,7 +128,10 @@
     int32_t mHeight;
 
     // Target bitrate set for the encoder, in bits per second.
-    int32_t mBitrate;
+    uint32_t mBitrate;
+
+    // If a request for a change it bitrate has been received.
+    bool mBitrateUpdated;
 
     // Bitrate control mode, either constant or variable
     vpx_rc_mode mBitrateControlMode;

media/libstagefright/codecs/on2/h264dec/SoftAVC.cpp

diff --git a/media/libstagefright/codecs/on2/h264dec/SoftAVC.cpp b/media/libstagefright/codecs/on2/h264dec/SoftAVC.cpp
index 7ddb13c..a7bde97 100644
--- a/media/libstagefright/codecs/on2/h264dec/SoftAVC.cpp
+++ b/media/libstagefright/codecs/on2/h264dec/SoftAVC.cpp
@@ -98,7 +98,7 @@
     return UNKNOWN_ERROR;
 }
 
-void SoftAVC::onQueueFilled(OMX_U32 portIndex) {
+void SoftAVC::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError || mOutputPortSettingsChange != NONE) {
         return;
     }

media/libstagefright/codecs/on2/h264dec/source/H264SwDecApi.c

diff --git a/media/libstagefright/codecs/on2/h264dec/source/H264SwDecApi.c b/media/libstagefright/codecs/on2/h264dec/source/H264SwDecApi.c
index 2bb4c4d..524a3f0 100644
--- a/media/libstagefright/codecs/on2/h264dec/source/H264SwDecApi.c
+++ b/media/libstagefright/codecs/on2/h264dec/source/H264SwDecApi.c
@@ -42,6 +42,8 @@
 #include "h264bsd_decoder.h"
 #include "h264bsd_util.h"
 
+#define UNUSED(x) (void)(x)
+
 /*------------------------------------------------------------------------------
        Version Information
 ------------------------------------------------------------------------------*/
@@ -73,6 +75,7 @@
 #endif
 
 void H264SwDecTrace(char *string) {
+    UNUSED(string);
 }
 
 void* H264SwDecMalloc(u32 size) {

media/libstagefright/codecs/on2/h264dec/source/h264bsd_conceal.c

diff --git a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_conceal.c b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_conceal.c
index 493fb9e..7a262ed 100755
--- a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_conceal.c
+++ b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_conceal.c
@@ -267,7 +267,7 @@
     i32 firstPhase[16];
     i32 *pTmp;
     /* neighbours above, below, left and right */
-    i32 a[4], b[4], l[4], r[4];
+    i32 a[4] = { 0,0,0,0 }, b[4], l[4] = { 0,0,0,0 }, r[4];
     u32 A, B, L, R;
 #ifdef H264DEC_OMXDL
     u8 fillBuff[32*21 + 15 + 32];

media/libstagefright/codecs/on2/h264dec/source/h264bsd_intra_prediction.c

diff --git a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_intra_prediction.c b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_intra_prediction.c
index 15eabfb..52c85e5 100755
--- a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_intra_prediction.c
+++ b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_intra_prediction.c
@@ -1110,7 +1110,7 @@
 
 /* Variables */
 
-    u32 i, j;
+    i32 i, j;
     i32 a, b, c;
     i32 tmp;
 
@@ -1123,20 +1123,20 @@
     a = 16 * (above[15] + left[15]);
 
     for (i = 0, b = 0; i < 8; i++)
-        b += ((i32)i + 1) * (above[8+i] - above[6-i]);
+        b += (i + 1) * (above[8+i] - above[6-i]);
     b = (5 * b + 32) >> 6;
 
     for (i = 0, c = 0; i < 7; i++)
-        c += ((i32)i + 1) * (left[8+i] - left[6-i]);
+        c += (i + 1) * (left[8+i] - left[6-i]);
     /* p[-1,-1] has to be accessed through above pointer */
-    c += ((i32)i + 1) * (left[8+i] - above[-1]);
+    c += (i + 1) * (left[8+i] - above[-1]);
     c = (5 * c + 32) >> 6;
 
     for (i = 0; i < 16; i++)
     {
         for (j = 0; j < 16; j++)
         {
-            tmp = (a + b * ((i32)j - 7) + c * ((i32)i - 7) + 16) >> 5;
+            tmp = (a + b * (j - 7) + c * (i - 7) + 16) >> 5;
             data[i*16+j] = (u8)CLIP1(tmp);
         }
     }

media/libstagefright/codecs/on2/h264dec/source/h264bsd_reconstruct.c

diff --git a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_reconstruct.c b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_reconstruct.c
index c948776..b409a06 100755
--- a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_reconstruct.c
+++ b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_reconstruct.c
@@ -42,6 +42,8 @@
 #include "armVC.h"
 #endif /* H264DEC_OMXDL */
 
+#define UNUSED(x) (void)(x)
+
 /*------------------------------------------------------------------------------
     2. External compiler flags
 --------------------------------------------------------------------------------
@@ -2136,7 +2138,8 @@
   i32 center,
   i32 right)
 {
-
+    UNUSED(left);
+    UNUSED(right);
     ASSERT(ref);
     ASSERT(fill);
 

media/libstagefright/codecs/on2/h264dec/source/h264bsd_slice_header.c

diff --git a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_slice_header.c b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_slice_header.c
index a7c6f64..23401c6 100755
--- a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_slice_header.c
+++ b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_slice_header.c
@@ -47,6 +47,8 @@
 #include "h264bsd_nal_unit.h"
 #include "h264bsd_dpb.h"
 
+#define UNUSED(x) (void)(x)
+
 /*------------------------------------------------------------------------------
     2. External compiler flags
 --------------------------------------------------------------------------------
@@ -1407,6 +1409,7 @@
     u32 tmp, value, i;
     i32 ivalue;
     strmData_t tmpStrmData[1];
+    UNUSED(nalUnitType);
 
 /* Code */
 

media/libstagefright/codecs/on2/h264dec/source/h264bsd_util.c

diff --git a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_util.c b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_util.c
index cc838fd..fb97a28 100755
--- a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_util.c
+++ b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_util.c
@@ -186,7 +186,7 @@
         return(HANTRO_FALSE);
 
     if ( (bits > 8) ||
-         ((h264bsdShowBits32(pStrmData)>>(32-bits)) != (1 << (bits-1))) )
+         ((h264bsdShowBits32(pStrmData)>>(32-bits)) != (1ul << (bits-1))) )
         return(HANTRO_TRUE);
     else
         return(HANTRO_FALSE);

media/libstagefright/codecs/on2/h264dec/source/h264bsd_util.h

diff --git a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_util.h b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_util.h
index cb3adda..216ad04 100755
--- a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_util.h
+++ b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_util.h
@@ -42,6 +42,7 @@
 #include <stdio.h>
 #endif
 
+#include <stdint.h>
 #include "basetype.h"
 #include "h264bsd_stream.h"
 #include "h264bsd_image.h"
@@ -150,7 +151,7 @@
 }
 
 #define ALIGN(ptr, bytePos) \
-        (ptr + ( ((bytePos - (int)ptr) & (bytePos - 1)) / sizeof(*ptr) ))
+        (ptr + ( ((bytePos - (uintptr_t)ptr) & (bytePos - 1)) / sizeof(*ptr) ))
 
 extern const u32 h264bsdQpC[52];
 

media/libstagefright/codecs/opus/Android.mk

diff --git a/media/libstagefright/codecs/opus/Android.mk b/media/libstagefright/codecs/opus/Android.mk
new file mode 100644
index 0000000..365b179
--- /dev/null
+++ b/media/libstagefright/codecs/opus/Android.mk
@@ -0,0 +1,4 @@
+LOCAL_PATH:= $(call my-dir)
+include $(CLEAR_VARS)
+
+include $(call all-makefiles-under,$(LOCAL_PATH))
\ No newline at end of file

media/libstagefright/codecs/opus/dec/Android.mk

diff --git a/media/libstagefright/codecs/opus/dec/Android.mk b/media/libstagefright/codecs/opus/dec/Android.mk
new file mode 100644
index 0000000..2379c5f
--- /dev/null
+++ b/media/libstagefright/codecs/opus/dec/Android.mk
@@ -0,0 +1,19 @@
+LOCAL_PATH:= $(call my-dir)
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+        SoftOpus.cpp
+
+LOCAL_C_INCLUDES := \
+        external/libopus/include \
+        frameworks/av/media/libstagefright/include \
+        frameworks/native/include/media/openmax \
+
+LOCAL_SHARED_LIBRARIES := \
+        libopus libstagefright libstagefright_omx \
+        libstagefright_foundation libutils liblog
+
+LOCAL_MODULE := libstagefright_soft_opusdec
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_SHARED_LIBRARY)
\ No newline at end of file

media/libstagefright/codecs/opus/dec/SoftOpus.cpp

diff --git a/media/libstagefright/codecs/opus/dec/SoftOpus.cpp b/media/libstagefright/codecs/opus/dec/SoftOpus.cpp
new file mode 100644
index 0000000..b8084ae
--- /dev/null
+++ b/media/libstagefright/codecs/opus/dec/SoftOpus.cpp
@@ -0,0 +1,540 @@
+/*
+ * 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 "SoftOpus"
+#include <utils/Log.h>
+
+#include "SoftOpus.h"
+#include <OMX_AudioExt.h>
+#include <OMX_IndexExt.h>
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/MediaDefs.h>
+
+extern "C" {
+    #include <opus.h>
+    #include <opus_multistream.h>
+}
+
+namespace android {
+
+static const int kRate = 48000;
+
+template<class T>
+static void InitOMXParams(T *params) {
+    params->nSize = sizeof(T);
+    params->nVersion.s.nVersionMajor = 1;
+    params->nVersion.s.nVersionMinor = 0;
+    params->nVersion.s.nRevision = 0;
+    params->nVersion.s.nStep = 0;
+}
+
+SoftOpus::SoftOpus(
+        const char *name,
+        const OMX_CALLBACKTYPE *callbacks,
+        OMX_PTR appData,
+        OMX_COMPONENTTYPE **component)
+    : SimpleSoftOMXComponent(name, callbacks, appData, component),
+      mInputBufferCount(0),
+      mDecoder(NULL),
+      mHeader(NULL),
+      mCodecDelay(0),
+      mSeekPreRoll(0),
+      mAnchorTimeUs(0),
+      mNumFramesOutput(0),
+      mOutputPortSettingsChange(NONE) {
+    initPorts();
+    CHECK_EQ(initDecoder(), (status_t)OK);
+}
+
+SoftOpus::~SoftOpus() {
+    if (mDecoder != NULL) {
+        opus_multistream_decoder_destroy(mDecoder);
+        mDecoder = NULL;
+    }
+    if (mHeader != NULL) {
+        delete mHeader;
+        mHeader = NULL;
+    }
+}
+
+void SoftOpus::initPorts() {
+    OMX_PARAM_PORTDEFINITIONTYPE def;
+    InitOMXParams(&def);
+
+    def.nPortIndex = 0;
+    def.eDir = OMX_DirInput;
+    def.nBufferCountMin = kNumBuffers;
+    def.nBufferCountActual = def.nBufferCountMin;
+    def.nBufferSize = 960 * 6;
+    def.bEnabled = OMX_TRUE;
+    def.bPopulated = OMX_FALSE;
+    def.eDomain = OMX_PortDomainAudio;
+    def.bBuffersContiguous = OMX_FALSE;
+    def.nBufferAlignment = 1;
+
+    def.format.audio.cMIMEType =
+        const_cast<char *>(MEDIA_MIMETYPE_AUDIO_OPUS);
+
+    def.format.audio.pNativeRender = NULL;
+    def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+    def.format.audio.eEncoding =
+        (OMX_AUDIO_CODINGTYPE)OMX_AUDIO_CodingAndroidOPUS;
+
+    addPort(def);
+
+    def.nPortIndex = 1;
+    def.eDir = OMX_DirOutput;
+    def.nBufferCountMin = kNumBuffers;
+    def.nBufferCountActual = def.nBufferCountMin;
+    def.nBufferSize = kMaxNumSamplesPerBuffer * sizeof(int16_t);
+    def.bEnabled = OMX_TRUE;
+    def.bPopulated = OMX_FALSE;
+    def.eDomain = OMX_PortDomainAudio;
+    def.bBuffersContiguous = OMX_FALSE;
+    def.nBufferAlignment = 2;
+
+    def.format.audio.cMIMEType = const_cast<char *>("audio/raw");
+    def.format.audio.pNativeRender = NULL;
+    def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+    def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
+
+    addPort(def);
+}
+
+status_t SoftOpus::initDecoder() {
+    return OK;
+}
+
+OMX_ERRORTYPE SoftOpus::internalGetParameter(
+        OMX_INDEXTYPE index, OMX_PTR params) {
+    switch ((int)index) {
+        case OMX_IndexParamAudioAndroidOpus:
+        {
+            OMX_AUDIO_PARAM_ANDROID_OPUSTYPE *opusParams =
+                (OMX_AUDIO_PARAM_ANDROID_OPUSTYPE *)params;
+
+            if (opusParams->nPortIndex != 0) {
+                return OMX_ErrorUndefined;
+            }
+
+            opusParams->nAudioBandWidth = 0;
+            opusParams->nSampleRate = kRate;
+            opusParams->nBitRate = 0;
+
+            if (!isConfigured()) {
+                opusParams->nChannels = 1;
+            } else {
+                opusParams->nChannels = mHeader->channels;
+            }
+
+            return OMX_ErrorNone;
+        }
+
+        case OMX_IndexParamAudioPcm:
+        {
+            OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+                (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+            if (pcmParams->nPortIndex != 1) {
+                return OMX_ErrorUndefined;
+            }
+
+            pcmParams->eNumData = OMX_NumericalDataSigned;
+            pcmParams->eEndian = OMX_EndianBig;
+            pcmParams->bInterleaved = OMX_TRUE;
+            pcmParams->nBitPerSample = 16;
+            pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
+            pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelLF;
+            pcmParams->eChannelMapping[1] = OMX_AUDIO_ChannelRF;
+            pcmParams->nSamplingRate = kRate;
+
+            if (!isConfigured()) {
+                pcmParams->nChannels = 1;
+            } else {
+                pcmParams->nChannels = mHeader->channels;
+            }
+
+            return OMX_ErrorNone;
+        }
+
+        default:
+            return SimpleSoftOMXComponent::internalGetParameter(index, params);
+    }
+}
+
+OMX_ERRORTYPE SoftOpus::internalSetParameter(
+        OMX_INDEXTYPE index, const OMX_PTR params) {
+    switch ((int)index) {
+        case OMX_IndexParamStandardComponentRole:
+        {
+            const OMX_PARAM_COMPONENTROLETYPE *roleParams =
+                (const OMX_PARAM_COMPONENTROLETYPE *)params;
+
+            if (strncmp((const char *)roleParams->cRole,
+                        "audio_decoder.opus",
+                        OMX_MAX_STRINGNAME_SIZE - 1)) {
+                return OMX_ErrorUndefined;
+            }
+
+            return OMX_ErrorNone;
+        }
+
+        case OMX_IndexParamAudioAndroidOpus:
+        {
+            const OMX_AUDIO_PARAM_ANDROID_OPUSTYPE *opusParams =
+                (const OMX_AUDIO_PARAM_ANDROID_OPUSTYPE *)params;
+
+            if (opusParams->nPortIndex != 0) {
+                return OMX_ErrorUndefined;
+            }
+
+            return OMX_ErrorNone;
+        }
+
+        default:
+            return SimpleSoftOMXComponent::internalSetParameter(index, params);
+    }
+}
+
+bool SoftOpus::isConfigured() const {
+    return mInputBufferCount >= 1;
+}
+
+static uint16_t ReadLE16(const uint8_t *data, size_t data_size,
+                         uint32_t read_offset) {
+    if (read_offset + 1 > data_size)
+        return 0;
+    uint16_t val;
+    val = data[read_offset];
+    val |= data[read_offset + 1] << 8;
+    return val;
+}
+
+// Opus uses Vorbis channel mapping, and Vorbis channel mapping specifies
+// mappings for up to 8 channels. This information is part of the Vorbis I
+// Specification:
+// http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html
+static const int kMaxChannels = 8;
+
+// Maximum packet size used in Xiph's opusdec.
+static const int kMaxOpusOutputPacketSizeSamples = 960 * 6;
+
+// Default audio output channel layout. Used to initialize |stream_map| in
+// OpusHeader, and passed to opus_multistream_decoder_create() when the header
+// does not contain mapping information. The values are valid only for mono and
+// stereo output: Opus streams with more than 2 channels require a stream map.
+static const int kMaxChannelsWithDefaultLayout = 2;
+static const uint8_t kDefaultOpusChannelLayout[kMaxChannelsWithDefaultLayout] = { 0, 1 };
+
+// Parses Opus Header. Header spec: http://wiki.xiph.org/OggOpus#ID_Header
+static bool ParseOpusHeader(const uint8_t *data, size_t data_size,
+                            OpusHeader* header) {
+    // Size of the Opus header excluding optional mapping information.
+    const size_t kOpusHeaderSize = 19;
+
+    // Offset to the channel count byte in the Opus header.
+    const size_t kOpusHeaderChannelsOffset = 9;
+
+    // Offset to the pre-skip value in the Opus header.
+    const size_t kOpusHeaderSkipSamplesOffset = 10;
+
+    // Offset to the gain value in the Opus header.
+    const size_t kOpusHeaderGainOffset = 16;
+
+    // Offset to the channel mapping byte in the Opus header.
+    const size_t kOpusHeaderChannelMappingOffset = 18;
+
+    // Opus Header contains a stream map. The mapping values are in the header
+    // beyond the always present |kOpusHeaderSize| bytes of data. The mapping
+    // data contains stream count, coupling information, and per channel mapping
+    // values:
+    //   - Byte 0: Number of streams.
+    //   - Byte 1: Number coupled.
+    //   - Byte 2: Starting at byte 2 are |header->channels| uint8 mapping
+    //             values.
+    const size_t kOpusHeaderNumStreamsOffset = kOpusHeaderSize;
+    const size_t kOpusHeaderNumCoupledOffset = kOpusHeaderNumStreamsOffset + 1;
+    const size_t kOpusHeaderStreamMapOffset = kOpusHeaderNumStreamsOffset + 2;
+
+    if (data_size < kOpusHeaderSize) {
+        ALOGV("Header size is too small.");
+        return false;
+    }
+    header->channels = *(data + kOpusHeaderChannelsOffset);
+
+    if (header->channels <= 0 || header->channels > kMaxChannels) {
+        ALOGV("Invalid Header, wrong channel count: %d", header->channels);
+        return false;
+    }
+    header->skip_samples = ReadLE16(data, data_size,
+                                        kOpusHeaderSkipSamplesOffset);
+    header->gain_db = static_cast<int16_t>(
+                              ReadLE16(data, data_size,
+                                       kOpusHeaderGainOffset));
+    header->channel_mapping = *(data + kOpusHeaderChannelMappingOffset);
+    if (!header->channel_mapping) {
+        if (header->channels > kMaxChannelsWithDefaultLayout) {
+            ALOGV("Invalid Header, missing stream map.");
+            return false;
+        }
+        header->num_streams = 1;
+        header->num_coupled = header->channels > 1;
+        header->stream_map[0] = 0;
+        header->stream_map[1] = 1;
+        return true;
+    }
+    if (data_size < kOpusHeaderStreamMapOffset + header->channels) {
+        ALOGV("Invalid stream map; insufficient data for current channel "
+              "count: %d", header->channels);
+        return false;
+    }
+    header->num_streams = *(data + kOpusHeaderNumStreamsOffset);
+    header->num_coupled = *(data + kOpusHeaderNumCoupledOffset);
+    if (header->num_streams + header->num_coupled != header->channels) {
+        ALOGV("Inconsistent channel mapping.");
+        return false;
+    }
+    for (int i = 0; i < header->channels; ++i)
+      header->stream_map[i] = *(data + kOpusHeaderStreamMapOffset + i);
+    return true;
+}
+
+// Convert nanoseconds to number of samples.
+static uint64_t ns_to_samples(uint64_t ns, int kRate) {
+    return static_cast<double>(ns) * kRate / 1000000000;
+}
+
+void SoftOpus::onQueueFilled(OMX_U32 portIndex) {
+    List<BufferInfo *> &inQueue = getPortQueue(0);
+    List<BufferInfo *> &outQueue = getPortQueue(1);
+
+    if (mOutputPortSettingsChange != NONE) {
+        return;
+    }
+
+    if (portIndex == 0 && mInputBufferCount < 3) {
+        BufferInfo *info = *inQueue.begin();
+        OMX_BUFFERHEADERTYPE *header = info->mHeader;
+
+        const uint8_t *data = header->pBuffer + header->nOffset;
+        size_t size = header->nFilledLen;
+
+        if (mInputBufferCount == 0) {
+            CHECK(mHeader == NULL);
+            mHeader = new OpusHeader();
+            memset(mHeader, 0, sizeof(*mHeader));
+            if (!ParseOpusHeader(data, size, mHeader)) {
+                ALOGV("Parsing Opus Header failed.");
+                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+                return;
+            }
+
+            uint8_t channel_mapping[kMaxChannels] = {0};
+            memcpy(&channel_mapping,
+                   kDefaultOpusChannelLayout,
+                   kMaxChannelsWithDefaultLayout);
+
+            int status = OPUS_INVALID_STATE;
+            mDecoder = opus_multistream_decoder_create(kRate,
+                                                       mHeader->channels,
+                                                       mHeader->num_streams,
+                                                       mHeader->num_coupled,
+                                                       channel_mapping,
+                                                       &status);
+            if (!mDecoder || status != OPUS_OK) {
+                ALOGV("opus_multistream_decoder_create failed status=%s",
+                      opus_strerror(status));
+                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+                return;
+            }
+            status =
+                opus_multistream_decoder_ctl(mDecoder,
+                                             OPUS_SET_GAIN(mHeader->gain_db));
+            if (status != OPUS_OK) {
+                ALOGV("Failed to set OPUS header gain; status=%s",
+                      opus_strerror(status));
+                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+                return;
+            }
+        } else if (mInputBufferCount == 1) {
+            mCodecDelay = ns_to_samples(
+                              *(reinterpret_cast<int64_t*>(header->pBuffer +
+                                                           header->nOffset)),
+                              kRate);
+            mSamplesToDiscard = mCodecDelay;
+        } else {
+            mSeekPreRoll = ns_to_samples(
+                               *(reinterpret_cast<int64_t*>(header->pBuffer +
+                                                            header->nOffset)),
+                               kRate);
+            notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
+            mOutputPortSettingsChange = AWAITING_DISABLED;
+        }
+
+        inQueue.erase(inQueue.begin());
+        info->mOwnedByUs = false;
+        notifyEmptyBufferDone(header);
+        ++mInputBufferCount;
+        return;
+    }
+
+    while (!inQueue.empty() && !outQueue.empty()) {
+        BufferInfo *inInfo = *inQueue.begin();
+        OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+
+        BufferInfo *outInfo = *outQueue.begin();
+        OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+
+        if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+            inQueue.erase(inQueue.begin());
+            inInfo->mOwnedByUs = false;
+            notifyEmptyBufferDone(inHeader);
+
+            outHeader->nFilledLen = 0;
+            outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+
+            outQueue.erase(outQueue.begin());
+            outInfo->mOwnedByUs = false;
+            notifyFillBufferDone(outHeader);
+            return;
+        }
+
+        if (inHeader->nOffset == 0) {
+            mAnchorTimeUs = inHeader->nTimeStamp;
+            mNumFramesOutput = 0;
+        }
+
+        // When seeking to zero, |mCodecDelay| samples has to be discarded
+        // instead of |mSeekPreRoll| samples (as we would when seeking to any
+        // other timestamp).
+        if (inHeader->nTimeStamp == 0) {
+            mSamplesToDiscard = mCodecDelay;
+        }
+
+        const uint8_t *data = inHeader->pBuffer + inHeader->nOffset;
+        const uint32_t size = inHeader->nFilledLen;
+
+        int numFrames = opus_multistream_decode(mDecoder,
+                                                data,
+                                                size,
+                                                (int16_t *)outHeader->pBuffer,
+                                                kMaxOpusOutputPacketSizeSamples,
+                                                0);
+        if (numFrames < 0) {
+            ALOGE("opus_multistream_decode returned %d", numFrames);
+            notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+            return;
+        }
+
+        outHeader->nOffset = 0;
+        if (mSamplesToDiscard > 0) {
+            if (mSamplesToDiscard > numFrames) {
+                mSamplesToDiscard -= numFrames;
+                numFrames = 0;
+            } else {
+                numFrames -= mSamplesToDiscard;
+                outHeader->nOffset = mSamplesToDiscard * sizeof(int16_t) *
+                                     mHeader->channels;
+                mSamplesToDiscard = 0;
+            }
+        }
+
+        outHeader->nFilledLen = numFrames * sizeof(int16_t) * mHeader->channels;
+        outHeader->nFlags = 0;
+
+        outHeader->nTimeStamp = mAnchorTimeUs +
+                                (mNumFramesOutput * 1000000ll) /
+                                kRate;
+
+        mNumFramesOutput += numFrames;
+
+        inInfo->mOwnedByUs = false;
+        inQueue.erase(inQueue.begin());
+        inInfo = NULL;
+        notifyEmptyBufferDone(inHeader);
+        inHeader = NULL;
+
+        outInfo->mOwnedByUs = false;
+        outQueue.erase(outQueue.begin());
+        outInfo = NULL;
+        notifyFillBufferDone(outHeader);
+        outHeader = NULL;
+
+        ++mInputBufferCount;
+    }
+}
+
+void SoftOpus::onPortFlushCompleted(OMX_U32 portIndex) {
+    if (portIndex == 0 && mDecoder != NULL) {
+        // Make sure that the next buffer output does not still
+        // depend on fragments from the last one decoded.
+        mNumFramesOutput = 0;
+        opus_multistream_decoder_ctl(mDecoder, OPUS_RESET_STATE);
+        mAnchorTimeUs = 0;
+        mSamplesToDiscard = mSeekPreRoll;
+    }
+}
+
+void SoftOpus::onReset() {
+    mInputBufferCount = 0;
+    mNumFramesOutput = 0;
+    if (mDecoder != NULL) {
+        opus_multistream_decoder_destroy(mDecoder);
+        mDecoder = NULL;
+    }
+    if (mHeader != NULL) {
+        delete mHeader;
+        mHeader = NULL;
+    }
+
+    mOutputPortSettingsChange = NONE;
+}
+
+void SoftOpus::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) {
+    if (portIndex != 1) {
+        return;
+    }
+
+    switch (mOutputPortSettingsChange) {
+        case NONE:
+            break;
+
+        case AWAITING_DISABLED:
+        {
+            CHECK(!enabled);
+            mOutputPortSettingsChange = AWAITING_ENABLED;
+            break;
+        }
+
+        default:
+        {
+            CHECK_EQ((int)mOutputPortSettingsChange, (int)AWAITING_ENABLED);
+            CHECK(enabled);
+            mOutputPortSettingsChange = NONE;
+            break;
+        }
+    }
+}
+
+}  // namespace android
+
+android::SoftOMXComponent *createSoftOMXComponent(
+        const char *name, const OMX_CALLBACKTYPE *callbacks,
+        OMX_PTR appData, OMX_COMPONENTTYPE **component) {
+    return new android::SoftOpus(name, callbacks, appData, component);
+}

media/libstagefright/codecs/opus/dec/SoftOpus.h

diff --git a/media/libstagefright/codecs/opus/dec/SoftOpus.h b/media/libstagefright/codecs/opus/dec/SoftOpus.h
new file mode 100644
index 0000000..97f6561
--- /dev/null
+++ b/media/libstagefright/codecs/opus/dec/SoftOpus.h
@@ -0,0 +1,94 @@
+/*
+ * 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.
+ */
+
+/*
+ * The Opus specification is part of IETF RFC 6716:
+ * http://tools.ietf.org/html/rfc6716
+ */
+
+#ifndef SOFT_OPUS_H_
+
+#define SOFT_OPUS_H_
+
+#include "SimpleSoftOMXComponent.h"
+
+struct OpusMSDecoder;
+
+namespace android {
+
+struct OpusHeader {
+  int channels;
+  int skip_samples;
+  int channel_mapping;
+  int num_streams;
+  int num_coupled;
+  int16_t gain_db;
+  uint8_t stream_map[8];
+};
+
+struct SoftOpus : public SimpleSoftOMXComponent {
+    SoftOpus(const char *name,
+             const OMX_CALLBACKTYPE *callbacks,
+             OMX_PTR appData,
+             OMX_COMPONENTTYPE **component);
+
+protected:
+    virtual ~SoftOpus();
+
+    virtual OMX_ERRORTYPE internalGetParameter(
+            OMX_INDEXTYPE index, OMX_PTR params);
+
+    virtual OMX_ERRORTYPE internalSetParameter(
+            OMX_INDEXTYPE index, const OMX_PTR params);
+
+    virtual void onQueueFilled(OMX_U32 portIndex);
+    virtual void onPortFlushCompleted(OMX_U32 portIndex);
+    virtual void onPortEnableCompleted(OMX_U32 portIndex, bool enabled);
+    virtual void onReset();
+
+private:
+    enum {
+        kNumBuffers = 4,
+        kMaxNumSamplesPerBuffer = 960 * 6
+    };
+
+    size_t mInputBufferCount;
+
+    OpusMSDecoder *mDecoder;
+    OpusHeader *mHeader;
+
+    int64_t mCodecDelay;
+    int64_t mSeekPreRoll;
+    int64_t mSamplesToDiscard;
+    int64_t mAnchorTimeUs;
+    int64_t mNumFramesOutput;
+
+    enum {
+        NONE,
+        AWAITING_DISABLED,
+        AWAITING_ENABLED
+    } mOutputPortSettingsChange;
+
+    void initPorts();
+    status_t initDecoder();
+    bool isConfigured() const;
+
+    DISALLOW_EVIL_CONSTRUCTORS(SoftOpus);
+};
+
+}  // namespace android
+
+#endif  // SOFT_OPUS_H_

media/libstagefright/codecs/raw/Android.mk

diff --git a/media/libstagefright/codecs/raw/Android.mk b/media/libstagefright/codecs/raw/Android.mk
index fe90a03..87080e7 100644
--- a/media/libstagefright/codecs/raw/Android.mk
+++ b/media/libstagefright/codecs/raw/Android.mk
@@ -8,6 +8,8 @@
         frameworks/av/media/libstagefright/include \
         frameworks/native/include/media/openmax
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_SHARED_LIBRARIES := \
         libstagefright_omx libstagefright_foundation libutils liblog
 

media/libstagefright/codecs/raw/SoftRaw.cpp

diff --git a/media/libstagefright/codecs/raw/SoftRaw.cpp b/media/libstagefright/codecs/raw/SoftRaw.cpp
index 19d6f13..9d514a6 100644
--- a/media/libstagefright/codecs/raw/SoftRaw.cpp
+++ b/media/libstagefright/codecs/raw/SoftRaw.cpp
@@ -163,7 +163,7 @@
     }
 }
 
-void SoftRaw::onQueueFilled(OMX_U32 portIndex) {
+void SoftRaw::onQueueFilled(OMX_U32 /* portIndex */) {
     if (mSignalledError) {
         return;
     }

media/libstagefright/codecs/vorbis/dec/Android.mk

diff --git a/media/libstagefright/codecs/vorbis/dec/Android.mk b/media/libstagefright/codecs/vorbis/dec/Android.mk
index 2232353..217a6d2 100644
--- a/media/libstagefright/codecs/vorbis/dec/Android.mk
+++ b/media/libstagefright/codecs/vorbis/dec/Android.mk
@@ -16,4 +16,6 @@
 LOCAL_MODULE := libstagefright_soft_vorbisdec
 LOCAL_MODULE_TAGS := optional
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_SHARED_LIBRARY)

media/libstagefright/foundation/ANetworkSession.cpp

diff --git a/media/libstagefright/foundation/ANetworkSession.cpp b/media/libstagefright/foundation/ANetworkSession.cpp
index e629588..08c4a87 100644
--- a/media/libstagefright/foundation/ANetworkSession.cpp
+++ b/media/libstagefright/foundation/ANetworkSession.cpp
@@ -521,7 +521,7 @@
     return err;
 }
 
-void ANetworkSession::Session::dumpFragmentStats(const Fragment &frag) {
+void ANetworkSession::Session::dumpFragmentStats(const Fragment & /* frag */) {
 #if 0
     int64_t nowUs = ALooper::GetNowUs();
     int64_t delayMs = (nowUs - frag.mTimeUs) / 1000ll;

media/libstagefright/foundation/Android.mk

diff --git a/media/libstagefright/foundation/Android.mk b/media/libstagefright/foundation/Android.mk
index ad2dab5..90a6a23 100644
--- a/media/libstagefright/foundation/Android.mk
+++ b/media/libstagefright/foundation/Android.mk
@@ -24,7 +24,7 @@
         libutils          \
         liblog
 
-LOCAL_CFLAGS += -Wno-multichar
+LOCAL_CFLAGS += -Wno-multichar -Werror
 
 LOCAL_MODULE:= libstagefright_foundation
 

media/libstagefright/foundation/base64.cpp

diff --git a/media/libstagefright/foundation/base64.cpp b/media/libstagefright/foundation/base64.cpp
index d5fb4e0..dcf5bef 100644
--- a/media/libstagefright/foundation/base64.cpp
+++ b/media/libstagefright/foundation/base64.cpp
@@ -33,6 +33,10 @@
 
         if (n >= 2 && s.c_str()[n - 2] == '=') {
             padding = 2;
+
+            if (n >= 3 && s.c_str()[n - 3] == '=') {
+                padding = 3;
+            }
         }
     }
 
@@ -71,7 +75,7 @@
         if (((i + 1) % 4) == 0) {
             out[j++] = (accum >> 16);
 
-            if (j < outLen) { out[j++] = (accum >> 8) & 0xff; } 
+            if (j < outLen) { out[j++] = (accum >> 8) & 0xff; }
             if (j < outLen) { out[j++] = accum & 0xff; }
 
             accum = 0;

media/libstagefright/http/Android.mk

diff --git a/media/libstagefright/http/Android.mk b/media/libstagefright/http/Android.mk
new file mode 100644
index 0000000..7f3307d
--- /dev/null
+++ b/media/libstagefright/http/Android.mk
@@ -0,0 +1,28 @@
+LOCAL_PATH:= $(call my-dir)
+
+ifneq ($(TARGET_BUILD_PDK), true)
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:=       \
+        HTTPHelper.cpp          \
+
+LOCAL_C_INCLUDES:= \
+	$(TOP)/frameworks/av/media/libstagefright \
+	$(TOP)/frameworks/native/include/media/openmax \
+	$(TOP)/frameworks/base/core/jni \
+
+LOCAL_SHARED_LIBRARIES := \
+	libstagefright liblog libutils libbinder libstagefright_foundation \
+        libandroid_runtime \
+        libmedia
+
+LOCAL_MODULE:= libstagefright_http_support
+
+LOCAL_CFLAGS += -Wno-multichar
+
+LOCAL_CFLAGS += -Werror
+
+include $(BUILD_SHARED_LIBRARY)
+
+endif

media/libstagefright/http/HTTPHelper.cpp

diff --git a/media/libstagefright/http/HTTPHelper.cpp b/media/libstagefright/http/HTTPHelper.cpp
new file mode 100644
index 0000000..77845e2
--- /dev/null
+++ b/media/libstagefright/http/HTTPHelper.cpp
@@ -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.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "HTTPHelper"
+#include <utils/Log.h>
+
+#include "HTTPHelper.h"
+
+#include "android_runtime/AndroidRuntime.h"
+#include "android_util_Binder.h"
+#include <media/IMediaHTTPService.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <nativehelper/ScopedLocalRef.h>
+#include "jni.h"
+
+namespace android {
+
+sp<IMediaHTTPService> CreateHTTPServiceInCurrentJavaContext() {
+    if (AndroidRuntime::getJavaVM() == NULL) {
+        ALOGE("CreateHTTPServiceInCurrentJavaContext called outside "
+              "JAVA environment.");
+        return NULL;
+    }
+
+    JNIEnv *env = AndroidRuntime::getJNIEnv();
+
+    ScopedLocalRef<jclass> clazz(
+            env, env->FindClass("android/media/MediaHTTPService"));
+    CHECK(clazz.get() != NULL);
+
+    jmethodID constructID = env->GetMethodID(clazz.get(), "<init>", "()V");
+    CHECK(constructID != NULL);
+
+    ScopedLocalRef<jobject> httpServiceObj(
+            env, env->NewObject(clazz.get(), constructID));
+
+    sp<IMediaHTTPService> httpService;
+    if (httpServiceObj.get() != NULL) {
+        jmethodID asBinderID =
+            env->GetMethodID(clazz.get(), "asBinder", "()Landroid/os/IBinder;");
+        CHECK(asBinderID != NULL);
+
+        ScopedLocalRef<jobject> httpServiceBinderObj(
+                env, env->CallObjectMethod(httpServiceObj.get(), asBinderID));
+        CHECK(httpServiceBinderObj.get() != NULL);
+
+        sp<IBinder> binder =
+            ibinderForJavaObject(env, httpServiceBinderObj.get());
+
+        httpService = interface_cast<IMediaHTTPService>(binder);
+    }
+
+    return httpService;
+}
+
+}  // namespace android

media/libstagefright/http/HTTPHelper.h

diff --git a/media/libstagefright/http/HTTPHelper.h b/media/libstagefright/http/HTTPHelper.h
new file mode 100644
index 0000000..8aef115
--- /dev/null
+++ b/media/libstagefright/http/HTTPHelper.h
@@ -0,0 +1,31 @@
+/*
+ * 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 HTTP_HELPER_H_
+
+#define HTTP_HELPER_H_
+
+#include <utils/RefBase.h>
+
+namespace android {
+
+struct IMediaHTTPService;
+
+sp<IMediaHTTPService> CreateHTTPServiceInCurrentJavaContext();
+
+}  // namespace android
+
+#endif  // HTTP_HELPER_H_

media/libstagefright/http/MediaHTTP.cpp

diff --git a/media/libstagefright/http/MediaHTTP.cpp b/media/libstagefright/http/MediaHTTP.cpp
new file mode 100644
index 0000000..2d29913
--- /dev/null
+++ b/media/libstagefright/http/MediaHTTP.cpp
@@ -0,0 +1,205 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MediaHTTP"
+#include <utils/Log.h>
+
+#include <media/stagefright/MediaHTTP.h>
+
+#include <binder/IServiceManager.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/Utils.h>
+
+#include <media/IMediaHTTPConnection.h>
+
+namespace android {
+
+MediaHTTP::MediaHTTP(const sp<IMediaHTTPConnection> &conn)
+    : mInitCheck(NO_INIT),
+      mHTTPConnection(conn),
+      mCachedSizeValid(false),
+      mCachedSize(0ll),
+      mDrmManagerClient(NULL) {
+    mInitCheck = OK;
+}
+
+MediaHTTP::~MediaHTTP() {
+    clearDRMState_l();
+}
+
+status_t MediaHTTP::connect(
+        const char *uri,
+        const KeyedVector<String8, String8> *headers,
+        off64_t /* offset */) {
+    if (mInitCheck != OK) {
+        return mInitCheck;
+    }
+
+    KeyedVector<String8, String8> extHeaders;
+    if (headers != NULL) {
+        extHeaders = *headers;
+    }
+    extHeaders.add(String8("User-Agent"), String8(MakeUserAgent().c_str()));
+
+    bool success = mHTTPConnection->connect(uri, &extHeaders);
+
+    mLastHeaders = extHeaders;
+    mLastURI = uri;
+
+    mCachedSizeValid = false;
+
+    return success ? OK : UNKNOWN_ERROR;
+}
+
+void MediaHTTP::disconnect() {
+    if (mInitCheck != OK) {
+        return;
+    }
+
+    mHTTPConnection->disconnect();
+}
+
+status_t MediaHTTP::initCheck() const {
+    return mInitCheck;
+}
+
+ssize_t MediaHTTP::readAt(off64_t offset, void *data, size_t size) {
+    if (mInitCheck != OK) {
+        return mInitCheck;
+    }
+
+    int64_t startTimeUs = ALooper::GetNowUs();
+
+    size_t numBytesRead = 0;
+    while (numBytesRead < size) {
+        size_t copy = size - numBytesRead;
+
+        if (copy > 64 * 1024) {
+            // limit the buffer sizes transferred across binder boundaries
+            // to avoid spurious transaction failures.
+            copy = 64 * 1024;
+        }
+
+        ssize_t n = mHTTPConnection->readAt(
+                offset + numBytesRead, (uint8_t *)data + numBytesRead, copy);
+
+        if (n < 0) {
+            return n;
+        } else if (n == 0) {
+            break;
+        }
+
+        numBytesRead += n;
+    }
+
+    int64_t delayUs = ALooper::GetNowUs() - startTimeUs;
+
+    addBandwidthMeasurement(numBytesRead, delayUs);
+
+    return numBytesRead;
+}
+
+status_t MediaHTTP::getSize(off64_t *size) {
+    if (mInitCheck != OK) {
+        return mInitCheck;
+    }
+
+    // Caching the returned size so that it stays valid even after a
+    // disconnect. NuCachedSource2 relies on this.
+
+    if (!mCachedSizeValid) {
+        mCachedSize = mHTTPConnection->getSize();
+        mCachedSizeValid = true;
+    }
+
+    *size = mCachedSize;
+
+    return *size < 0 ? *size : OK;
+}
+
+uint32_t MediaHTTP::flags() {
+    return kWantsPrefetching | kIsHTTPBasedSource;
+}
+
+status_t MediaHTTP::reconnectAtOffset(off64_t offset) {
+    return connect(mLastURI.c_str(), &mLastHeaders, offset);
+}
+
+// DRM...
+
+sp<DecryptHandle> MediaHTTP::DrmInitialization(const char* mime) {
+    if (mDrmManagerClient == NULL) {
+        mDrmManagerClient = new DrmManagerClient();
+    }
+
+    if (mDrmManagerClient == NULL) {
+        return NULL;
+    }
+
+    if (mDecryptHandle == NULL) {
+        mDecryptHandle = mDrmManagerClient->openDecryptSession(
+                String8(mLastURI.c_str()), mime);
+    }
+
+    if (mDecryptHandle == NULL) {
+        delete mDrmManagerClient;
+        mDrmManagerClient = NULL;
+    }
+
+    return mDecryptHandle;
+}
+
+void MediaHTTP::getDrmInfo(
+        sp<DecryptHandle> &handle, DrmManagerClient **client) {
+    handle = mDecryptHandle;
+    *client = mDrmManagerClient;
+}
+
+String8 MediaHTTP::getUri() {
+    String8 uri;
+    if (OK == mHTTPConnection->getUri(&uri)) {
+        return uri;
+    }
+    return String8(mLastURI.c_str());
+}
+
+String8 MediaHTTP::getMIMEType() const {
+    if (mInitCheck != OK) {
+        return String8("application/octet-stream");
+    }
+
+    String8 mimeType;
+    status_t err = mHTTPConnection->getMIMEType(&mimeType);
+
+    if (err != OK) {
+        return String8("application/octet-stream");
+    }
+
+    return mimeType;
+}
+
+void MediaHTTP::clearDRMState_l() {
+    if (mDecryptHandle != NULL) {
+        // To release mDecryptHandle
+        CHECK(mDrmManagerClient);
+        mDrmManagerClient->closeDecryptSession(mDecryptHandle);
+        mDecryptHandle = NULL;
+    }
+}
+
+}  // namespace android

media/libstagefright/httplive/Android.mk

diff --git a/media/libstagefright/httplive/Android.mk b/media/libstagefright/httplive/Android.mk
index f3529f9..e8d558c 100644
--- a/media/libstagefright/httplive/Android.mk
+++ b/media/libstagefright/httplive/Android.mk
@@ -13,6 +13,8 @@
 	$(TOP)/frameworks/native/include/media/openmax \
 	$(TOP)/external/openssl/include
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_SHARED_LIBRARIES := \
         libbinder \
         libcrypto \

media/libstagefright/httplive/LiveSession.cpp

diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index bd12ddc..2af4682 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -27,6 +27,8 @@
 #include "mpeg2ts/AnotherPacketSource.h"
 
 #include <cutils/properties.h>
+#include <media/IMediaHTTPConnection.h>
+#include <media/IMediaHTTPService.h>
 #include <media/stagefright/foundation/hexdump.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -34,9 +36,12 @@
 #include <media/stagefright/DataSource.h>
 #include <media/stagefright/FileSource.h>
 #include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaHTTP.h>
 #include <media/stagefright/MetaData.h>
 #include <media/stagefright/Utils.h>
 
+#include <utils/Mutex.h>
+
 #include <ctype.h>
 #include <openssl/aes.h>
 #include <openssl/md5.h>
@@ -44,45 +49,62 @@
 namespace android {
 
 LiveSession::LiveSession(
-        const sp<AMessage> &notify, uint32_t flags, bool uidValid, uid_t uid)
+        const sp<AMessage> &notify, uint32_t flags,
+        const sp<IMediaHTTPService> &httpService)
     : mNotify(notify),
       mFlags(flags),
-      mUIDValid(uidValid),
-      mUID(uid),
+      mHTTPService(httpService),
       mInPreparationPhase(true),
-      mHTTPDataSource(
-              HTTPBase::Create(
-                  (mFlags & kFlagIncognito)
-                    ? HTTPBase::kFlagIncognito
-                    : 0)),
+      mHTTPDataSource(new MediaHTTP(mHTTPService->makeHTTPConnection())),
       mPrevBandwidthIndex(-1),
       mStreamMask(0),
+      mNewStreamMask(0),
+      mSwapMask(0),
       mCheckBandwidthGeneration(0),
+      mSwitchGeneration(0),
       mLastDequeuedTimeUs(0ll),
       mRealTimeBaseUs(0ll),
       mReconfigurationInProgress(false),
+      mSwitchInProgress(false),
       mDisconnectReplyID(0) {
-    if (mUIDValid) {
-        mHTTPDataSource->setUID(mUID);
+
+    mStreams[kAudioIndex] = StreamItem("audio");
+    mStreams[kVideoIndex] = StreamItem("video");
+    mStreams[kSubtitleIndex] = StreamItem("subtitle");
+
+    for (size_t i = 0; i < kMaxStreams; ++i) {
+        mPacketSources.add(indexToType(i), new AnotherPacketSource(NULL /* meta */));
+        mPacketSources2.add(indexToType(i), new AnotherPacketSource(NULL /* meta */));
     }
-
-    mPacketSources.add(
-            STREAMTYPE_AUDIO, new AnotherPacketSource(NULL /* meta */));
-
-    mPacketSources.add(
-            STREAMTYPE_VIDEO, new AnotherPacketSource(NULL /* meta */));
-
-    mPacketSources.add(
-            STREAMTYPE_SUBTITLES, new AnotherPacketSource(NULL /* meta */));
 }
 
 LiveSession::~LiveSession() {
 }
 
+sp<ABuffer> LiveSession::createFormatChangeBuffer(bool swap) {
+    ABuffer *discontinuity = new ABuffer(0);
+    discontinuity->meta()->setInt32("discontinuity", ATSParser::DISCONTINUITY_FORMATCHANGE);
+    discontinuity->meta()->setInt32("swapPacketSource", swap);
+    discontinuity->meta()->setInt32("switchGeneration", mSwitchGeneration);
+    discontinuity->meta()->setInt64("timeUs", -1);
+    return discontinuity;
+}
+
+void LiveSession::swapPacketSource(StreamType stream) {
+    sp<AnotherPacketSource> &aps = mPacketSources.editValueFor(stream);
+    sp<AnotherPacketSource> &aps2 = mPacketSources2.editValueFor(stream);
+    sp<AnotherPacketSource> tmp = aps;
+    aps = aps2;
+    aps2 = tmp;
+    aps2->clear();
+}
+
 status_t LiveSession::dequeueAccessUnit(
         StreamType stream, sp<ABuffer> *accessUnit) {
     if (!(mStreamMask & stream)) {
-        return UNKNOWN_ERROR;
+        // return -EWOULDBLOCK to avoid halting the decoder
+        // when switching between audio/video and audio only.
+        return -EWOULDBLOCK;
     }
 
     sp<AnotherPacketSource> packetSource = mPacketSources.valueFor(stream);
@@ -122,6 +144,25 @@
               streamStr,
               type,
               extra == NULL ? "NULL" : extra->debugString().c_str());
+
+        int32_t swap;
+        if (type == ATSParser::DISCONTINUITY_FORMATCHANGE
+                && (*accessUnit)->meta()->findInt32("swapPacketSource", &swap)
+                && swap) {
+
+            int32_t switchGeneration;
+            CHECK((*accessUnit)->meta()->findInt32("switchGeneration", &switchGeneration));
+            {
+                Mutex::Autolock lock(mSwapMutex);
+                if (switchGeneration == mSwitchGeneration) {
+                    swapPacketSource(stream);
+                    sp<AMessage> msg = new AMessage(kWhatSwapped, id());
+                    msg->setInt32("stream", stream);
+                    msg->setInt32("switchGeneration", switchGeneration);
+                    msg->post();
+                }
+            }
+        }
     } else if (err == OK) {
         if (stream == STREAMTYPE_AUDIO || stream == STREAMTYPE_VIDEO) {
             int64_t timeUs;
@@ -143,6 +184,7 @@
 }
 
 status_t LiveSession::getStreamFormat(StreamType stream, sp<AMessage> *format) {
+    // No swapPacketSource race condition; called from the same thread as dequeueAccessUnit.
     if (!(mStreamMask & stream)) {
         return UNKNOWN_ERROR;
     }
@@ -239,7 +281,12 @@
                     if (what == PlaylistFetcher::kWhatStopped) {
                         AString uri;
                         CHECK(msg->findString("uri", &uri));
-                        mFetcherInfos.removeItem(uri);
+                        if (mFetcherInfos.removeItem(uri) < 0) {
+                            // ignore duplicated kWhatStopped messages.
+                            break;
+                        }
+
+                        tryToFinishBandwidthSwitch();
                     }
 
                     if (mContinuation != NULL) {
@@ -275,6 +322,8 @@
                         postPrepared(err);
                     }
 
+                    cancelBandwidthSwitch();
+
                     mPacketSources.valueFor(STREAMTYPE_AUDIO)->signalEOS(err);
 
                     mPacketSources.valueFor(STREAMTYPE_VIDEO)->signalEOS(err);
@@ -313,6 +362,27 @@
                     break;
                 }
 
+                case PlaylistFetcher::kWhatStartedAt:
+                {
+                    int32_t switchGeneration;
+                    CHECK(msg->findInt32("switchGeneration", &switchGeneration));
+
+                    if (switchGeneration != mSwitchGeneration) {
+                        break;
+                    }
+
+                    // Resume fetcher for the original variant; the resumed fetcher should
+                    // continue until the timestamps found in msg, which is stored by the
+                    // new fetcher to indicate where the new variant has started buffering.
+                    for (size_t i = 0; i < mFetcherInfos.size(); i++) {
+                        const FetcherInfo info = mFetcherInfos.valueAt(i);
+                        if (info.mToBeRemoved) {
+                            info.mFetcher->resumeUntilAsync(msg);
+                        }
+                    }
+                    break;
+                }
+
                 default:
                     TRESPASS();
             }
@@ -357,6 +427,11 @@
             break;
         }
 
+        case kWhatSwapped:
+        {
+            onSwapped(msg);
+            break;
+        }
         default:
             TRESPASS();
             break;
@@ -374,6 +449,12 @@
     return 1;
 }
 
+// static
+LiveSession::StreamType LiveSession::indexToType(int idx) {
+    CHECK(idx >= 0 && idx < kMaxStreams);
+    return (StreamType)(1 << idx);
+}
+
 void LiveSession::onConnect(const sp<AMessage> &msg) {
     AString url;
     CHECK(msg->findString("url", &url));
@@ -461,6 +542,10 @@
     // during disconnection either.
     cancelCheckBandwidthEvent();
 
+    // Protect mPacketSources from a swapPacketSource race condition through disconnect.
+    // (finishDisconnect, onFinishDisconnect2)
+    cancelBandwidthSwitch();
+
     for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
         mFetcherInfos.valueAt(i).mFetcher->stopAsync();
     }
@@ -500,11 +585,13 @@
 
     sp<AMessage> notify = new AMessage(kWhatFetcherNotify, id());
     notify->setString("uri", uri);
+    notify->setInt32("switchGeneration", mSwitchGeneration);
 
     FetcherInfo info;
     info.mFetcher = new PlaylistFetcher(notify, this, uri);
     info.mDurationUs = -1ll;
     info.mIsPrepared = false;
+    info.mToBeRemoved = false;
     looper()->registerHandler(info.mFetcher);
 
     mFetcherInfos.add(uri, info);
@@ -512,53 +599,81 @@
     return info.mFetcher;
 }
 
+/*
+ * Illustration of parameters:
+ *
+ * 0      `range_offset`
+ * +------------+-------------------------------------------------------+--+--+
+ * |            |                                 | next block to fetch |  |  |
+ * |            | `source` handle => `out` buffer |                     |  |  |
+ * | `url` file |<--------- buffer size --------->|<--- `block_size` -->|  |  |
+ * |            |<----------- `range_length` / buffer capacity ----------->|  |
+ * |<------------------------------ file_size ------------------------------->|
+ *
+ * Special parameter values:
+ * - range_length == -1 means entire file
+ * - block_size == 0 means entire range
+ *
+ */
 status_t LiveSession::fetchFile(
         const char *url, sp<ABuffer> *out,
-        int64_t range_offset, int64_t range_length) {
-    *out = NULL;
-
-    sp<DataSource> source;
-
-    if (!strncasecmp(url, "file://", 7)) {
-        source = new FileSource(url + 7);
-    } else if (strncasecmp(url, "http://", 7)
-            && strncasecmp(url, "https://", 8)) {
-        return ERROR_UNSUPPORTED;
-    } else {
-        KeyedVector<String8, String8> headers = mExtraHeaders;
-        if (range_offset > 0 || range_length >= 0) {
-            headers.add(
-                    String8("Range"),
-                    String8(
-                        StringPrintf(
-                            "bytes=%lld-%s",
-                            range_offset,
-                            range_length < 0
-                                ? "" : StringPrintf("%lld", range_offset + range_length - 1).c_str()).c_str()));
-        }
-        status_t err = mHTTPDataSource->connect(url, &headers);
-
-        if (err != OK) {
-            return err;
-        }
-
-        source = mHTTPDataSource;
+        int64_t range_offset, int64_t range_length,
+        uint32_t block_size, /* download block size */
+        sp<DataSource> *source, /* to return and reuse source */
+        String8 *actualUrl) {
+    off64_t size;
+    sp<DataSource> temp_source;
+    if (source == NULL) {
+        source = &temp_source;
     }
 
-    off64_t size;
-    status_t err = source->getSize(&size);
+    if (*source == NULL) {
+        if (!strncasecmp(url, "file://", 7)) {
+            *source = new FileSource(url + 7);
+        } else if (strncasecmp(url, "http://", 7)
+                && strncasecmp(url, "https://", 8)) {
+            return ERROR_UNSUPPORTED;
+        } else {
+            KeyedVector<String8, String8> headers = mExtraHeaders;
+            if (range_offset > 0 || range_length >= 0) {
+                headers.add(
+                        String8("Range"),
+                        String8(
+                            StringPrintf(
+                                "bytes=%lld-%s",
+                                range_offset,
+                                range_length < 0
+                                    ? "" : StringPrintf("%lld",
+                                            range_offset + range_length - 1).c_str()).c_str()));
+            }
+            status_t err = mHTTPDataSource->connect(url, &headers);
 
-    if (err != OK) {
+            if (err != OK) {
+                return err;
+            }
+
+            *source = mHTTPDataSource;
+        }
+    }
+
+    status_t getSizeErr = (*source)->getSize(&size);
+    if (getSizeErr != OK) {
         size = 65536;
     }
 
-    sp<ABuffer> buffer = new ABuffer(size);
-    buffer->setRange(0, 0);
+    sp<ABuffer> buffer = *out != NULL ? *out : new ABuffer(size);
+    if (*out == NULL) {
+        buffer->setRange(0, 0);
+    }
 
+    // adjust range_length if only reading partial block
+    if (block_size > 0 && (range_length == -1 || buffer->size() + block_size < range_length)) {
+        range_length = buffer->size() + block_size;
+    }
     for (;;) {
+        // Only resize when we don't know the size.
         size_t bufferRemaining = buffer->capacity() - buffer->size();
-
-        if (bufferRemaining == 0) {
+        if (bufferRemaining == 0 && getSizeErr != OK) {
             bufferRemaining = 32768;
 
             ALOGV("increasing download buffer to %d bytes",
@@ -583,7 +698,9 @@
             }
         }
 
-        ssize_t n = source->readAt(
+        // The DataSource is responsible for informing us of error (n < 0) or eof (n == 0)
+        // to help us break out of the loop.
+        ssize_t n = (*source)->readAt(
                 buffer->size(), buffer->data() + buffer->size(),
                 maxBytesToRead);
 
@@ -599,6 +716,12 @@
     }
 
     *out = buffer;
+    if (actualUrl != NULL) {
+        *actualUrl = (*source)->getUri();
+        if (actualUrl->isEmpty()) {
+            *actualUrl = url;
+        }
+    }
 
     return OK;
 }
@@ -610,7 +733,8 @@
     *unchanged = false;
 
     sp<ABuffer> buffer;
-    status_t err = fetchFile(url, &buffer);
+    String8 actualUrl;
+    status_t err = fetchFile(url, &buffer, 0, -1, 0, NULL, &actualUrl);
 
     if (err != OK) {
         return NULL;
@@ -632,9 +756,6 @@
         // playlist unchanged
         *unchanged = true;
 
-        ALOGV("Playlist unchanged, refresh state is now %d",
-             (int)mRefreshState);
-
         return NULL;
     }
 
@@ -644,7 +765,7 @@
 #endif
 
     sp<M3UParser> playlist =
-        new M3UParser(url, buffer->data(), buffer->size());
+        new M3UParser(actualUrl.string(), buffer->data(), buffer->size());
 
     if (playlist->initCheck() != OK) {
         ALOGE("failed to parse .m3u8 playlist");
@@ -810,8 +931,25 @@
     return err;
 }
 
+bool LiveSession::canSwitchUp() {
+    // Allow upwards bandwidth switch when a stream has buffered at least 10 seconds.
+    status_t err = OK;
+    for (size_t i = 0; i < mPacketSources.size(); ++i) {
+        sp<AnotherPacketSource> source = mPacketSources.valueAt(i);
+        int64_t dur = source->getBufferedDurationUs(&err);
+        if (err == OK && dur > 10000000) {
+            return true;
+        }
+    }
+    return false;
+}
+
 void LiveSession::changeConfiguration(
         int64_t timeUs, size_t bandwidthIndex, bool pickTrack) {
+    // Protect mPacketSources from a swapPacketSource race condition through reconfiguration.
+    // (changeConfiguration, onChangeConfiguration2, onChangeConfiguration3).
+    cancelBandwidthSwitch();
+
     CHECK(!mReconfigurationInProgress);
     mReconfigurationInProgress = true;
 
@@ -827,21 +965,14 @@
     CHECK_LT(bandwidthIndex, mBandwidthItems.size());
     const BandwidthItem &item = mBandwidthItems.itemAt(bandwidthIndex);
 
-    uint32_t streamMask = 0;
+    uint32_t streamMask = 0; // streams that should be fetched by the new fetcher
+    uint32_t resumeMask = 0; // streams that should be fetched by the original fetcher
 
-    AString audioURI;
-    if (mPlaylist->getAudioURI(item.mPlaylistIndex, &audioURI)) {
-        streamMask |= STREAMTYPE_AUDIO;
-    }
-
-    AString videoURI;
-    if (mPlaylist->getVideoURI(item.mPlaylistIndex, &videoURI)) {
-        streamMask |= STREAMTYPE_VIDEO;
-    }
-
-    AString subtitleURI;
-    if (mPlaylist->getSubtitleURI(item.mPlaylistIndex, &subtitleURI)) {
-        streamMask |= STREAMTYPE_SUBTITLES;
+    AString URIs[kMaxStreams];
+    for (size_t i = 0; i < kMaxStreams; ++i) {
+        if (mPlaylist->getTypeURI(item.mPlaylistIndex, mStreams[i].mType, &URIs[i])) {
+            streamMask |= indexToType(i);
+        }
     }
 
     // Step 1, stop and discard fetchers that are no longer needed.
@@ -853,10 +984,15 @@
 
         // If we're seeking all current fetchers are discarded.
         if (timeUs < 0ll) {
-            if (((streamMask & STREAMTYPE_AUDIO) && uri == audioURI)
-                    || ((streamMask & STREAMTYPE_VIDEO) && uri == videoURI)
-                    || ((streamMask & STREAMTYPE_SUBTITLES) && uri == subtitleURI)) {
-                discardFetcher = false;
+            // delay fetcher removal
+            discardFetcher = false;
+
+            for (size_t j = 0; j < kMaxStreams; ++j) {
+                StreamType type = indexToType(j);
+                if ((streamMask & type) && uri == URIs[j]) {
+                    resumeMask |= type;
+                    streamMask &= ~type;
+                }
             }
         }
 
@@ -867,17 +1003,20 @@
         }
     }
 
-    sp<AMessage> msg = new AMessage(kWhatChangeConfiguration2, id());
+    sp<AMessage> msg;
+    if (timeUs < 0ll) {
+        // skip onChangeConfiguration2 (decoder destruction) if switching.
+        msg = new AMessage(kWhatChangeConfiguration3, id());
+    } else {
+        msg = new AMessage(kWhatChangeConfiguration2, id());
+    }
     msg->setInt32("streamMask", streamMask);
+    msg->setInt32("resumeMask", resumeMask);
     msg->setInt64("timeUs", timeUs);
-    if (streamMask & STREAMTYPE_AUDIO) {
-        msg->setString("audioURI", audioURI.c_str());
-    }
-    if (streamMask & STREAMTYPE_VIDEO) {
-        msg->setString("videoURI", videoURI.c_str());
-    }
-    if (streamMask & STREAMTYPE_SUBTITLES) {
-        msg->setString("subtitleURI", subtitleURI.c_str());
+    for (size_t i = 0; i < kMaxStreams; ++i) {
+        if (streamMask & indexToType(i)) {
+            msg->setString(mStreams[i].uriKey().c_str(), URIs[i].c_str());
+        }
     }
 
     // Every time a fetcher acknowledges the stopAsync or pauseAsync request
@@ -908,18 +1047,13 @@
     uint32_t streamMask;
     CHECK(msg->findInt32("streamMask", (int32_t *)&streamMask));
 
-    AString audioURI, videoURI, subtitleURI;
-    if (streamMask & STREAMTYPE_AUDIO) {
-        CHECK(msg->findString("audioURI", &audioURI));
-        ALOGV("audioURI = '%s'", audioURI.c_str());
-    }
-    if (streamMask & STREAMTYPE_VIDEO) {
-        CHECK(msg->findString("videoURI", &videoURI));
-        ALOGV("videoURI = '%s'", videoURI.c_str());
-    }
-    if (streamMask & STREAMTYPE_SUBTITLES) {
-        CHECK(msg->findString("subtitleURI", &subtitleURI));
-        ALOGV("subtitleURI = '%s'", subtitleURI.c_str());
+    AString URIs[kMaxStreams];
+    for (size_t i = 0; i < kMaxStreams; ++i) {
+        if (streamMask & indexToType(i)) {
+            const AString &uriKey = mStreams[i].uriKey();
+            CHECK(msg->findString(uriKey.c_str(), &URIs[i]));
+            ALOGV("%s = '%s'", uriKey.c_str(), URIs[i].c_str());
+        }
     }
 
     // Determine which decoders to shutdown on the player side,
@@ -929,15 +1063,12 @@
     // 2) its streamtype was already active and still is but the URI
     //    has changed.
     uint32_t changedMask = 0;
-    if (((mStreamMask & streamMask & STREAMTYPE_AUDIO)
-                && !(audioURI == mAudioURI))
-        || (mStreamMask & ~streamMask & STREAMTYPE_AUDIO)) {
-        changedMask |= STREAMTYPE_AUDIO;
-    }
-    if (((mStreamMask & streamMask & STREAMTYPE_VIDEO)
-                && !(videoURI == mVideoURI))
-        || (mStreamMask & ~streamMask & STREAMTYPE_VIDEO)) {
-        changedMask |= STREAMTYPE_VIDEO;
+    for (size_t i = 0; i < kMaxStreams && i != kSubtitleIndex; ++i) {
+        if (((mStreamMask & streamMask & indexToType(i))
+                && !(URIs[i] == mStreams[i].mUri))
+                || (mStreamMask & ~streamMask & indexToType(i))) {
+            changedMask |= indexToType(i);
+        }
     }
 
     if (changedMask == 0) {
@@ -963,68 +1094,54 @@
 }
 
 void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) {
+    mContinuation.clear();
     // All remaining fetchers are still suspended, the player has shutdown
     // any decoders that needed it.
 
-    uint32_t streamMask;
+    uint32_t streamMask, resumeMask;
     CHECK(msg->findInt32("streamMask", (int32_t *)&streamMask));
+    CHECK(msg->findInt32("resumeMask", (int32_t *)&resumeMask));
 
-    AString audioURI, videoURI, subtitleURI;
-    if (streamMask & STREAMTYPE_AUDIO) {
-        CHECK(msg->findString("audioURI", &audioURI));
-    }
-    if (streamMask & STREAMTYPE_VIDEO) {
-        CHECK(msg->findString("videoURI", &videoURI));
-    }
-    if (streamMask & STREAMTYPE_SUBTITLES) {
-        CHECK(msg->findString("subtitleURI", &subtitleURI));
+    for (size_t i = 0; i < kMaxStreams; ++i) {
+        if (streamMask & indexToType(i)) {
+            CHECK(msg->findString(mStreams[i].uriKey().c_str(), &mStreams[i].mUri));
+        }
     }
 
     int64_t timeUs;
+    bool switching = false;
     CHECK(msg->findInt64("timeUs", &timeUs));
 
     if (timeUs < 0ll) {
         timeUs = mLastDequeuedTimeUs;
+        switching = true;
     }
     mRealTimeBaseUs = ALooper::GetNowUs() - timeUs;
 
-    mStreamMask = streamMask;
-    mAudioURI = audioURI;
-    mVideoURI = videoURI;
-    mSubtitleURI = subtitleURI;
+    mNewStreamMask = streamMask;
 
-    // Resume all existing fetchers and assign them packet sources.
+    // Of all existing fetchers:
+    // * Resume fetchers that are still needed and assign them original packet sources.
+    // * Mark otherwise unneeded fetchers for removal.
+    ALOGV("resuming fetchers for mask 0x%08x", resumeMask);
     for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
         const AString &uri = mFetcherInfos.keyAt(i);
 
-        uint32_t resumeMask = 0;
-
-        sp<AnotherPacketSource> audioSource;
-        if ((streamMask & STREAMTYPE_AUDIO) && uri == audioURI) {
-            audioSource = mPacketSources.valueFor(STREAMTYPE_AUDIO);
-            resumeMask |= STREAMTYPE_AUDIO;
+        sp<AnotherPacketSource> sources[kMaxStreams];
+        for (size_t j = 0; j < kMaxStreams; ++j) {
+            if ((resumeMask & indexToType(j)) && uri == mStreams[j].mUri) {
+                sources[j] = mPacketSources.valueFor(indexToType(j));
+            }
         }
 
-        sp<AnotherPacketSource> videoSource;
-        if ((streamMask & STREAMTYPE_VIDEO) && uri == videoURI) {
-            videoSource = mPacketSources.valueFor(STREAMTYPE_VIDEO);
-            resumeMask |= STREAMTYPE_VIDEO;
+        FetcherInfo &info = mFetcherInfos.editValueAt(i);
+        if (sources[kAudioIndex] != NULL || sources[kVideoIndex] != NULL
+                || sources[kSubtitleIndex] != NULL) {
+            info.mFetcher->startAsync(
+                    sources[kAudioIndex], sources[kVideoIndex], sources[kSubtitleIndex]);
+        } else {
+            info.mToBeRemoved = true;
         }
-
-        sp<AnotherPacketSource> subtitleSource;
-        if ((streamMask & STREAMTYPE_SUBTITLES) && uri == subtitleURI) {
-            subtitleSource = mPacketSources.valueFor(STREAMTYPE_SUBTITLES);
-            resumeMask |= STREAMTYPE_SUBTITLES;
-        }
-
-        CHECK_NE(resumeMask, 0u);
-
-        ALOGV("resuming fetchers for mask 0x%08x", resumeMask);
-
-        streamMask &= ~resumeMask;
-
-        mFetcherInfos.valueAt(i).mFetcher->startAsync(
-                audioSource, videoSource, subtitleSource);
     }
 
     // streamMask now only contains the types that need a new fetcher created.
@@ -1033,52 +1150,65 @@
         ALOGV("creating new fetchers for mask 0x%08x", streamMask);
     }
 
-    while (streamMask != 0) {
-        StreamType streamType = (StreamType)(streamMask & ~(streamMask - 1));
+    // Find out when the original fetchers have buffered up to and start the new fetchers
+    // at a later timestamp.
+    for (size_t i = 0; i < kMaxStreams; i++) {
+        if (!(indexToType(i) & streamMask)) {
+            continue;
+        }
 
         AString uri;
-        switch (streamType) {
-            case STREAMTYPE_AUDIO:
-                uri = audioURI;
-                break;
-            case STREAMTYPE_VIDEO:
-                uri = videoURI;
-                break;
-            case STREAMTYPE_SUBTITLES:
-                uri = subtitleURI;
-                break;
-            default:
-                TRESPASS();
-        }
+        uri = mStreams[i].mUri;
 
         sp<PlaylistFetcher> fetcher = addFetcher(uri.c_str());
         CHECK(fetcher != NULL);
 
-        sp<AnotherPacketSource> audioSource;
-        if ((streamMask & STREAMTYPE_AUDIO) && uri == audioURI) {
-            audioSource = mPacketSources.valueFor(STREAMTYPE_AUDIO);
-            audioSource->clear();
+        int32_t latestSeq = -1;
+        int64_t latestTimeUs = 0ll;
+        sp<AnotherPacketSource> sources[kMaxStreams];
 
-            streamMask &= ~STREAMTYPE_AUDIO;
+        // TRICKY: looping from i as earlier streams are already removed from streamMask
+        for (size_t j = i; j < kMaxStreams; ++j) {
+            if ((streamMask & indexToType(j)) && uri == mStreams[j].mUri) {
+                sources[j] = mPacketSources.valueFor(indexToType(j));
+
+                if (!switching) {
+                    sources[j]->clear();
+                } else {
+                    int32_t type, seq;
+                    int64_t srcTimeUs;
+                    sp<AMessage> meta = sources[j]->getLatestMeta();
+
+                    if (meta != NULL && !meta->findInt32("discontinuity", &type)) {
+                        CHECK(meta->findInt32("seq", &seq));
+                        if (seq > latestSeq) {
+                            latestSeq = seq;
+                        }
+                        CHECK(meta->findInt64("timeUs", &srcTimeUs));
+                        if (srcTimeUs > latestTimeUs) {
+                            latestTimeUs = srcTimeUs;
+                        }
+                    }
+
+                    sources[j] = mPacketSources2.valueFor(indexToType(j));
+                    sources[j]->clear();
+                    uint32_t extraStreams = mNewStreamMask & (~mStreamMask);
+                    if (extraStreams & indexToType(j)) {
+                        sources[j]->queueAccessUnit(createFormatChangeBuffer(/* swap = */ false));
+                    }
+                }
+
+                streamMask &= ~indexToType(j);
+            }
         }
 
-        sp<AnotherPacketSource> videoSource;
-        if ((streamMask & STREAMTYPE_VIDEO) && uri == videoURI) {
-            videoSource = mPacketSources.valueFor(STREAMTYPE_VIDEO);
-            videoSource->clear();
-
-            streamMask &= ~STREAMTYPE_VIDEO;
-        }
-
-        sp<AnotherPacketSource> subtitleSource;
-        if ((streamMask & STREAMTYPE_SUBTITLES) && uri == subtitleURI) {
-            subtitleSource = mPacketSources.valueFor(STREAMTYPE_SUBTITLES);
-            subtitleSource->clear();
-
-            streamMask &= ~STREAMTYPE_SUBTITLES;
-        }
-
-        fetcher->startAsync(audioSource, videoSource, subtitleSource, timeUs);
+        fetcher->startAsync(
+                sources[kAudioIndex],
+                sources[kVideoIndex],
+                sources[kSubtitleIndex],
+                timeUs,
+                latestTimeUs /* min start time(us) */,
+                latestSeq >= 0 ? latestSeq + 1 : -1 /* starting sequence number hint */ );
     }
 
     // All fetchers have now been started, the configuration change
@@ -1087,14 +1217,61 @@
     scheduleCheckBandwidthEvent();
 
     ALOGV("XXX configuration change completed.");
-
     mReconfigurationInProgress = false;
+    if (switching) {
+        mSwitchInProgress = true;
+    } else {
+        mStreamMask = mNewStreamMask;
+    }
 
     if (mDisconnectReplyID != 0) {
         finishDisconnect();
     }
 }
 
+void LiveSession::onSwapped(const sp<AMessage> &msg) {
+    int32_t switchGeneration;
+    CHECK(msg->findInt32("switchGeneration", &switchGeneration));
+    if (switchGeneration != mSwitchGeneration) {
+        return;
+    }
+
+    int32_t stream;
+    CHECK(msg->findInt32("stream", &stream));
+    mSwapMask |= stream;
+    if (mSwapMask != mStreamMask) {
+        return;
+    }
+
+    // Check if new variant contains extra streams.
+    uint32_t extraStreams = mNewStreamMask & (~mStreamMask);
+    while (extraStreams) {
+        StreamType extraStream = (StreamType) (extraStreams & ~(extraStreams - 1));
+        swapPacketSource(extraStream);
+        extraStreams &= ~extraStream;
+    }
+
+    tryToFinishBandwidthSwitch();
+}
+
+// Mark switch done when:
+//   1. all old buffers are swapped out, AND
+//   2. all old fetchers are removed.
+void LiveSession::tryToFinishBandwidthSwitch() {
+    bool needToRemoveFetchers = false;
+    for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
+        if (mFetcherInfos.valueAt(i).mToBeRemoved) {
+            needToRemoveFetchers = true;
+            break;
+        }
+    }
+    if (!needToRemoveFetchers && mSwapMask == mStreamMask) {
+        mStreamMask = mNewStreamMask;
+        mSwitchInProgress = false;
+        mSwapMask = 0;
+    }
+}
+
 void LiveSession::scheduleCheckBandwidthEvent() {
     sp<AMessage> msg = new AMessage(kWhatCheckBandwidth, id());
     msg->setInt32("generation", mCheckBandwidthGeneration);
@@ -1105,16 +1282,37 @@
     ++mCheckBandwidthGeneration;
 }
 
-void LiveSession::onCheckBandwidth() {
-    if (mReconfigurationInProgress) {
-        scheduleCheckBandwidthEvent();
-        return;
+void LiveSession::cancelBandwidthSwitch() {
+    Mutex::Autolock lock(mSwapMutex);
+    mSwitchGeneration++;
+    mSwitchInProgress = false;
+    mSwapMask = 0;
+}
+
+bool LiveSession::canSwitchBandwidthTo(size_t bandwidthIndex) {
+    if (mReconfigurationInProgress || mSwitchInProgress) {
+        return false;
     }
 
+    if (mPrevBandwidthIndex < 0) {
+        return true;
+    }
+
+    if (bandwidthIndex == (size_t)mPrevBandwidthIndex) {
+        return false;
+    } else if (bandwidthIndex > (size_t)mPrevBandwidthIndex) {
+        return canSwitchUp();
+    } else {
+        return true;
+    }
+}
+
+void LiveSession::onCheckBandwidth() {
     size_t bandwidthIndex = getBandwidthIndex();
-    if (mPrevBandwidthIndex < 0
-            || bandwidthIndex != (size_t)mPrevBandwidthIndex) {
+    if (canSwitchBandwidthTo(bandwidthIndex)) {
         changeConfiguration(-1ll /* timeUs */, bandwidthIndex);
+    } else {
+        scheduleCheckBandwidthEvent();
     }
 
     // Handling the kWhatCheckBandwidth even here does _not_ automatically

media/libstagefright/httplive/LiveSession.h

diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h
index 99b480a8..91bbcea 100644
--- a/media/libstagefright/httplive/LiveSession.h
+++ b/media/libstagefright/httplive/LiveSession.h
@@ -28,6 +28,7 @@
 struct AnotherPacketSource;
 struct DataSource;
 struct HTTPBase;
+struct IMediaHTTPService;
 struct LiveDataSource;
 struct M3UParser;
 struct PlaylistFetcher;
@@ -40,12 +41,20 @@
     };
     LiveSession(
             const sp<AMessage> &notify,
-            uint32_t flags = 0, bool uidValid = false, uid_t uid = 0);
+            uint32_t flags,
+            const sp<IMediaHTTPService> &httpService);
+
+    enum StreamIndex {
+        kAudioIndex    = 0,
+        kVideoIndex    = 1,
+        kSubtitleIndex = 2,
+        kMaxStreams    = 3,
+    };
 
     enum StreamType {
-        STREAMTYPE_AUDIO        = 1,
-        STREAMTYPE_VIDEO        = 2,
-        STREAMTYPE_SUBTITLES    = 4,
+        STREAMTYPE_AUDIO        = 1 << kAudioIndex,
+        STREAMTYPE_VIDEO        = 1 << kVideoIndex,
+        STREAMTYPE_SUBTITLES    = 1 << kSubtitleIndex,
     };
     status_t dequeueAccessUnit(StreamType stream, sp<ABuffer> *accessUnit);
 
@@ -74,6 +83,11 @@
         kWhatPreparationFailed,
     };
 
+    // create a format-change discontinuity
+    //
+    // swap:
+    //   whether is format-change discontinuity should trigger a buffer swap
+    sp<ABuffer> createFormatChangeBuffer(bool swap = true);
 protected:
     virtual ~LiveSession();
 
@@ -92,6 +106,7 @@
         kWhatChangeConfiguration2       = 'chC2',
         kWhatChangeConfiguration3       = 'chC3',
         kWhatFinishDisconnect2          = 'fin2',
+        kWhatSwapped                    = 'swap',
     };
 
     struct BandwidthItem {
@@ -103,12 +118,25 @@
         sp<PlaylistFetcher> mFetcher;
         int64_t mDurationUs;
         bool mIsPrepared;
+        bool mToBeRemoved;
     };
 
+    struct StreamItem {
+        const char *mType;
+        AString mUri;
+        StreamItem() : mType("") {}
+        StreamItem(const char *type) : mType(type) {}
+        AString uriKey() {
+            AString key(mType);
+            key.append("URI");
+            return key;
+        }
+    };
+    StreamItem mStreams[kMaxStreams];
+
     sp<AMessage> mNotify;
     uint32_t mFlags;
-    bool mUIDValid;
-    uid_t mUID;
+    sp<IMediaHTTPService> mHTTPService;
 
     bool mInPreparationPhase;
 
@@ -123,12 +151,28 @@
     sp<M3UParser> mPlaylist;
 
     KeyedVector<AString, FetcherInfo> mFetcherInfos;
-    AString mAudioURI, mVideoURI, mSubtitleURI;
     uint32_t mStreamMask;
 
+    // Masks used during reconfiguration:
+    // mNewStreamMask: streams in the variant playlist we're switching to;
+    // we don't want to immediately overwrite the original value.
+    uint32_t mNewStreamMask;
+
+    // mSwapMask: streams that have started to playback content in the new variant playlist;
+    // we use this to track reconfiguration progress.
+    uint32_t mSwapMask;
+
     KeyedVector<StreamType, sp<AnotherPacketSource> > mPacketSources;
+    // A second set of packet sources that buffer content for the variant we're switching to.
+    KeyedVector<StreamType, sp<AnotherPacketSource> > mPacketSources2;
+
+    // A mutex used to serialize two sets of events:
+    // * the swapping of packet sources in dequeueAccessUnit on the player thread, AND
+    // * a forced bandwidth switch termination in cancelSwitch on the live looper.
+    Mutex mSwapMutex;
 
     int32_t mCheckBandwidthGeneration;
+    int32_t mSwitchGeneration;
 
     size_t mContinuationCounter;
     sp<AMessage> mContinuation;
@@ -137,6 +181,7 @@
     int64_t mRealTimeBaseUs;
 
     bool mReconfigurationInProgress;
+    bool mSwitchInProgress;
     uint32_t mDisconnectReplyID;
 
     sp<PlaylistFetcher> addFetcher(const char *uri);
@@ -145,9 +190,26 @@
     status_t onSeek(const sp<AMessage> &msg);
     void onFinishDisconnect2();
 
+    // If given a non-zero block_size (default 0), it is used to cap the number of
+    // bytes read in from the DataSource. If given a non-NULL buffer, new content
+    // is read into the end.
+    //
+    // The DataSource we read from is responsible for signaling error or EOF to help us
+    // break out of the read loop. The DataSource can be returned to the caller, so
+    // that the caller can reuse it for subsequent fetches (within the initially
+    // requested range).
+    //
+    // For reused HTTP sources, the caller must download a file sequentially without
+    // any overlaps or gaps to prevent reconnection.
     status_t fetchFile(
             const char *url, sp<ABuffer> *out,
-            int64_t range_offset = 0, int64_t range_length = -1);
+            /* request/open a file starting at range_offset for range_length bytes */
+            int64_t range_offset = 0, int64_t range_length = -1,
+            /* download block size */
+            uint32_t block_size = 0,
+            /* reuse DataSource if doing partial fetch */
+            sp<DataSource> *source = NULL,
+            String8 *actualUrl = NULL);
 
     sp<M3UParser> fetchPlaylist(
             const char *url, uint8_t *curPlaylistHash, bool *unchanged);
@@ -155,22 +217,34 @@
     size_t getBandwidthIndex();
 
     static int SortByBandwidth(const BandwidthItem *, const BandwidthItem *);
+    static StreamType indexToType(int idx);
 
     void changeConfiguration(
             int64_t timeUs, size_t bandwidthIndex, bool pickTrack = false);
     void onChangeConfiguration(const sp<AMessage> &msg);
     void onChangeConfiguration2(const sp<AMessage> &msg);
     void onChangeConfiguration3(const sp<AMessage> &msg);
+    void onSwapped(const sp<AMessage> &msg);
+    void tryToFinishBandwidthSwitch();
 
     void scheduleCheckBandwidthEvent();
     void cancelCheckBandwidthEvent();
 
+    // cancelBandwidthSwitch is atomic wrt swapPacketSource; call it to prevent packet sources
+    // from being swapped out on stale discontinuities while manipulating
+    // mPacketSources/mPacketSources2.
+    void cancelBandwidthSwitch();
+
+    bool canSwitchBandwidthTo(size_t bandwidthIndex);
     void onCheckBandwidth();
 
     void finishDisconnect();
 
     void postPrepared(status_t err);
 
+    void swapPacketSource(StreamType stream);
+    bool canSwitchUp();
+
     DISALLOW_EVIL_CONSTRUCTORS(LiveSession);
 };
 

media/libstagefright/httplive/M3UParser.cpp

diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp
index 243888c..587a6d5 100644
--- a/media/libstagefright/httplive/M3UParser.cpp
+++ b/media/libstagefright/httplive/M3UParser.cpp
@@ -24,6 +24,7 @@
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AMessage.h>
 #include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/Utils.h>
 #include <media/mediaplayer.h>
 
 namespace android {
@@ -125,7 +126,7 @@
                 mSelectedIndex = strtoul(value, &end, 10);
                 CHECK(end > value && *end == '\0');
 
-                if (mSelectedIndex >= mMediaItems.size()) {
+                if (mSelectedIndex >= (ssize_t)mMediaItems.size()) {
                     mSelectedIndex = mMediaItems.size() - 1;
                 }
             } else {
@@ -165,14 +166,14 @@
             ALOGE("track %d does not exist", index);
             return INVALID_OPERATION;
         }
-        if (mSelectedIndex == index) {
+        if (mSelectedIndex == (ssize_t)index) {
             ALOGE("track %d already selected", index);
             return BAD_VALUE;
         }
         ALOGV("selected track %d", index);
         mSelectedIndex = index;
     } else {
-        if (mSelectedIndex != index) {
+        if (mSelectedIndex != (ssize_t)index) {
             ALOGE("track %d is not selected", index);
             return BAD_VALUE;
         }
@@ -352,9 +353,27 @@
     if (!meta->findString(key, &groupID)) {
         *uri = mItems.itemAt(index).mURI;
 
-        // Assume media without any more specific attribute contains
-        // audio and video, but no subtitles.
-        return !strcmp("audio", key) || !strcmp("video", key);
+        AString codecs;
+        if (!meta->findString("codecs", &codecs)) {
+            // Assume media without any more specific attribute contains
+            // audio and video, but no subtitles.
+            return !strcmp("audio", key) || !strcmp("video", key);
+        } else {
+            // Split the comma separated list of codecs.
+            size_t offset = 0;
+            ssize_t commaPos = -1;
+            codecs.append(',');
+            while ((commaPos = codecs.find(",", offset)) >= 0) {
+                AString codec(codecs, offset, commaPos - offset);
+                // return true only if a codec of type `key` ("audio"/"video")
+                // is found.
+                if (codecIsType(codec, key)) {
+                    return true;
+                }
+                offset = commaPos + 1;
+            }
+            return false;
+        }
     }
 
     sp<MediaGroup> group = mMediaGroups.valueFor(groupID);
@@ -369,18 +388,6 @@
     return true;
 }
 
-bool M3UParser::getAudioURI(size_t index, AString *uri) const {
-    return getTypeURI(index, "audio", uri);
-}
-
-bool M3UParser::getVideoURI(size_t index, AString *uri) const {
-    return getTypeURI(index, "video", uri);
-}
-
-bool M3UParser::getSubtitleURI(size_t index, AString *uri) const {
-    return getTypeURI(index, "subtitles", uri);
-}
-
 static bool MakeURL(const char *baseURL, const char *url, AString *out) {
     out->clear();
 
@@ -416,22 +423,32 @@
     } else {
         // URL is a relative path
 
-        size_t n = strlen(baseURL);
-        if (baseURL[n - 1] == '/') {
-            out->setTo(baseURL);
-            out->append(url);
+        // Check for a possible query string
+        const char *qsPos = strchr(baseURL, '?');
+        size_t end;
+        if (qsPos != NULL) {
+            end = qsPos - baseURL;
         } else {
-            const char *slashPos = strrchr(baseURL, '/');
-
-            if (slashPos > &baseURL[6]) {
-                out->setTo(baseURL, slashPos - baseURL);
-            } else {
-                out->setTo(baseURL);
-            }
-
-            out->append("/");
-            out->append(url);
+            end = strlen(baseURL);
         }
+        // Check for the last slash before a potential query string
+        for (ssize_t pos = end - 1; pos >= 0; pos--) {
+            if (baseURL[pos] == '/') {
+                end = pos;
+                break;
+            }
+        }
+
+        // Check whether the found slash actually is part of the path
+        // and not part of the "http://".
+        if (end > 6) {
+            out->setTo(baseURL, end);
+        } else {
+            out->setTo(baseURL);
+        }
+
+        out->append("/");
+        out->append(url);
     }
 
     ALOGV("base:'%s', url:'%s' => '%s'", baseURL, url, out->c_str());
@@ -608,7 +625,7 @@
     if (meta->get() == NULL) {
         *meta = new AMessage;
     }
-    (*meta)->setInt64(key, (int64_t)x * 1E6);
+    (*meta)->setInt64(key, (int64_t)(x * 1E6));
 
     return OK;
 }
@@ -684,12 +701,22 @@
                 *meta = new AMessage;
             }
             (*meta)->setInt32("bandwidth", x);
+        } else if (!strcasecmp("codecs", key.c_str())) {
+            if (!isQuotedString(val)) {
+                ALOGE("Expected quoted string for %s attribute, "
+                      "got '%s' instead.",
+                      key.c_str(), val.c_str());;
+
+                return ERROR_MALFORMED;
+            }
+
+            key.tolower();
+            const AString &codecs = unquoteString(val);
+            (*meta)->setString(key.c_str(), codecs.c_str());
         } else if (!strcasecmp("audio", key.c_str())
                 || !strcasecmp("video", key.c_str())
                 || !strcasecmp("subtitles", key.c_str())) {
-            if (val.size() < 2
-                    || val.c_str()[0] != '"'
-                    || val.c_str()[val.size() - 1] != '"') {
+            if (!isQuotedString(val)) {
                 ALOGE("Expected quoted string for %s attribute, "
                       "got '%s' instead.",
                       key.c_str(), val.c_str());
@@ -697,7 +724,7 @@
                 return ERROR_MALFORMED;
             }
 
-            AString groupID(val, 1, val.size() - 2);
+            const AString &groupID = unquoteString(val);
             ssize_t groupIndex = mMediaGroups.indexOfKey(groupID);
 
             if (groupIndex < 0) {
@@ -1086,4 +1113,121 @@
     return OK;
 }
 
+// static
+bool M3UParser::isQuotedString(const AString &str) {
+    if (str.size() < 2
+            || str.c_str()[0] != '"'
+            || str.c_str()[str.size() - 1] != '"') {
+        return false;
+    }
+    return true;
+}
+
+// static
+AString M3UParser::unquoteString(const AString &str) {
+     if (!isQuotedString(str)) {
+         return str;
+     }
+     return AString(str, 1, str.size() - 2);
+}
+
+// static
+bool M3UParser::codecIsType(const AString &codec, const char *type) {
+    if (codec.size() < 4) {
+        return false;
+    }
+    const char *c = codec.c_str();
+    switch (FOURCC(c[0], c[1], c[2], c[3])) {
+        // List extracted from http://www.mp4ra.org/codecs.html
+        case 'ac-3':
+        case 'alac':
+        case 'dra1':
+        case 'dtsc':
+        case 'dtse':
+        case 'dtsh':
+        case 'dtsl':
+        case 'ec-3':
+        case 'enca':
+        case 'g719':
+        case 'g726':
+        case 'm4ae':
+        case 'mlpa':
+        case 'mp4a':
+        case 'raw ':
+        case 'samr':
+        case 'sawb':
+        case 'sawp':
+        case 'sevc':
+        case 'sqcp':
+        case 'ssmv':
+        case 'twos':
+        case 'agsm':
+        case 'alaw':
+        case 'dvi ':
+        case 'fl32':
+        case 'fl64':
+        case 'ima4':
+        case 'in24':
+        case 'in32':
+        case 'lpcm':
+        case 'Qclp':
+        case 'QDM2':
+        case 'QDMC':
+        case 'ulaw':
+        case 'vdva':
+            return !strcmp("audio", type);
+
+        case 'avc1':
+        case 'avc2':
+        case 'avcp':
+        case 'drac':
+        case 'encv':
+        case 'mjp2':
+        case 'mp4v':
+        case 'mvc1':
+        case 'mvc2':
+        case 'resv':
+        case 's263':
+        case 'svc1':
+        case 'vc-1':
+        case 'CFHD':
+        case 'civd':
+        case 'DV10':
+        case 'dvh5':
+        case 'dvh6':
+        case 'dvhp':
+        case 'DVOO':
+        case 'DVOR':
+        case 'DVTV':
+        case 'DVVT':
+        case 'flic':
+        case 'gif ':
+        case 'h261':
+        case 'h263':
+        case 'HD10':
+        case 'jpeg':
+        case 'M105':
+        case 'mjpa':
+        case 'mjpb':
+        case 'png ':
+        case 'PNTG':
+        case 'rle ':
+        case 'rpza':
+        case 'Shr0':
+        case 'Shr1':
+        case 'Shr2':
+        case 'Shr3':
+        case 'Shr4':
+        case 'SVQ1':
+        case 'SVQ3':
+        case 'tga ':
+        case 'tiff':
+        case 'WRLE':
+            return !strcmp("video", type);
+
+        default:
+            return false;
+    }
+}
+
 }  // namespace android

media/libstagefright/httplive/M3UParser.h

diff --git a/media/libstagefright/httplive/M3UParser.h b/media/libstagefright/httplive/M3UParser.h
index 5248004..ccd6556 100644
--- a/media/libstagefright/httplive/M3UParser.h
+++ b/media/libstagefright/httplive/M3UParser.h
@@ -45,9 +45,7 @@
     status_t getTrackInfo(Parcel* reply) const;
     ssize_t getSelectedIndex() const;
 
-    bool getAudioURI(size_t index, AString *uri) const;
-    bool getVideoURI(size_t index, AString *uri) const;
-    bool getSubtitleURI(size_t index, AString *uri) const;
+    bool getTypeURI(size_t index, const char *key, AString *uri) const;
 
 protected:
     virtual ~M3UParser();
@@ -95,11 +93,13 @@
 
     status_t parseMedia(const AString &line);
 
-    bool getTypeURI(size_t index, const char *key, AString *uri) const;
-
     static status_t ParseInt32(const char *s, int32_t *x);
     static status_t ParseDouble(const char *s, double *x);
 
+    static bool isQuotedString(const AString &str);
+    static AString unquoteString(const AString &str);
+    static bool codecIsType(const AString &codec, const char *type);
+
     DISALLOW_EVIL_CONSTRUCTORS(M3UParser);
 };
 

media/libstagefright/httplive/PlaylistFetcher.cpp

diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index 973b779..2649705 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -47,26 +47,34 @@
 
 // static
 const int64_t PlaylistFetcher::kMinBufferedDurationUs = 10000000ll;
+const int64_t PlaylistFetcher::kMaxMonitorDelayUs = 3000000ll;
+const int32_t PlaylistFetcher::kNumSkipFrames = 10;
 
 PlaylistFetcher::PlaylistFetcher(
         const sp<AMessage> &notify,
         const sp<LiveSession> &session,
         const char *uri)
     : mNotify(notify),
+      mStartTimeUsNotify(notify->dup()),
       mSession(session),
       mURI(uri),
       mStreamTypeMask(0),
       mStartTimeUs(-1ll),
+      mMinStartTimeUs(0ll),
+      mStopParams(NULL),
       mLastPlaylistFetchTimeUs(-1ll),
       mSeqNumber(-1),
       mNumRetries(0),
       mStartup(true),
+      mPrepared(false),
       mNextPTSTimeUs(-1ll),
       mMonitorQueueGeneration(0),
       mRefreshState(INITIAL_MINIMUM_RELOAD_DELAY),
       mFirstPTSValid(false),
       mAbsoluteTimeAnchorUs(0ll) {
     memset(mPlaylistHash, 0, sizeof(mPlaylistHash));
+    mStartTimeUsNotify->setInt32("what", kWhatStartedAt);
+    mStartTimeUsNotify->setInt32("streamMask", 0);
 }
 
 PlaylistFetcher::~PlaylistFetcher() {
@@ -103,10 +111,16 @@
     return segmentStartUs;
 }
 
-bool PlaylistFetcher::timeToRefreshPlaylist(int64_t nowUs) const {
-    if (mPlaylist == NULL) {
+int64_t PlaylistFetcher::delayUsToRefreshPlaylist() const {
+    int64_t nowUs = ALooper::GetNowUs();
+
+    if (mPlaylist == NULL || mLastPlaylistFetchTimeUs < 0ll) {
         CHECK_EQ((int)mRefreshState, (int)INITIAL_MINIMUM_RELOAD_DELAY);
-        return true;
+        return 0ll;
+    }
+
+    if (mPlaylist->isComplete()) {
+        return (~0llu >> 1);
     }
 
     int32_t targetDurationSecs;
@@ -157,11 +171,13 @@
             break;
     }
 
-    return mLastPlaylistFetchTimeUs + minPlaylistAgeUs <= nowUs;
+    int64_t delayUs = mLastPlaylistFetchTimeUs + minPlaylistAgeUs - nowUs;
+    return delayUs > 0ll ? delayUs : 0ll;
 }
 
 status_t PlaylistFetcher::decryptBuffer(
-        size_t playlistIndex, const sp<ABuffer> &buffer) {
+        size_t playlistIndex, const sp<ABuffer> &buffer,
+        bool first) {
     sp<AMessage> itemMeta;
     bool found = false;
     AString method;
@@ -179,6 +195,7 @@
     if (!found) {
         method = "NONE";
     }
+    buffer->meta()->setString("cipher-method", method.c_str());
 
     if (method == "NONE") {
         return OK;
@@ -218,63 +235,89 @@
         return UNKNOWN_ERROR;
     }
 
-    unsigned char aes_ivec[16];
+    size_t n = buffer->size();
+    if (!n) {
+        return OK;
+    }
+    CHECK(n % 16 == 0);
 
-    AString iv;
-    if (itemMeta->findString("cipher-iv", &iv)) {
-        if ((!iv.startsWith("0x") && !iv.startsWith("0X"))
-                || iv.size() != 16 * 2 + 2) {
-            ALOGE("malformed cipher IV '%s'.", iv.c_str());
-            return ERROR_MALFORMED;
-        }
+    if (first) {
+        // If decrypting the first block in a file, read the iv from the manifest
+        // or derive the iv from the file's sequence number.
 
-        memset(aes_ivec, 0, sizeof(aes_ivec));
-        for (size_t i = 0; i < 16; ++i) {
-            char c1 = tolower(iv.c_str()[2 + 2 * i]);
-            char c2 = tolower(iv.c_str()[3 + 2 * i]);
-            if (!isxdigit(c1) || !isxdigit(c2)) {
+        AString iv;
+        if (itemMeta->findString("cipher-iv", &iv)) {
+            if ((!iv.startsWith("0x") && !iv.startsWith("0X"))
+                    || iv.size() != 16 * 2 + 2) {
                 ALOGE("malformed cipher IV '%s'.", iv.c_str());
                 return ERROR_MALFORMED;
             }
-            uint8_t nibble1 = isdigit(c1) ? c1 - '0' : c1 - 'a' + 10;
-            uint8_t nibble2 = isdigit(c2) ? c2 - '0' : c2 - 'a' + 10;
 
-            aes_ivec[i] = nibble1 << 4 | nibble2;
+            memset(mAESInitVec, 0, sizeof(mAESInitVec));
+            for (size_t i = 0; i < 16; ++i) {
+                char c1 = tolower(iv.c_str()[2 + 2 * i]);
+                char c2 = tolower(iv.c_str()[3 + 2 * i]);
+                if (!isxdigit(c1) || !isxdigit(c2)) {
+                    ALOGE("malformed cipher IV '%s'.", iv.c_str());
+                    return ERROR_MALFORMED;
+                }
+                uint8_t nibble1 = isdigit(c1) ? c1 - '0' : c1 - 'a' + 10;
+                uint8_t nibble2 = isdigit(c2) ? c2 - '0' : c2 - 'a' + 10;
+
+                mAESInitVec[i] = nibble1 << 4 | nibble2;
+            }
+        } else {
+            memset(mAESInitVec, 0, sizeof(mAESInitVec));
+            mAESInitVec[15] = mSeqNumber & 0xff;
+            mAESInitVec[14] = (mSeqNumber >> 8) & 0xff;
+            mAESInitVec[13] = (mSeqNumber >> 16) & 0xff;
+            mAESInitVec[12] = (mSeqNumber >> 24) & 0xff;
         }
-    } else {
-        memset(aes_ivec, 0, sizeof(aes_ivec));
-        aes_ivec[15] = mSeqNumber & 0xff;
-        aes_ivec[14] = (mSeqNumber >> 8) & 0xff;
-        aes_ivec[13] = (mSeqNumber >> 16) & 0xff;
-        aes_ivec[12] = (mSeqNumber >> 24) & 0xff;
     }
 
     AES_cbc_encrypt(
             buffer->data(), buffer->data(), buffer->size(),
-            &aes_key, aes_ivec, AES_DECRYPT);
-
-    // hexdump(buffer->data(), buffer->size());
-
-    size_t n = buffer->size();
-    CHECK_GT(n, 0u);
-
-    size_t pad = buffer->data()[n - 1];
-
-    CHECK_GT(pad, 0u);
-    CHECK_LE(pad, 16u);
-    CHECK_GE((size_t)n, pad);
-    for (size_t i = 0; i < pad; ++i) {
-        CHECK_EQ((unsigned)buffer->data()[n - 1 - i], pad);
-    }
-
-    n -= pad;
-
-    buffer->setRange(buffer->offset(), n);
+            &aes_key, mAESInitVec, AES_DECRYPT);
 
     return OK;
 }
 
-void PlaylistFetcher::postMonitorQueue(int64_t delayUs) {
+status_t PlaylistFetcher::checkDecryptPadding(const sp<ABuffer> &buffer) {
+    status_t err;
+    AString method;
+    CHECK(buffer->meta()->findString("cipher-method", &method));
+    if (method == "NONE") {
+        return OK;
+    }
+
+    uint8_t padding = 0;
+    if (buffer->size() > 0) {
+        padding = buffer->data()[buffer->size() - 1];
+    }
+
+    if (padding > 16) {
+        return ERROR_MALFORMED;
+    }
+
+    for (size_t i = buffer->size() - padding; i < padding; i++) {
+        if (buffer->data()[i] != padding) {
+            return ERROR_MALFORMED;
+        }
+    }
+
+    buffer->setRange(buffer->offset(), buffer->size() - padding);
+    return OK;
+}
+
+void PlaylistFetcher::postMonitorQueue(int64_t delayUs, int64_t minDelayUs) {
+    int64_t maxDelayUs = delayUsToRefreshPlaylist();
+    if (maxDelayUs < minDelayUs) {
+        maxDelayUs = minDelayUs;
+    }
+    if (delayUs > maxDelayUs) {
+        ALOGV("Need to refresh playlist in %lld", maxDelayUs);
+        delayUs = maxDelayUs;
+    }
     sp<AMessage> msg = new AMessage(kWhatMonitorQueue, id());
     msg->setInt32("generation", mMonitorQueueGeneration);
     msg->post(delayUs);
@@ -288,7 +331,9 @@
         const sp<AnotherPacketSource> &audioSource,
         const sp<AnotherPacketSource> &videoSource,
         const sp<AnotherPacketSource> &subtitleSource,
-        int64_t startTimeUs) {
+        int64_t startTimeUs,
+        int64_t minStartTimeUs,
+        int32_t startSeqNumberHint) {
     sp<AMessage> msg = new AMessage(kWhatStart, id());
 
     uint32_t streamTypeMask = 0ul;
@@ -310,6 +355,8 @@
 
     msg->setInt32("streamTypeMask", streamTypeMask);
     msg->setInt64("startTimeUs", startTimeUs);
+    msg->setInt64("minStartTimeUs", minStartTimeUs);
+    msg->setInt32("startSeqNumberHint", startSeqNumberHint);
     msg->post();
 }
 
@@ -321,6 +368,12 @@
     (new AMessage(kWhatStop, id()))->post();
 }
 
+void PlaylistFetcher::resumeUntilAsync(const sp<AMessage> &params) {
+    AMessage* msg = new AMessage(kWhatResumeUntil, id());
+    msg->setMessage("params", params);
+    msg->post();
+}
+
 void PlaylistFetcher::onMessageReceived(const sp<AMessage> &msg) {
     switch (msg->what()) {
         case kWhatStart:
@@ -355,6 +408,7 @@
         }
 
         case kWhatMonitorQueue:
+        case kWhatDownloadNext:
         {
             int32_t generation;
             CHECK(msg->findInt32("generation", &generation));
@@ -364,7 +418,17 @@
                 break;
             }
 
-            onMonitorQueue();
+            if (msg->what() == kWhatMonitorQueue) {
+                onMonitorQueue();
+            } else {
+                onDownloadNext();
+            }
+            break;
+        }
+
+        case kWhatResumeUntil:
+        {
+            onResumeUntil(msg);
             break;
         }
 
@@ -380,7 +444,10 @@
     CHECK(msg->findInt32("streamTypeMask", (int32_t *)&streamTypeMask));
 
     int64_t startTimeUs;
+    int32_t startSeqNumberHint;
     CHECK(msg->findInt64("startTimeUs", &startTimeUs));
+    CHECK(msg->findInt64("minStartTimeUs", (int64_t *) &mMinStartTimeUs));
+    CHECK(msg->findInt32("startSeqNumberHint", &startSeqNumberHint));
 
     if (streamTypeMask & LiveSession::STREAMTYPE_AUDIO) {
         void *ptr;
@@ -415,6 +482,11 @@
     if (mStartTimeUs >= 0ll) {
         mSeqNumber = -1;
         mStartup = true;
+        mPrepared = false;
+    }
+
+    if (startSeqNumberHint >= 0) {
+        mSeqNumber = startSeqNumberHint;
     }
 
     postMonitorQueue();
@@ -424,22 +496,72 @@
 
 void PlaylistFetcher::onPause() {
     cancelMonitorQueue();
-
-    mPacketSources.clear();
-    mStreamTypeMask = 0;
 }
 
 void PlaylistFetcher::onStop() {
     cancelMonitorQueue();
 
-    for (size_t i = 0; i < mPacketSources.size(); ++i) {
-        mPacketSources.valueAt(i)->clear();
-    }
-
     mPacketSources.clear();
     mStreamTypeMask = 0;
 }
 
+// Resume until we have reached the boundary timestamps listed in `msg`; when
+// the remaining time is too short (within a resume threshold) stop immediately
+// instead.
+status_t PlaylistFetcher::onResumeUntil(const sp<AMessage> &msg) {
+    sp<AMessage> params;
+    CHECK(msg->findMessage("params", &params));
+
+    bool stop = false;
+    for (size_t i = 0; i < mPacketSources.size(); i++) {
+        sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i);
+
+        const char *stopKey;
+        int streamType = mPacketSources.keyAt(i);
+        switch (streamType) {
+        case LiveSession::STREAMTYPE_VIDEO:
+            stopKey = "timeUsVideo";
+            break;
+
+        case LiveSession::STREAMTYPE_AUDIO:
+            stopKey = "timeUsAudio";
+            break;
+
+        case LiveSession::STREAMTYPE_SUBTITLES:
+            stopKey = "timeUsSubtitle";
+            break;
+
+        default:
+            TRESPASS();
+        }
+
+        // Don't resume if we would stop within a resume threshold.
+        int64_t latestTimeUs = 0, stopTimeUs = 0;
+        sp<AMessage> latestMeta = packetSource->getLatestMeta();
+        if (latestMeta != NULL
+                && (latestMeta->findInt64("timeUs", &latestTimeUs)
+                && params->findInt64(stopKey, &stopTimeUs))) {
+            int64_t diffUs = stopTimeUs - latestTimeUs;
+            if (diffUs < resumeThreshold(latestMeta)) {
+                stop = true;
+            }
+        }
+    }
+
+    if (stop) {
+        for (size_t i = 0; i < mPacketSources.size(); i++) {
+            mPacketSources.valueAt(i)->queueAccessUnit(mSession->createFormatChangeBuffer());
+        }
+        stopAsync();
+        return OK;
+    }
+
+    mStopParams = params;
+    postMonitorQueue();
+
+    return OK;
+}
+
 void PlaylistFetcher::notifyError(status_t err) {
     sp<AMessage> notify = mNotify->dup();
     notify->setInt32("what", kWhatError);
@@ -456,41 +578,70 @@
 
 void PlaylistFetcher::onMonitorQueue() {
     bool downloadMore = false;
+    refreshPlaylist();
 
-    status_t finalResult;
+    int32_t targetDurationSecs;
+    int64_t targetDurationUs = kMinBufferedDurationUs;
+    if (mPlaylist != NULL) {
+        CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs));
+        targetDurationUs = targetDurationSecs * 1000000ll;
+    }
+
+    // buffer at least 3 times the target duration, or up to 10 seconds
+    int64_t durationToBufferUs = targetDurationUs * 3;
+    if (durationToBufferUs > kMinBufferedDurationUs)  {
+        durationToBufferUs = kMinBufferedDurationUs;
+    }
+
+    int64_t bufferedDurationUs = 0ll;
+    status_t finalResult = NOT_ENOUGH_DATA;
     if (mStreamTypeMask == LiveSession::STREAMTYPE_SUBTITLES) {
         sp<AnotherPacketSource> packetSource =
             mPacketSources.valueFor(LiveSession::STREAMTYPE_SUBTITLES);
 
-        int64_t bufferedDurationUs =
+        bufferedDurationUs =
                 packetSource->getBufferedDurationUs(&finalResult);
-
-        downloadMore = (bufferedDurationUs < kMinBufferedDurationUs);
         finalResult = OK;
     } else {
-        bool first = true;
-        int64_t minBufferedDurationUs = 0ll;
-
+        // Use max stream duration to prevent us from waiting on a non-existent stream;
+        // when we cannot make out from the manifest what streams are included in a playlist
+        // we might assume extra streams.
         for (size_t i = 0; i < mPacketSources.size(); ++i) {
             if ((mStreamTypeMask & mPacketSources.keyAt(i)) == 0) {
                 continue;
             }
 
-            int64_t bufferedDurationUs =
+            int64_t bufferedStreamDurationUs =
                 mPacketSources.valueAt(i)->getBufferedDurationUs(&finalResult);
-
-            if (first || bufferedDurationUs < minBufferedDurationUs) {
-                minBufferedDurationUs = bufferedDurationUs;
-                first = false;
+            ALOGV("buffered %lld for stream %d",
+                    bufferedStreamDurationUs, mPacketSources.keyAt(i));
+            if (bufferedStreamDurationUs > bufferedDurationUs) {
+                bufferedDurationUs = bufferedStreamDurationUs;
             }
         }
+    }
+    downloadMore = (bufferedDurationUs < durationToBufferUs);
 
-        downloadMore =
-            !first && (minBufferedDurationUs < kMinBufferedDurationUs);
+    // signal start if buffered up at least the target size
+    if (!mPrepared && bufferedDurationUs > targetDurationUs && downloadMore) {
+        mPrepared = true;
+
+        ALOGV("prepared, buffered=%lld > %lld",
+                bufferedDurationUs, targetDurationUs);
+        sp<AMessage> msg = mNotify->dup();
+        msg->setInt32("what", kWhatTemporarilyDoneFetching);
+        msg->post();
     }
 
     if (finalResult == OK && downloadMore) {
-        onDownloadNext();
+        ALOGV("monitoring, buffered=%lld < %lld",
+                bufferedDurationUs, durationToBufferUs);
+        // delay the next download slightly; hopefully this gives other concurrent fetchers
+        // a better chance to run.
+        // onDownloadNext();
+        sp<AMessage> msg = new AMessage(kWhatDownloadNext, id());
+        msg->setInt32("generation", mMonitorQueueGeneration);
+        msg->post(1000l);
     } else {
         // Nothing to do yet, try again in a second.
 
@@ -498,15 +649,17 @@
         msg->setInt32("what", kWhatTemporarilyDoneFetching);
         msg->post();
 
-        postMonitorQueue(1000000ll);
+        int64_t delayUs = mPrepared ? kMaxMonitorDelayUs : targetDurationUs / 2;
+        ALOGV("pausing for %lld, buffered=%lld > %lld",
+                delayUs, bufferedDurationUs, durationToBufferUs);
+        // :TRICKY: need to enforce minimum delay because the delay to
+        // refresh the playlist will become 0
+        postMonitorQueue(delayUs, mPrepared ? targetDurationUs * 2 : 0);
     }
 }
 
-void PlaylistFetcher::onDownloadNext() {
-    int64_t nowUs = ALooper::GetNowUs();
-
-    if (mLastPlaylistFetchTimeUs < 0ll
-            || (!mPlaylist->isComplete() && timeToRefreshPlaylist(nowUs))) {
+status_t PlaylistFetcher::refreshPlaylist() {
+    if (delayUsToRefreshPlaylist() <= 0) {
         bool unchanged;
         sp<M3UParser> playlist = mSession->fetchPlaylist(
                 mURI.c_str(), mPlaylistHash, &unchanged);
@@ -522,7 +675,7 @@
             } else {
                 ALOGE("failed to load playlist at url '%s'", mURI.c_str());
                 notifyError(ERROR_IO);
-                return;
+                return ERROR_IO;
             }
         } else {
             mRefreshState = INITIAL_MINIMUM_RELOAD_DELAY;
@@ -535,6 +688,13 @@
 
         mLastPlaylistFetchTimeUs = ALooper::GetNowUs();
     }
+    return OK;
+}
+
+void PlaylistFetcher::onDownloadNext() {
+    if (refreshPlaylist() != OK) {
+        return;
+    }
 
     int32_t firstSeqNumberInPlaylist;
     if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32(
@@ -548,17 +708,29 @@
     const int32_t lastSeqNumberInPlaylist =
         firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1;
 
+    if (mStartup && mSeqNumber >= 0
+            && (mSeqNumber < firstSeqNumberInPlaylist || mSeqNumber > lastSeqNumberInPlaylist)) {
+        // in case we guessed wrong during reconfiguration, try fetching the latest content.
+        mSeqNumber = lastSeqNumberInPlaylist;
+    }
+
     if (mSeqNumber < 0) {
         CHECK_GE(mStartTimeUs, 0ll);
 
         if (mPlaylist->isComplete() || mPlaylist->isEvent()) {
             mSeqNumber = getSeqNumberForTime(mStartTimeUs);
+            ALOGV("Initial sequence number for time %lld is %d from (%d .. %d)",
+                    mStartTimeUs, mSeqNumber, firstSeqNumberInPlaylist,
+                    lastSeqNumberInPlaylist);
         } else {
             // If this is a live session, start 3 segments from the end.
             mSeqNumber = lastSeqNumberInPlaylist - 3;
             if (mSeqNumber < firstSeqNumberInPlaylist) {
                 mSeqNumber = firstSeqNumberInPlaylist;
             }
+            ALOGV("Initial sequence number for live event %d from (%d .. %d)",
+                    mSeqNumber, firstSeqNumberInPlaylist,
+                    lastSeqNumberInPlaylist);
         }
 
         mStartTimeUs = -1ll;
@@ -570,16 +742,35 @@
             ++mNumRetries;
 
             if (mSeqNumber > lastSeqNumberInPlaylist) {
-                mLastPlaylistFetchTimeUs = -1;
-                postMonitorQueue(3000000ll);
+                // refresh in increasing fraction (1/2, 1/3, ...) of the
+                // playlist's target duration or 3 seconds, whichever is less
+                int32_t targetDurationSecs;
+                CHECK(mPlaylist->meta()->findInt32(
+                        "target-duration", &targetDurationSecs));
+                int64_t delayUs = mPlaylist->size() * targetDurationSecs *
+                        1000000ll / (1 + mNumRetries);
+                if (delayUs > kMaxMonitorDelayUs) {
+                    delayUs = kMaxMonitorDelayUs;
+                }
+                ALOGV("sequence number high: %d from (%d .. %d), "
+                      "monitor in %lld (retry=%d)",
+                        mSeqNumber, firstSeqNumberInPlaylist,
+                        lastSeqNumberInPlaylist, delayUs, mNumRetries);
+                postMonitorQueue(delayUs);
                 return;
             }
 
             // we've missed the boat, let's start from the lowest sequence
             // number available and signal a discontinuity.
 
-            ALOGI("We've missed the boat, restarting playback.");
-            mSeqNumber = lastSeqNumberInPlaylist;
+            ALOGI("We've missed the boat, restarting playback."
+                  "  mStartup=%d, was  looking for %d in %d-%d",
+                    mStartup, mSeqNumber, firstSeqNumberInPlaylist,
+                    lastSeqNumberInPlaylist);
+            mSeqNumber = lastSeqNumberInPlaylist - 3;
+            if (mSeqNumber < firstSeqNumberInPlaylist) {
+                mSeqNumber = firstSeqNumberInPlaylist;
+            }
             explicitDiscontinuity = true;
 
             // fall through
@@ -633,6 +824,9 @@
     CHECK(buffer != NULL);
 
     err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, buffer);
+    if (err == OK) {
+        err = checkDecryptPadding(buffer);
+    }
 
     if (err != OK) {
         ALOGE("decryptBuffer failed w/ error %d", err);
@@ -665,6 +859,18 @@
 
     err = extractAndQueueAccessUnits(buffer, itemMeta);
 
+    if (err == -EAGAIN) {
+        // bad starting sequence number hint
+        postMonitorQueue();
+        return;
+    }
+
+    if (err == ERROR_OUT_OF_RANGE) {
+        // reached stopping point
+        stopAsync();
+        return;
+    }
+
     if (err != OK) {
         notifyError(err);
         return;
@@ -721,12 +927,15 @@
         }
 
         if (mTSParser == NULL) {
-            mTSParser = new ATSParser;
+            // Use TS_TIMESTAMPS_ARE_ABSOLUTE so pts carry over between fetchers.
+            mTSParser = new ATSParser(ATSParser::TS_TIMESTAMPS_ARE_ABSOLUTE);
         }
 
         if (mNextPTSTimeUs >= 0ll) {
             sp<AMessage> extra = new AMessage;
-            extra->setInt64(IStreamListener::kKeyMediaTimeUs, mNextPTSTimeUs);
+            // Since we are using absolute timestamps, signal an offset of 0 to prevent
+            // ATSParser from skewing the timestamps of access units.
+            extra->setInt64(IStreamListener::kKeyMediaTimeUs, 0);
 
             mTSParser->signalDiscontinuity(
                     ATSParser::DISCONTINUITY_SEEK, extra);
@@ -745,17 +954,23 @@
             offset += 188;
         }
 
+        status_t err = OK;
         for (size_t i = mPacketSources.size(); i-- > 0;) {
             sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i);
 
+            const char *key;
             ATSParser::SourceType type;
-            switch (mPacketSources.keyAt(i)) {
+            const LiveSession::StreamType stream = mPacketSources.keyAt(i);
+            switch (stream) {
+
                 case LiveSession::STREAMTYPE_VIDEO:
                     type = ATSParser::VIDEO;
+                    key = "timeUsVideo";
                     break;
 
                 case LiveSession::STREAMTYPE_AUDIO:
                     type = ATSParser::AUDIO;
+                    key = "timeUsAudio";
                     break;
 
                 case LiveSession::STREAMTYPE_SUBTITLES:
@@ -782,20 +997,89 @@
                 continue;
             }
 
+            int64_t timeUs;
             sp<ABuffer> accessUnit;
             status_t finalResult;
             while (source->hasBufferAvailable(&finalResult)
                     && source->dequeueAccessUnit(&accessUnit) == OK) {
-                // Note that we do NOT dequeue any discontinuities.
 
+                CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
+                if (mMinStartTimeUs > 0) {
+                    if (timeUs < mMinStartTimeUs) {
+                        // TODO untested path
+                        // try a later ts
+                        int32_t targetDuration;
+                        mPlaylist->meta()->findInt32("target-duration", &targetDuration);
+                        int32_t incr = (mMinStartTimeUs - timeUs) / 1000000 / targetDuration;
+                        if (incr == 0) {
+                            // increment mSeqNumber by at least one
+                            incr = 1;
+                        }
+                        mSeqNumber += incr;
+                        err = -EAGAIN;
+                        break;
+                    } else {
+                        int64_t startTimeUs;
+                        if (mStartTimeUsNotify != NULL
+                                && !mStartTimeUsNotify->findInt64(key, &startTimeUs)) {
+                            mStartTimeUsNotify->setInt64(key, timeUs);
+
+                            uint32_t streamMask = 0;
+                            mStartTimeUsNotify->findInt32("streamMask", (int32_t *) &streamMask);
+                            streamMask |= mPacketSources.keyAt(i);
+                            mStartTimeUsNotify->setInt32("streamMask", streamMask);
+
+                            if (streamMask == mStreamTypeMask) {
+                                mStartTimeUsNotify->post();
+                                mStartTimeUsNotify.clear();
+                            }
+                        }
+                    }
+                }
+
+                if (mStopParams != NULL) {
+                    // Queue discontinuity in original stream.
+                    int64_t stopTimeUs;
+                    if (!mStopParams->findInt64(key, &stopTimeUs) || timeUs >= stopTimeUs) {
+                        packetSource->queueAccessUnit(mSession->createFormatChangeBuffer());
+                        mStreamTypeMask &= ~stream;
+                        mPacketSources.removeItemsAt(i);
+                        break;
+                    }
+                }
+
+                // Note that we do NOT dequeue any discontinuities except for format change.
+
+                // for simplicity, store a reference to the format in each unit
+                sp<MetaData> format = source->getFormat();
+                if (format != NULL) {
+                    accessUnit->meta()->setObject("format", format);
+                }
+
+                // Stash the sequence number so we can hint future fetchers where to start at.
+                accessUnit->meta()->setInt32("seq", mSeqNumber);
                 packetSource->queueAccessUnit(accessUnit);
             }
 
-            if (packetSource->getFormat() == NULL) {
-                packetSource->setFormat(source->getFormat());
+            if (err != OK) {
+                break;
             }
         }
 
+        if (err != OK) {
+            for (size_t i = mPacketSources.size(); i-- > 0;) {
+                sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i);
+                packetSource->clear();
+            }
+            return err;
+        }
+
+        if (!mStreamTypeMask) {
+            // Signal gap is filled between original and new stream.
+            ALOGV("ERROR OUT OF RANGE");
+            return ERROR_OUT_OF_RANGE;
+        }
+
         return OK;
     } else if (buffer->size() >= 7 && !memcmp("WEBVTT\n", buffer->data(), 7)) {
         if (mStreamTypeMask != LiveSession::STREAMTYPE_SUBTITLES) {
@@ -810,6 +1094,7 @@
         CHECK(itemMeta->findInt64("durationUs", &durationUs));
         buffer->meta()->setInt64("timeUs", getSegmentStartTimeUs(mSeqNumber));
         buffer->meta()->setInt64("durationUs", durationUs);
+        buffer->meta()->setInt32("seq", mSeqNumber);
 
         packetSource->queueAccessUnit(buffer);
         return OK;
@@ -935,6 +1220,18 @@
             | (adtsHeader[4] << 3)
             | (adtsHeader[5] >> 5);
 
+        if (aac_frame_length == 0) {
+            const uint8_t *id3Header = adtsHeader;
+            if (!memcmp(id3Header, "ID3", 3)) {
+                ID3 id3(id3Header, buffer->size() - offset, true);
+                if (id3.isValid()) {
+                    offset += id3.rawSize();
+                    continue;
+                };
+            }
+            return ERROR_MALFORMED;
+        }
+
         CHECK_LE(offset + aac_frame_length, buffer->size());
 
         sp<ABuffer> unit = new ABuffer(aac_frame_length);
@@ -946,6 +1243,7 @@
         // Each AAC frame encodes 1024 samples.
         numSamples += 1024;
 
+        unit->meta()->setInt32("seq", mSeqNumber);
         packetSource->queueAccessUnit(unit);
 
         offset += aac_frame_length;
@@ -973,4 +1271,33 @@
     msg->post();
 }
 
+int64_t PlaylistFetcher::resumeThreshold(const sp<AMessage> &msg) {
+    int64_t durationUs, threshold;
+    if (msg->findInt64("durationUs", &durationUs)) {
+        return kNumSkipFrames * durationUs;
+    }
+
+    sp<RefBase> obj;
+    msg->findObject("format", &obj);
+    MetaData *format = static_cast<MetaData *>(obj.get());
+
+    const char *mime;
+    CHECK(format->findCString(kKeyMIMEType, &mime));
+    bool audio = !strncasecmp(mime, "audio/", 6);
+    if (audio) {
+        // Assumes 1000 samples per frame.
+        int32_t sampleRate;
+        CHECK(format->findInt32(kKeySampleRate, &sampleRate));
+        return kNumSkipFrames  /* frames */ * 1000 /* samples */
+                * (1000000 / sampleRate) /* sample duration (us) */;
+    } else {
+        int32_t frameRate;
+        if (format->findInt32(kKeyFrameRate, &frameRate) && frameRate > 0) {
+            return kNumSkipFrames * (1000000 / frameRate);
+        }
+    }
+
+    return 500000ll;
+}
+
 }  // namespace android

media/libstagefright/httplive/PlaylistFetcher.h

diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h
index 1648e02..2e0349f 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.h
+++ b/media/libstagefright/httplive/PlaylistFetcher.h
@@ -43,6 +43,7 @@
         kWhatTemporarilyDoneFetching,
         kWhatPrepared,
         kWhatPreparationFailed,
+        kWhatStartedAt,
     };
 
     PlaylistFetcher(
@@ -56,12 +57,16 @@
             const sp<AnotherPacketSource> &audioSource,
             const sp<AnotherPacketSource> &videoSource,
             const sp<AnotherPacketSource> &subtitleSource,
-            int64_t startTimeUs = -1ll);
+            int64_t startTimeUs = -1ll,
+            int64_t minStartTimeUs = 0ll /* start after this timestamp */,
+            int32_t startSeqNumberHint = -1 /* try starting at this sequence number */);
 
     void pauseAsync();
 
     void stopAsync();
 
+    void resumeUntilAsync(const sp<AMessage> &params);
+
 protected:
     virtual ~PlaylistFetcher();
     virtual void onMessageReceived(const sp<AMessage> &msg);
@@ -76,16 +81,25 @@
         kWhatPause          = 'paus',
         kWhatStop           = 'stop',
         kWhatMonitorQueue   = 'moni',
+        kWhatResumeUntil    = 'rsme',
+        kWhatDownloadNext   = 'dlnx',
     };
 
     static const int64_t kMinBufferedDurationUs;
+    static const int64_t kMaxMonitorDelayUs;
+    static const int32_t kNumSkipFrames;
 
+    // notifications to mSession
     sp<AMessage> mNotify;
+    sp<AMessage> mStartTimeUsNotify;
+
     sp<LiveSession> mSession;
     AString mURI;
 
     uint32_t mStreamTypeMask;
     int64_t mStartTimeUs;
+    int64_t mMinStartTimeUs; // start fetching no earlier than this value
+    sp<AMessage> mStopParams; // message containing the latest timestamps we should fetch.
 
     KeyedVector<LiveSession::StreamType, sp<AnotherPacketSource> >
         mPacketSources;
@@ -97,6 +111,7 @@
     int32_t mSeqNumber;
     int32_t mNumRetries;
     bool mStartup;
+    bool mPrepared;
     int64_t mNextPTSTimeUs;
 
     int32_t mMonitorQueueGeneration;
@@ -117,13 +132,29 @@
     uint64_t mFirstPTS;
     int64_t mAbsoluteTimeAnchorUs;
 
-    status_t decryptBuffer(
-            size_t playlistIndex, const sp<ABuffer> &buffer);
+    // Stores the initialization vector to decrypt the next block of cipher text, which can
+    // either be derived from the sequence number, read from the manifest, or copied from
+    // the last block of cipher text (cipher-block chaining).
+    unsigned char mAESInitVec[16];
 
-    void postMonitorQueue(int64_t delayUs = 0);
+    // Set first to true if decrypting the first segment of a playlist segment. When
+    // first is true, reset the initialization vector based on the available
+    // information in the manifest; otherwise, use the initialization vector as
+    // updated by the last call to AES_cbc_encrypt.
+    //
+    // For the input to decrypt correctly, decryptBuffer must be called on
+    // consecutive byte ranges on block boundaries, e.g. 0..15, 16..47, 48..63,
+    // and so on.
+    status_t decryptBuffer(
+            size_t playlistIndex, const sp<ABuffer> &buffer,
+            bool first = true);
+    status_t checkDecryptPadding(const sp<ABuffer> &buffer);
+
+    void postMonitorQueue(int64_t delayUs = 0, int64_t minDelayUs = 0);
     void cancelMonitorQueue();
 
-    bool timeToRefreshPlaylist(int64_t nowUs) const;
+    int64_t delayUsToRefreshPlaylist() const;
+    status_t refreshPlaylist();
 
     // Returns the media time in us of the segment specified by seqNumber.
     // This is computed by summing the durations of all segments before it.
@@ -135,6 +166,9 @@
     void onMonitorQueue();
     void onDownloadNext();
 
+    // Resume a fetcher to continue until the stopping point stored in msg.
+    status_t onResumeUntil(const sp<AMessage> &msg);
+
     status_t extractAndQueueAccessUnits(
             const sp<ABuffer> &buffer, const sp<AMessage> &itemMeta);
 
@@ -147,6 +181,10 @@
 
     void updateDuration();
 
+    // Before resuming a fetcher in onResume, check the remaining duration is longer than that
+    // returned by resumeThreshold.
+    int64_t resumeThreshold(const sp<AMessage> &msg);
+
     DISALLOW_EVIL_CONSTRUCTORS(PlaylistFetcher);
 };
 

media/libstagefright/id3/Android.mk

diff --git a/media/libstagefright/id3/Android.mk b/media/libstagefright/id3/Android.mk
index bf6f7bb..2194c38 100644
--- a/media/libstagefright/id3/Android.mk
+++ b/media/libstagefright/id3/Android.mk
@@ -4,6 +4,8 @@
 LOCAL_SRC_FILES := \
 	ID3.cpp
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_MODULE := libstagefright_id3
 
 include $(BUILD_STATIC_LIBRARY)
@@ -15,6 +17,8 @@
 LOCAL_SRC_FILES := \
 	testid3.cpp
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_SHARED_LIBRARIES := \
 	libstagefright libutils liblog libbinder libstagefright_foundation
 

media/libstagefright/id3/ID3.cpp

diff --git a/media/libstagefright/id3/ID3.cpp b/media/libstagefright/id3/ID3.cpp
index 34d671a..f0f203c 100644
--- a/media/libstagefright/id3/ID3.cpp
+++ b/media/libstagefright/id3/ID3.cpp
@@ -56,14 +56,14 @@
     DISALLOW_EVIL_CONSTRUCTORS(MemorySource);
 };
 
-ID3::ID3(const sp<DataSource> &source, bool ignoreV1)
+ID3::ID3(const sp<DataSource> &source, bool ignoreV1, off64_t offset)
     : mIsValid(false),
       mData(NULL),
       mSize(0),
       mFirstFrameOffset(0),
       mVersion(ID3_UNKNOWN),
       mRawSize(0) {
-    mIsValid = parseV2(source);
+    mIsValid = parseV2(source, offset);
 
     if (!mIsValid && !ignoreV1) {
         mIsValid = parseV1(source);
@@ -79,7 +79,7 @@
       mRawSize(0) {
     sp<MemorySource> source = new MemorySource(data, size);
 
-    mIsValid = parseV2(source);
+    mIsValid = parseV2(source, 0);
 
     if (!mIsValid && !ignoreV1) {
         mIsValid = parseV1(source);
@@ -115,7 +115,7 @@
     return true;
 }
 
-bool ID3::parseV2(const sp<DataSource> &source) {
+bool ID3::parseV2(const sp<DataSource> &source, off64_t offset) {
 struct id3_header {
     char id[3];
     uint8_t version_major;
@@ -126,7 +126,7 @@
 
     id3_header header;
     if (source->readAt(
-                0, &header, sizeof(header)) != (ssize_t)sizeof(header)) {
+                offset, &header, sizeof(header)) != (ssize_t)sizeof(header)) {
         return false;
     }
 
@@ -185,7 +185,7 @@
     mSize = size;
     mRawSize = mSize + sizeof(header);
 
-    if (source->readAt(sizeof(header), mData, mSize) != (ssize_t)mSize) {
+    if (source->readAt(offset + sizeof(header), mData, mSize) != (ssize_t)mSize) {
         free(mData);
         mData = NULL;
 
@@ -468,49 +468,6 @@
     }
 }
 
-static void convertISO8859ToString8(
-        const uint8_t *data, size_t size,
-        String8 *s) {
-    size_t utf8len = 0;
-    for (size_t i = 0; i < size; ++i) {
-        if (data[i] == '\0') {
-            size = i;
-            break;
-        } else if (data[i] < 0x80) {
-            ++utf8len;
-        } else {
-            utf8len += 2;
-        }
-    }
-
-    if (utf8len == size) {
-        // Only ASCII characters present.
-
-        s->setTo((const char *)data, size);
-        return;
-    }
-
-    char *tmp = new char[utf8len];
-    char *ptr = tmp;
-    for (size_t i = 0; i < size; ++i) {
-        if (data[i] == '\0') {
-            break;
-        } else if (data[i] < 0x80) {
-            *ptr++ = data[i];
-        } else if (data[i] < 0xc0) {
-            *ptr++ = 0xc2;
-            *ptr++ = data[i];
-        } else {
-            *ptr++ = 0xc3;
-            *ptr++ = data[i] - 64;
-        }
-    }
-
-    s->setTo(tmp, utf8len);
-
-    delete[] tmp;
-    tmp = NULL;
-}
 
 // the 2nd argument is used to get the data following the \0 in a comment field
 void ID3::Iterator::getString(String8 *id, String8 *comment) const {
@@ -543,7 +500,9 @@
             return;
         }
 
-        convertISO8859ToString8(frameData, mFrameSize, id);
+        // this is supposed to be ISO-8859-1, but pass it up as-is to the caller, who will figure
+        // out the real encoding
+        id->setTo((const char*)frameData, mFrameSize);
         return;
     }
 
@@ -561,13 +520,13 @@
     }
 
     if (encoding == 0x00) {
-        // ISO 8859-1
-        convertISO8859ToString8(frameData + 1, n, id);
+        // supposedly ISO 8859-1
+        id->setTo((const char*)frameData + 1, n);
     } else if (encoding == 0x03) {
-        // UTF-8
+        // supposedly UTF-8
         id->setTo((const char *)(frameData + 1), n);
     } else if (encoding == 0x02) {
-        // UTF-16 BE, no byte order mark.
+        // supposedly UTF-16 BE, no byte order mark.
         // API wants number of characters, not number of bytes...
         int len = n / 2;
         const char16_t *framedata = (const char16_t *) (frameData + 1);
@@ -583,7 +542,7 @@
         if (framedatacopy != NULL) {
             delete[] framedatacopy;
         }
-    } else {
+    } else if (encoding == 0x01) {
         // UCS-2
         // API wants number of characters, not number of bytes...
         int len = n / 2;
@@ -602,7 +561,27 @@
             framedata++;
             len--;
         }
-        id->setTo(framedata, len);
+
+        // check if the resulting data consists entirely of 8-bit values
+        bool eightBit = true;
+        for (int i = 0; i < len; i++) {
+            if (framedata[i] > 0xff) {
+                eightBit = false;
+                break;
+            }
+        }
+        if (eightBit) {
+            // collapse to 8 bit, then let the media scanner client figure out the real encoding
+            char *frame8 = new char[len];
+            for (int i = 0; i < len; i++) {
+                frame8[i] = framedata[i];
+            }
+            id->setTo(frame8, len);
+            delete [] frame8;
+        } else {
+            id->setTo(framedata, len);
+        }
+
         if (framedatacopy != NULL) {
             delete[] framedatacopy;
         }

media/libstagefright/id3/testid3.cpp

diff --git a/media/libstagefright/id3/testid3.cpp b/media/libstagefright/id3/testid3.cpp
index bc4572c..b2f4188 100644
--- a/media/libstagefright/id3/testid3.cpp
+++ b/media/libstagefright/id3/testid3.cpp
@@ -33,7 +33,7 @@
     const uint8_t *data = (const uint8_t *)_data;
     size_t offset = 0;
     while (offset < size) {
-        printf("0x%04x  ", offset);
+        printf("0x%04zx  ", offset);
 
         size_t n = size - offset;
         if (n > 16) {
@@ -101,7 +101,7 @@
         const void *data = tag.getAlbumArt(&dataSize, &mime);
 
         if (data) {
-            printf("found album art: size=%d mime='%s'\n", dataSize,
+            printf("found album art: size=%zu mime='%s'\n", dataSize,
                    mime.string());
 
             hexdump(data, dataSize > 128 ? 128 : dataSize);

media/libstagefright/include/AwesomePlayer.h

diff --git a/media/libstagefright/include/AwesomePlayer.h b/media/libstagefright/include/AwesomePlayer.h
index b001cf4..a81bbba 100644
--- a/media/libstagefright/include/AwesomePlayer.h
+++ b/media/libstagefright/include/AwesomePlayer.h
@@ -63,6 +63,7 @@
     void setUID(uid_t uid);
 
     status_t setDataSource(
+            const sp<IMediaHTTPService> &httpService,
             const char *uri,
             const KeyedVector<String8, String8> *headers = NULL);
 
@@ -159,6 +160,7 @@
     SystemTimeSource mSystemTimeSource;
     TimeSource *mTimeSource;
 
+    sp<IMediaHTTPService> mHTTPService;
     String8 mUri;
     KeyedVector<String8, String8> mUriHeaders;
 
@@ -247,6 +249,7 @@
     sp<MediaExtractor> mExtractor;
 
     status_t setDataSource_l(
+            const sp<IMediaHTTPService> &httpService,
             const char *uri,
             const KeyedVector<String8, String8> *headers = NULL);
 
@@ -342,6 +345,8 @@
 
     bool    mOffloadAudio;
     bool    mAudioTearDown;
+    bool    mAudioTearDownWasPlaying;
+    int64_t mAudioTearDownPosition;
 
     status_t setVideoScalingMode(int32_t mode);
     status_t setVideoScalingMode_l(int32_t mode);

media/libstagefright/include/ChromiumHTTPDataSource.h

diff --git a/media/libstagefright/include/ChromiumHTTPDataSource.h b/media/libstagefright/include/ChromiumHTTPDataSource.h
deleted file mode 100644
index 785f939..0000000
--- a/media/libstagefright/include/ChromiumHTTPDataSource.h
+++ /dev/null
@@ -1,124 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef CHROME_HTTP_DATA_SOURCE_H_
-
-#define CHROME_HTTP_DATA_SOURCE_H_
-
-#include <media/stagefright/foundation/AString.h>
-#include <utils/threads.h>
-
-#include "HTTPBase.h"
-
-namespace android {
-
-struct SfDelegate;
-
-struct ChromiumHTTPDataSource : public HTTPBase {
-    ChromiumHTTPDataSource(uint32_t flags = 0);
-
-    virtual status_t connect(
-            const char *uri,
-            const KeyedVector<String8, String8> *headers = NULL,
-            off64_t offset = 0);
-
-    virtual void disconnect();
-
-    virtual status_t initCheck() const;
-
-    virtual ssize_t readAt(off64_t offset, void *data, size_t size);
-    virtual status_t getSize(off64_t *size);
-    virtual uint32_t flags();
-
-    virtual sp<DecryptHandle> DrmInitialization(const char *mime);
-
-    virtual void getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client);
-
-    virtual String8 getUri();
-
-    virtual String8 getMIMEType() const;
-
-    virtual status_t reconnectAtOffset(off64_t offset);
-
-    static status_t UpdateProxyConfig(
-            const char *host, int32_t port, const char *exclusionList);
-
-protected:
-    virtual ~ChromiumHTTPDataSource();
-
-private:
-    friend struct SfDelegate;
-
-    enum State {
-        DISCONNECTED,
-        CONNECTING,
-        CONNECTED,
-        READING,
-        DISCONNECTING
-    };
-
-    const uint32_t mFlags;
-
-    mutable Mutex mLock;
-    Condition mCondition;
-
-    State mState;
-
-    SfDelegate *mDelegate;
-
-    AString mURI;
-    KeyedVector<String8, String8> mHeaders;
-
-    off64_t mCurrentOffset;
-
-    // Any connection error or the result of a read operation
-    // (for the lattter this is the number of bytes read, if successful).
-    ssize_t mIOResult;
-
-    int64_t mContentSize;
-
-    String8 mContentType;
-
-    sp<DecryptHandle> mDecryptHandle;
-    DrmManagerClient *mDrmManagerClient;
-
-    void disconnect_l();
-
-    status_t connect_l(
-            const char *uri,
-            const KeyedVector<String8, String8> *headers,
-            off64_t offset);
-
-    static void InitiateRead(
-            ChromiumHTTPDataSource *me, void *data, size_t size);
-
-    void initiateRead(void *data, size_t size);
-
-    void onConnectionEstablished(
-            int64_t contentSize, const char *contentType);
-
-    void onConnectionFailed(status_t err);
-    void onReadCompleted(ssize_t size);
-    void onDisconnectComplete();
-
-    void clearDRMState_l();
-
-    DISALLOW_EVIL_CONSTRUCTORS(ChromiumHTTPDataSource);
-};
-
-}  // namespace android
-
-#endif  // CHROME_HTTP_DATA_SOURCE_H_

media/libstagefright/include/FragmentedMP4Parser.h

diff --git a/media/libstagefright/include/FragmentedMP4Parser.h b/media/libstagefright/include/FragmentedMP4Parser.h
deleted file mode 100644
index dbe02b8..0000000
--- a/media/libstagefright/include/FragmentedMP4Parser.h
+++ /dev/null
@@ -1,274 +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 PARSER_H_
-
-#define PARSER_H_
-
-#include <media/stagefright/foundation/AHandler.h>
-#include <media/stagefright/DataSource.h>
-#include <utils/Vector.h>
-
-namespace android {
-
-struct ABuffer;
-
-struct FragmentedMP4Parser : public AHandler {
-    struct Source : public RefBase {
-        Source() {}
-
-        virtual ssize_t readAt(off64_t offset, void *data, size_t size) = 0;
-        virtual bool isSeekable() = 0;
-
-        protected:
-        virtual ~Source() {}
-
-        private:
-        DISALLOW_EVIL_CONSTRUCTORS(Source);
-    };
-
-    FragmentedMP4Parser();
-
-    void start(const char *filename);
-    void start(const sp<Source> &source);
-    void start(sp<DataSource> &source);
-
-    sp<AMessage> getFormat(bool audio, bool synchronous = false);
-    status_t dequeueAccessUnit(bool audio, sp<ABuffer> *accessUnit, bool synchronous = false);
-    status_t seekTo(bool audio, int64_t timeUs);
-    bool isSeekable() const;
-
-    virtual void onMessageReceived(const sp<AMessage> &msg);
-
-protected:
-    virtual ~FragmentedMP4Parser();
-
-private:
-    enum {
-        kWhatStart,
-        kWhatProceed,
-        kWhatReadMore,
-        kWhatGetFormat,
-        kWhatDequeueAccessUnit,
-        kWhatSeekTo,
-    };
-
-    struct TrackFragment;
-    struct DynamicTrackFragment;
-    struct StaticTrackFragment;
-
-    struct DispatchEntry {
-        uint32_t mType;
-        uint32_t mParentType;
-        status_t (FragmentedMP4Parser::*mHandler)(uint32_t, size_t, uint64_t);
-    };
-
-    struct Container {
-        uint64_t mOffset;
-        uint64_t mBytesRemaining;
-        uint32_t mType;
-        bool mExtendsToEOF;
-    };
-
-    struct SampleDescription {
-        uint32_t mType;
-        uint16_t mDataRefIndex;
-
-        sp<AMessage> mFormat;
-    };
-
-    struct SampleInfo {
-        off64_t mOffset;
-        size_t mSize;
-        uint32_t mPresentationTime;
-        size_t mSampleDescIndex;
-        uint32_t mFlags;
-    };
-
-    struct MediaDataInfo {
-        sp<ABuffer> mBuffer;
-        off64_t mOffset;
-    };
-
-    struct SidxEntry {
-        size_t mSize;
-        uint32_t mDurationUs;
-    };
-
-    struct TrackInfo {
-        enum Flags {
-            kTrackEnabled     = 0x01,
-            kTrackInMovie     = 0x02,
-            kTrackInPreview   = 0x04,
-        };
-
-        uint32_t mTrackID;
-        uint32_t mFlags;
-        uint32_t mDuration;  // This is the duration in terms of movie timescale!
-        uint64_t mSidxDuration; // usec, from sidx box, which can use a different timescale
-
-        uint32_t mMediaTimeScale;
-
-        uint32_t mMediaHandlerType;
-        Vector<SampleDescription> mSampleDescs;
-
-        // from track extends:
-        uint32_t mDefaultSampleDescriptionIndex;
-        uint32_t mDefaultSampleDuration;
-        uint32_t mDefaultSampleSize;
-        uint32_t mDefaultSampleFlags;
-
-        uint32_t mDecodingTime;
-
-        Vector<SidxEntry> mSidx;
-        sp<StaticTrackFragment> mStaticFragment;
-        List<sp<TrackFragment> > mFragments;
-    };
-
-    struct TrackFragmentHeaderInfo {
-        enum Flags {
-            kBaseDataOffsetPresent         = 0x01,
-            kSampleDescriptionIndexPresent = 0x02,
-            kDefaultSampleDurationPresent  = 0x08,
-            kDefaultSampleSizePresent      = 0x10,
-            kDefaultSampleFlagsPresent     = 0x20,
-            kDurationIsEmpty               = 0x10000,
-        };
-
-        uint32_t mTrackID;
-        uint32_t mFlags;
-        uint64_t mBaseDataOffset;
-        uint32_t mSampleDescriptionIndex;
-        uint32_t mDefaultSampleDuration;
-        uint32_t mDefaultSampleSize;
-        uint32_t mDefaultSampleFlags;
-
-        uint64_t mDataOffset;
-    };
-
-    static const DispatchEntry kDispatchTable[];
-
-    sp<Source> mSource;
-    off_t mBufferPos;
-    bool mSuspended;
-    bool mDoneWithMoov;
-    off_t mFirstMoofOffset; // used as the starting point for offsets calculated from the sidx box
-    sp<ABuffer> mBuffer;
-    Vector<Container> mStack;
-    KeyedVector<uint32_t, TrackInfo> mTracks;  // TrackInfo by trackID
-    Vector<MediaDataInfo> mMediaData;
-
-    uint32_t mCurrentTrackID;
-
-    status_t mFinalResult;
-
-    TrackFragmentHeaderInfo mTrackFragmentHeaderInfo;
-
-    status_t onProceed();
-    status_t onDequeueAccessUnit(size_t trackIndex, sp<ABuffer> *accessUnit);
-    status_t onSeekTo(bool wantAudio, int64_t position);
-
-    void enter(off64_t offset, uint32_t type, uint64_t size);
-
-    uint16_t readU16(size_t offset);
-    uint32_t readU32(size_t offset);
-    uint64_t readU64(size_t offset);
-    void skip(off_t distance);
-    status_t need(size_t size);
-    bool fitsContainer(uint64_t size) const;
-
-    status_t parseTrackHeader(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseMediaHeader(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseMediaHandler(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseTrackExtends(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseTrackFragmentHeader(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseTrackFragmentRun(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseVisualSampleEntry(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseAudioSampleEntry(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseSampleSizes(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseCompactSampleSizes(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseSampleToChunk(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseChunkOffsets(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseChunkOffsets64(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseAVCCodecSpecificData(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseESDSCodecSpecificData(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseMediaData(
-            uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseSegmentIndex(
-            uint32_t type, size_t offset, uint64_t size);
-
-    TrackInfo *editTrack(uint32_t trackID, bool createIfNecessary = false);
-
-    ssize_t findTrack(bool wantAudio) const;
-
-    status_t makeAccessUnit(
-            TrackInfo *info,
-            const SampleInfo &sample,
-            const MediaDataInfo &mdatInfo,
-            sp<ABuffer> *accessUnit);
-
-    status_t getSample(
-            TrackInfo *info,
-            sp<TrackFragment> *fragment,
-            SampleInfo *sampleInfo);
-
-    static int CompareSampleLocation(
-        const SampleInfo &sample, const MediaDataInfo &mdatInfo);
-
-    void resumeIfNecessary();
-
-    void copyBuffer(
-            sp<ABuffer> *dst,
-            size_t offset, uint64_t size) const;
-
-    DISALLOW_EVIL_CONSTRUCTORS(FragmentedMP4Parser);
-};
-
-}  // namespace android
-
-#endif  // PARSER_H_
-

media/libstagefright/include/HTTPBase.h

diff --git a/media/libstagefright/include/HTTPBase.h b/media/libstagefright/include/HTTPBase.h
index d4b7f9f..1c3cd5e 100644
--- a/media/libstagefright/include/HTTPBase.h
+++ b/media/libstagefright/include/HTTPBase.h
@@ -48,14 +48,6 @@
 
     virtual status_t setBandwidthStatCollectFreq(int32_t freqMs);
 
-    static status_t UpdateProxyConfig(
-            const char *host, int32_t port, const char *exclusionList);
-
-    void setUID(uid_t uid);
-    bool getUID(uid_t *uid) const;
-
-    static sp<HTTPBase> Create(uint32_t flags = 0);
-
     static void RegisterSocketUserTag(int sockfd, uid_t uid, uint32_t kTag);
     static void UnRegisterSocketUserTag(int sockfd);
 
@@ -87,9 +79,6 @@
     int32_t mPrevEstimatedBandWidthKbps;
     int32_t mBandWidthCollectFreqMs;
 
-    bool mUIDValid;
-    uid_t mUID;
-
     DISALLOW_EVIL_CONSTRUCTORS(HTTPBase);
 };
 

media/libstagefright/include/ID3.h

diff --git a/media/libstagefright/include/ID3.h b/media/libstagefright/include/ID3.h
index cca83ab..e83f3ef 100644
--- a/media/libstagefright/include/ID3.h
+++ b/media/libstagefright/include/ID3.h
@@ -35,7 +35,7 @@
         ID3_V2_4,
     };
 
-    ID3(const sp<DataSource> &source, bool ignoreV1 = false);
+    ID3(const sp<DataSource> &source, bool ignoreV1 = false, off64_t offset = 0);
     ID3(const uint8_t *data, size_t size, bool ignoreV1 = false);
     ~ID3();
 
@@ -86,7 +86,7 @@
     size_t mRawSize;
 
     bool parseV1(const sp<DataSource> &source);
-    bool parseV2(const sp<DataSource> &source);
+    bool parseV2(const sp<DataSource> &source, off64_t offset);
     void removeUnsynchronization();
     bool removeUnsynchronizationV2_4(bool iTunesHack);
 

media/libstagefright/include/MPEG4Extractor.h

diff --git a/media/libstagefright/include/MPEG4Extractor.h b/media/libstagefright/include/MPEG4Extractor.h
index bd5e4b9..7b4bc6d 100644
--- a/media/libstagefright/include/MPEG4Extractor.h
+++ b/media/libstagefright/include/MPEG4Extractor.h
@@ -97,6 +97,7 @@
     status_t parseChunk(off64_t *offset, int depth);
     status_t parseITunesMetaData(off64_t offset, size_t size);
     status_t parse3GPPMetaData(off64_t offset, size_t size, int depth);
+    void parseID3v2MetaData(off64_t offset);
 
     status_t updateAudioTrackInfoFromESDS_MPEG4Audio(
             const void *esds_data, size_t esds_size);

media/libstagefright/include/OMX.h

diff --git a/media/libstagefright/include/OMX.h b/media/libstagefright/include/OMX.h
index 7e53af3..31a5077 100644
--- a/media/libstagefright/include/OMX.h
+++ b/media/libstagefright/include/OMX.h
@@ -71,6 +71,10 @@
     virtual status_t storeMetaDataInBuffers(
             node_id node, OMX_U32 port_index, OMX_BOOL enable);
 
+    virtual status_t prepareForAdaptivePlayback(
+            node_id node, OMX_U32 portIndex, OMX_BOOL enable,
+            OMX_U32 max_frame_width, OMX_U32 max_frame_height);
+
     virtual status_t useBuffer(
             node_id node, OMX_U32 port_index, const sp<IMemory> &params,
             buffer_id *buffer);

media/libstagefright/include/OMXNodeInstance.h

diff --git a/media/libstagefright/include/OMXNodeInstance.h b/media/libstagefright/include/OMXNodeInstance.h
index ae498b4..339179e 100644
--- a/media/libstagefright/include/OMXNodeInstance.h
+++ b/media/libstagefright/include/OMXNodeInstance.h
@@ -58,6 +58,10 @@
 
     status_t storeMetaDataInBuffers(OMX_U32 portIndex, OMX_BOOL enable);
 
+    status_t prepareForAdaptivePlayback(
+            OMX_U32 portIndex, OMX_BOOL enable,
+            OMX_U32 maxFrameWidth, OMX_U32 maxFrameHeight);
+
     status_t useBuffer(
             OMX_U32 portIndex, const sp<IMemory> &params,
             OMX::buffer_id *buffer);

media/libstagefright/include/SDPLoader.h

diff --git a/media/libstagefright/include/SDPLoader.h b/media/libstagefright/include/SDPLoader.h
index ca59dc0..2c4f543 100644
--- a/media/libstagefright/include/SDPLoader.h
+++ b/media/libstagefright/include/SDPLoader.h
@@ -25,6 +25,7 @@
 namespace android {
 
 struct HTTPBase;
+struct IMediaHTTPService;
 
 struct SDPLoader : public AHandler {
     enum Flags {
@@ -34,7 +35,10 @@
     enum {
         kWhatSDPLoaded = 'sdpl'
     };
-    SDPLoader(const sp<AMessage> &notify, uint32_t flags = 0, bool uidValid = false, uid_t uid = 0);
+    SDPLoader(
+            const sp<AMessage> &notify,
+            uint32_t flags,
+            const sp<IMediaHTTPService> &httpService);
 
     void load(const char* url, const KeyedVector<String8, String8> *headers);
 
@@ -55,8 +59,6 @@
     sp<AMessage> mNotify;
     const char* mUrl;
     uint32_t mFlags;
-    bool mUIDValid;
-    uid_t mUID;
     sp<ALooper> mNetLooper;
     bool mCancelled;
 

media/libstagefright/include/StagefrightMetadataRetriever.h

diff --git a/media/libstagefright/include/StagefrightMetadataRetriever.h b/media/libstagefright/include/StagefrightMetadataRetriever.h
index b02ed0e..6632c27 100644
--- a/media/libstagefright/include/StagefrightMetadataRetriever.h
+++ b/media/libstagefright/include/StagefrightMetadataRetriever.h
@@ -33,6 +33,7 @@
     virtual ~StagefrightMetadataRetriever();
 
     virtual status_t setDataSource(
+            const sp<IMediaHTTPService> &httpService,
             const char *url,
             const KeyedVector<String8, String8> *headers);
 

media/libstagefright/include/TimedEventQueue.h

diff --git a/media/libstagefright/include/TimedEventQueue.h b/media/libstagefright/include/TimedEventQueue.h
index 11f844c..2963150 100644
--- a/media/libstagefright/include/TimedEventQueue.h
+++ b/media/libstagefright/include/TimedEventQueue.h
@@ -23,6 +23,7 @@
 #include <utils/List.h>
 #include <utils/RefBase.h>
 #include <utils/threads.h>
+#include <powermanager/IPowerManager.h>
 
 namespace android {
 
@@ -57,6 +58,21 @@
         Event &operator=(const Event &);
     };
 
+    class PMDeathRecipient : public IBinder::DeathRecipient {
+    public:
+                    PMDeathRecipient(TimedEventQueue *queue) : mQueue(queue) {}
+        virtual     ~PMDeathRecipient() {}
+
+        // IBinder::DeathRecipient
+        virtual     void        binderDied(const wp<IBinder>& who);
+
+    private:
+                    PMDeathRecipient(const PMDeathRecipient&);
+                    PMDeathRecipient& operator = (const PMDeathRecipient&);
+
+                    TimedEventQueue *mQueue;
+    };
+
     TimedEventQueue();
     ~TimedEventQueue();
 
@@ -96,14 +112,17 @@
 
     static int64_t getRealTimeUs();
 
+    void clearPowerManager();
+
 private:
     struct QueueItem {
         sp<Event> event;
         int64_t realtime_us;
+        bool has_wakelock;
     };
 
     struct StopEvent : public TimedEventQueue::Event {
-        virtual void fire(TimedEventQueue *queue, int64_t now_us) {
+        virtual void fire(TimedEventQueue *queue, int64_t /* now_us */) {
             queue->mStopped = true;
         }
     };
@@ -118,10 +137,18 @@
     bool mRunning;
     bool mStopped;
 
+    sp<IPowerManager>       mPowerManager;
+    sp<IBinder>             mWakeLockToken;
+    const sp<PMDeathRecipient> mDeathRecipient;
+    uint32_t                mWakeLockCount;
+
     static void *ThreadWrapper(void *me);
     void threadEntry();
 
-    sp<Event> removeEventFromQueue_l(event_id id);
+    sp<Event> removeEventFromQueue_l(event_id id, bool *wakeLocked);
+
+    void acquireWakeLock_l();
+    void releaseWakeLock_l(bool force = false);
 
     TimedEventQueue(const TimedEventQueue &);
     TimedEventQueue &operator=(const TimedEventQueue &);

media/libstagefright/include/chromium_http_stub.h

diff --git a/media/libstagefright/include/chromium_http_stub.h b/media/libstagefright/include/chromium_http_stub.h
deleted file mode 100644
index e0651a4..0000000
--- a/media/libstagefright/include/chromium_http_stub.h
+++ /dev/null
@@ -1,34 +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 CHROMIUM_HTTP_STUB_H_
-#define CHROMIUM_HTTP_STUB_H_
-
-#include <include/HTTPBase.h>
-#include <media/stagefright/DataSource.h>
-
-namespace android {
-extern "C" {
-HTTPBase *createChromiumHTTPDataSource(uint32_t flags);
-
-status_t UpdateChromiumHTTPDataSourceProxyConfig(
-        const char *host, int32_t port, const char *exclusionList);
-
-DataSource *createDataUriSource(const char *uri);
-}
-}
-
-#endif

media/libstagefright/matroska/Android.mk

diff --git a/media/libstagefright/matroska/Android.mk b/media/libstagefright/matroska/Android.mk
index 2d8c1e1..446ff8c 100644
--- a/media/libstagefright/matroska/Android.mk
+++ b/media/libstagefright/matroska/Android.mk
@@ -8,7 +8,7 @@
         $(TOP)/external/libvpx/libwebm \
         $(TOP)/frameworks/native/include/media/openmax \
 
-LOCAL_CFLAGS += -Wno-multichar
+LOCAL_CFLAGS += -Wno-multichar -Werror
 
 LOCAL_MODULE:= libstagefright_matroska
 

media/libstagefright/matroska/MatroskaExtractor.cpp

diff --git a/media/libstagefright/matroska/MatroskaExtractor.cpp b/media/libstagefright/matroska/MatroskaExtractor.cpp
index d260d0f..6ec9263 100644
--- a/media/libstagefright/matroska/MatroskaExtractor.cpp
+++ b/media/libstagefright/matroska/MatroskaExtractor.cpp
@@ -193,7 +193,7 @@
     clearPendingFrames();
 }
 
-status_t MatroskaSource::start(MetaData *params) {
+status_t MatroskaSource::start(MetaData * /* params */) {
     mBlockIter.reset();
 
     return OK;
@@ -313,7 +313,7 @@
 
     *actualFrameTimeUs = -1ll;
 
-    const int64_t seekTimeNs = seekTimeUs * 1000ll;
+    const int64_t seekTimeNs = seekTimeUs * 1000ll - mExtractor->mSeekPreRollNs;
 
     mkvparser::Segment* const pSegment = mExtractor->mSegment;
 
@@ -410,7 +410,7 @@
             // Accept the first key frame
             *actualFrameTimeUs = (block()->GetTime(mCluster) + 500LL) / 1000LL;
             ALOGV("Requested seek point: %lld actual: %lld",
-                  seekTimeUs, actualFrameTimeUs);
+                  seekTimeUs, *actualFrameTimeUs);
             break;
         }
     }
@@ -628,7 +628,8 @@
       mReader(new DataSourceReader(mDataSource)),
       mSegment(NULL),
       mExtractedThumbnails(false),
-      mIsWebm(false) {
+      mIsWebm(false),
+      mSeekPreRollNs(0) {
     off64_t size;
     mIsLiveStreaming =
         (mDataSource->flags()
@@ -716,41 +717,61 @@
     return mIsLiveStreaming;
 }
 
+static int bytesForSize(size_t size) {
+    // use at most 28 bits (4 times 7)
+    CHECK(size <= 0xfffffff);
+
+    if (size > 0x1fffff) {
+        return 4;
+    } else if (size > 0x3fff) {
+        return 3;
+    } else if (size > 0x7f) {
+        return 2;
+    }
+    return 1;
+}
+
+static void storeSize(uint8_t *data, size_t &idx, size_t size) {
+    int numBytes = bytesForSize(size);
+    idx += numBytes;
+
+    data += idx;
+    size_t next = 0;
+    while (numBytes--) {
+        *--data = (size & 0x7f) | next;
+        size >>= 7;
+        next = 0x80;
+    }
+}
+
 static void addESDSFromCodecPrivate(
         const sp<MetaData> &meta,
         bool isAudio, const void *priv, size_t privSize) {
-    static const uint8_t kStaticESDS[] = {
-        0x03, 22,
-        0x00, 0x00,     // ES_ID
-        0x00,           // streamDependenceFlag, URL_Flag, OCRstreamFlag
 
-        0x04, 17,
-        0x40,           // ObjectTypeIndication
-        0x00, 0x00, 0x00, 0x00,
-        0x00, 0x00, 0x00, 0x00,
-        0x00, 0x00, 0x00, 0x00,
-
-        0x05,
-        // CodecSpecificInfo (with size prefix) follows
-    };
-
-    // Make sure all sizes can be coded in a single byte.
-    CHECK(privSize + 22 - 2 < 128);
-    size_t esdsSize = sizeof(kStaticESDS) + privSize + 1;
+    int privSizeBytesRequired = bytesForSize(privSize);
+    int esdsSize2 = 14 + privSizeBytesRequired + privSize;
+    int esdsSize2BytesRequired = bytesForSize(esdsSize2);
+    int esdsSize1 = 4 + esdsSize2BytesRequired + esdsSize2;
+    int esdsSize1BytesRequired = bytesForSize(esdsSize1);
+    size_t esdsSize = 1 + esdsSize1BytesRequired + esdsSize1;
     uint8_t *esds = new uint8_t[esdsSize];
-    memcpy(esds, kStaticESDS, sizeof(kStaticESDS));
-    uint8_t *ptr = esds + sizeof(kStaticESDS);
-    *ptr++ = privSize;
-    memcpy(ptr, priv, privSize);
 
-    // Increment by codecPrivateSize less 2 bytes that are accounted for
-    // already in lengths of 22/17
-    esds[1] += privSize - 2;
-    esds[6] += privSize - 2;
-
-    // Set ObjectTypeIndication.
-    esds[7] = isAudio ? 0x40   // Audio ISO/IEC 14496-3
-                      : 0x20;  // Visual ISO/IEC 14496-2
+    size_t idx = 0;
+    esds[idx++] = 0x03;
+    storeSize(esds, idx, esdsSize1);
+    esds[idx++] = 0x00; // ES_ID
+    esds[idx++] = 0x00; // ES_ID
+    esds[idx++] = 0x00; // streamDependenceFlag, URL_Flag, OCRstreamFlag
+    esds[idx++] = 0x04;
+    storeSize(esds, idx, esdsSize2);
+    esds[idx++] = isAudio ? 0x40   // Audio ISO/IEC 14496-3
+                          : 0x20;  // Visual ISO/IEC 14496-2
+    for (int i = 0; i < 12; i++) {
+        esds[idx++] = 0x00;
+    }
+    esds[idx++] = 0x05;
+    storeSize(esds, idx, privSize);
+    memcpy(esds + idx, priv, privSize);
 
     meta->setData(kKeyESDS, 0, esds, esdsSize);
 
@@ -899,6 +920,12 @@
 
                     err = addVorbisCodecInfo(
                             meta, codecPrivate, codecPrivateSize);
+                } else if (!strcmp("A_OPUS", codecID)) {
+                    meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_OPUS);
+                    meta->setData(kKeyOpusHeader, 0, codecPrivate, codecPrivateSize);
+                    meta->setInt64(kKeyOpusCodecDelay, track->GetCodecDelay());
+                    meta->setInt64(kKeyOpusSeekPreRoll, track->GetSeekPreRoll());
+                    mSeekPreRollNs = track->GetSeekPreRoll();
                 } else if (!strcmp("A_MPEG/L3", codecID)) {
                     meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG);
                 } else {

media/libstagefright/matroska/MatroskaExtractor.h

diff --git a/media/libstagefright/matroska/MatroskaExtractor.h b/media/libstagefright/matroska/MatroskaExtractor.h
index 1294b4f..cf200f3 100644
--- a/media/libstagefright/matroska/MatroskaExtractor.h
+++ b/media/libstagefright/matroska/MatroskaExtractor.h
@@ -69,6 +69,7 @@
     bool mExtractedThumbnails;
     bool mIsLiveStreaming;
     bool mIsWebm;
+    int64_t mSeekPreRollNs;
 
     void addTracks();
     void findThumbnails();

media/libstagefright/mp4/FragmentedMP4Parser.cpp

diff --git a/media/libstagefright/mp4/FragmentedMP4Parser.cpp b/media/libstagefright/mp4/FragmentedMP4Parser.cpp
deleted file mode 100644
index 0102656..0000000
--- a/media/libstagefright/mp4/FragmentedMP4Parser.cpp
+++ /dev/null
@@ -1,1993 +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.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "FragmentedMP4Parser"
-#include <utils/Log.h>
-
-#include "include/avc_utils.h"
-#include "include/ESDS.h"
-#include "include/FragmentedMP4Parser.h"
-#include "TrackFragment.h"
-
-
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/foundation/hexdump.h>
-#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/Utils.h>
-
-
-namespace android {
-
-static const char *Fourcc2String(uint32_t fourcc) {
-    static char buffer[5];
-    buffer[4] = '\0';
-    buffer[0] = fourcc >> 24;
-    buffer[1] = (fourcc >> 16) & 0xff;
-    buffer[2] = (fourcc >> 8) & 0xff;
-    buffer[3] = fourcc & 0xff;
-
-    return buffer;
-}
-
-static const char *IndentString(size_t n) {
-    static const char kSpace[] = "                              ";
-    return kSpace + sizeof(kSpace) - 2 * n - 1;
-}
-
-// static
-const FragmentedMP4Parser::DispatchEntry FragmentedMP4Parser::kDispatchTable[] = {
-    { FOURCC('m', 'o', 'o', 'v'), 0, NULL },
-    { FOURCC('t', 'r', 'a', 'k'), FOURCC('m', 'o', 'o', 'v'), NULL },
-    { FOURCC('u', 'd', 't', 'a'), FOURCC('t', 'r', 'a', 'k'), NULL },
-    { FOURCC('u', 'd', 't', 'a'), FOURCC('m', 'o', 'o', 'v'), NULL },
-    { FOURCC('m', 'e', 't', 'a'), FOURCC('u', 'd', 't', 'a'), NULL },
-    { FOURCC('i', 'l', 's', 't'), FOURCC('m', 'e', 't', 'a'), NULL },
-
-    { FOURCC('t', 'k', 'h', 'd'), FOURCC('t', 'r', 'a', 'k'),
-        &FragmentedMP4Parser::parseTrackHeader
-    },
-
-    { FOURCC('m', 'v', 'e', 'x'), FOURCC('m', 'o', 'o', 'v'), NULL },
-
-    { FOURCC('t', 'r', 'e', 'x'), FOURCC('m', 'v', 'e', 'x'),
-        &FragmentedMP4Parser::parseTrackExtends
-    },
-
-    { FOURCC('e', 'd', 't', 's'), FOURCC('t', 'r', 'a', 'k'), NULL },
-    { FOURCC('m', 'd', 'i', 'a'), FOURCC('t', 'r', 'a', 'k'), NULL },
-
-    { FOURCC('m', 'd', 'h', 'd'), FOURCC('m', 'd', 'i', 'a'),
-        &FragmentedMP4Parser::parseMediaHeader
-    },
-
-    { FOURCC('h', 'd', 'l', 'r'), FOURCC('m', 'd', 'i', 'a'),
-        &FragmentedMP4Parser::parseMediaHandler
-    },
-
-    { FOURCC('m', 'i', 'n', 'f'), FOURCC('m', 'd', 'i', 'a'), NULL },
-    { FOURCC('d', 'i', 'n', 'f'), FOURCC('m', 'i', 'n', 'f'), NULL },
-    { FOURCC('s', 't', 'b', 'l'), FOURCC('m', 'i', 'n', 'f'), NULL },
-    { FOURCC('s', 't', 's', 'd'), FOURCC('s', 't', 'b', 'l'), NULL },
-
-    { FOURCC('s', 't', 's', 'z'), FOURCC('s', 't', 'b', 'l'),
-        &FragmentedMP4Parser::parseSampleSizes },
-
-    { FOURCC('s', 't', 'z', '2'), FOURCC('s', 't', 'b', 'l'),
-        &FragmentedMP4Parser::parseCompactSampleSizes },
-
-    { FOURCC('s', 't', 's', 'c'), FOURCC('s', 't', 'b', 'l'),
-        &FragmentedMP4Parser::parseSampleToChunk },
-
-    { FOURCC('s', 't', 'c', 'o'), FOURCC('s', 't', 'b', 'l'),
-        &FragmentedMP4Parser::parseChunkOffsets },
-
-    { FOURCC('c', 'o', '6', '4'), FOURCC('s', 't', 'b', 'l'),
-        &FragmentedMP4Parser::parseChunkOffsets64 },
-
-    { FOURCC('a', 'v', 'c', 'C'), FOURCC('a', 'v', 'c', '1'),
-        &FragmentedMP4Parser::parseAVCCodecSpecificData },
-
-    { FOURCC('e', 's', 'd', 's'), FOURCC('m', 'p', '4', 'a'),
-        &FragmentedMP4Parser::parseESDSCodecSpecificData },
-
-    { FOURCC('e', 's', 'd', 's'), FOURCC('m', 'p', '4', 'v'),
-        &FragmentedMP4Parser::parseESDSCodecSpecificData },
-
-    { FOURCC('m', 'd', 'a', 't'), 0, &FragmentedMP4Parser::parseMediaData },
-
-    { FOURCC('m', 'o', 'o', 'f'), 0, NULL },
-    { FOURCC('t', 'r', 'a', 'f'), FOURCC('m', 'o', 'o', 'f'), NULL },
-
-    { FOURCC('t', 'f', 'h', 'd'), FOURCC('t', 'r', 'a', 'f'),
-        &FragmentedMP4Parser::parseTrackFragmentHeader
-    },
-    { FOURCC('t', 'r', 'u', 'n'), FOURCC('t', 'r', 'a', 'f'),
-        &FragmentedMP4Parser::parseTrackFragmentRun
-    },
-
-    { FOURCC('m', 'f', 'r', 'a'), 0, NULL },
-
-    { FOURCC('s', 'i', 'd', 'x'), 0, &FragmentedMP4Parser::parseSegmentIndex },
-};
-
-struct FileSource : public FragmentedMP4Parser::Source {
-    FileSource(const char *filename)
-        : mFile(fopen(filename, "rb")) {
-            CHECK(mFile != NULL);
-        }
-
-    virtual ~FileSource() {
-        fclose(mFile);
-    }
-
-    virtual ssize_t readAt(off64_t offset, void *data, size_t size) {
-        fseek(mFile, offset, SEEK_SET);
-        return fread(data, 1, size, mFile);
-    }
-
-    virtual bool isSeekable() {
-        return true;
-    }
-
-    private:
-    FILE *mFile;
-
-    DISALLOW_EVIL_CONSTRUCTORS(FileSource);
-};
-
-struct ReadTracker : public RefBase {
-    ReadTracker(off64_t size) {
-        allocSize = 1 + size / 8192; // 1 bit per kilobyte
-        bitmap = (char*) calloc(1, allocSize);
-    }
-    virtual ~ReadTracker() {
-        dumpToLog();
-        free(bitmap);
-    }
-    void mark(off64_t offset, size_t size) {
-        int firstbit = offset / 1024;
-        int lastbit = (offset + size - 1) / 1024;
-        for (int i = firstbit; i <= lastbit; i++) {
-            bitmap[i/8] |= (0x80 >> (i & 7));
-        }
-    }
-
- private:
-    void dumpToLog() {
-        // 96 chars per line, each char represents one kilobyte, 1 kb per bit
-        int numlines = allocSize / 12;
-        char buf[97];
-        char *cur = bitmap;
-        for (int i = 0; i < numlines; i++ && cur) {
-            for (int j = 0; j < 12; j++) {
-                for (int k = 0; k < 8; k++) {
-                    buf[(j * 8) + k] = (*cur & (0x80 >> k)) ? 'X' : '.';
-                }
-                cur++;
-            }
-            buf[96] = '\0';
-            ALOGI("%5dk: %s", i * 96, buf);
-        }
-    }
-
-    size_t allocSize;
-    char *bitmap;
-};
-
-struct DataSourceSource : public FragmentedMP4Parser::Source {
-    DataSourceSource(sp<DataSource> &source)
-        : mDataSource(source) {
-            CHECK(mDataSource != NULL);
-#if 0
-            off64_t size;
-            if (source->getSize(&size) == OK) {
-                mReadTracker = new ReadTracker(size);
-            } else {
-                ALOGE("couldn't get data source size");
-            }
-#endif
-        }
-
-    virtual ssize_t readAt(off64_t offset, void *data, size_t size) {
-        if (mReadTracker != NULL) {
-            mReadTracker->mark(offset, size);
-        }
-        return mDataSource->readAt(offset, data, size);
-    }
-
-    virtual bool isSeekable() {
-        return true;
-    }
-
-    private:
-    sp<DataSource> mDataSource;
-    sp<ReadTracker> mReadTracker;
-
-    DISALLOW_EVIL_CONSTRUCTORS(DataSourceSource);
-};
-
-FragmentedMP4Parser::FragmentedMP4Parser()
-    : mBufferPos(0),
-      mSuspended(false),
-      mDoneWithMoov(false),
-      mFirstMoofOffset(0),
-      mFinalResult(OK) {
-}
-
-FragmentedMP4Parser::~FragmentedMP4Parser() {
-}
-
-void FragmentedMP4Parser::start(const char *filename) {
-    sp<AMessage> msg = new AMessage(kWhatStart, id());
-    msg->setObject("source", new FileSource(filename));
-    msg->post();
-    ALOGV("Parser::start(%s)", filename);
-}
-
-void FragmentedMP4Parser::start(const sp<Source> &source) {
-    sp<AMessage> msg = new AMessage(kWhatStart, id());
-    msg->setObject("source", source);
-    msg->post();
-    ALOGV("Parser::start(Source)");
-}
-
-void FragmentedMP4Parser::start(sp<DataSource> &source) {
-    sp<AMessage> msg = new AMessage(kWhatStart, id());
-    msg->setObject("source", new DataSourceSource(source));
-    msg->post();
-    ALOGV("Parser::start(DataSource)");
-}
-
-sp<AMessage> FragmentedMP4Parser::getFormat(bool audio, bool synchronous) {
-
-    while (true) {
-        bool moovDone = mDoneWithMoov;
-        sp<AMessage> msg = new AMessage(kWhatGetFormat, id());
-        msg->setInt32("audio", audio);
-
-        sp<AMessage> response;
-        status_t err = msg->postAndAwaitResponse(&response);
-
-        if (err != OK) {
-            ALOGV("getFormat post failed: %d", err);
-            return NULL;
-        }
-
-        if (response->findInt32("err", &err) && err != OK) {
-            if (synchronous && err == -EWOULDBLOCK && !moovDone) {
-                resumeIfNecessary();
-                ALOGV("@getFormat parser not ready yet, retrying");
-                usleep(10000);
-                continue;
-            }
-            ALOGV("getFormat failed: %d", err);
-            return NULL;
-        }
-
-        sp<AMessage> format;
-        CHECK(response->findMessage("format", &format));
-
-        ALOGV("returning format %s", format->debugString().c_str());
-        return format;
-    }
-}
-
-status_t FragmentedMP4Parser::seekTo(bool wantAudio, int64_t timeUs) {
-    sp<AMessage> msg = new AMessage(kWhatSeekTo, id());
-    msg->setInt32("audio", wantAudio);
-    msg->setInt64("position", timeUs);
-
-    sp<AMessage> response;
-    status_t err = msg->postAndAwaitResponse(&response);
-    return err;
-}
-
-bool FragmentedMP4Parser::isSeekable() const {
-    while (mFirstMoofOffset == 0 && mFinalResult == OK) {
-        usleep(10000);
-    }
-    bool seekable = mSource->isSeekable();
-    for (size_t i = 0; seekable && i < mTracks.size(); i++) {
-        const TrackInfo *info = &mTracks.valueAt(i);
-        seekable &= !info->mSidx.empty();
-    }
-    return seekable;
-}
-
-status_t FragmentedMP4Parser::onSeekTo(bool wantAudio, int64_t position) {
-    status_t err = -EINVAL;
-    ssize_t trackIndex = findTrack(wantAudio);
-    if (trackIndex < 0) {
-        err = trackIndex;
-    } else {
-        TrackInfo *info = &mTracks.editValueAt(trackIndex);
-
-        int numSidxEntries = info->mSidx.size();
-        int64_t totalTime = 0;
-        off_t totalOffset = mFirstMoofOffset;
-        for (int i = 0; i < numSidxEntries; i++) {
-            const SidxEntry *se = &info->mSidx[i];
-            if (totalTime + se->mDurationUs > position) {
-                mBuffer->setRange(0,0);
-                mBufferPos = totalOffset;
-                if (mFinalResult == ERROR_END_OF_STREAM) {
-                    mFinalResult = OK;
-                    mSuspended = true; // force resume
-                    resumeIfNecessary();
-                }
-                info->mFragments.clear();
-                info->mDecodingTime = totalTime * info->mMediaTimeScale / 1000000ll;
-                return OK;
-            }
-            totalTime += se->mDurationUs;
-            totalOffset += se->mSize;
-        }
-    }
-    ALOGV("seekTo out of range");
-    return err;
-}
-
-status_t FragmentedMP4Parser::dequeueAccessUnit(bool audio, sp<ABuffer> *accessUnit,
-                                                bool synchronous) {
-
-    while (true) {
-        sp<AMessage> msg = new AMessage(kWhatDequeueAccessUnit, id());
-        msg->setInt32("audio", audio);
-
-        sp<AMessage> response;
-        status_t err = msg->postAndAwaitResponse(&response);
-
-        if (err != OK) {
-            ALOGV("dequeue fail 1: %d", err);
-            return err;
-        }
-
-        if (response->findInt32("err", &err) && err != OK) {
-            if (synchronous && err == -EWOULDBLOCK) {
-                resumeIfNecessary();
-                ALOGV("Parser not ready yet, retrying");
-                usleep(10000);
-                continue;
-            }
-            ALOGV("dequeue fail 2: %d, %d", err, synchronous);
-            return err;
-        }
-
-        CHECK(response->findBuffer("accessUnit", accessUnit));
-
-        return OK;
-    }
-}
-
-ssize_t FragmentedMP4Parser::findTrack(bool wantAudio) const {
-    for (size_t i = 0; i < mTracks.size(); ++i) {
-        const TrackInfo *info = &mTracks.valueAt(i);
-
-        bool isAudio =
-            info->mMediaHandlerType == FOURCC('s', 'o', 'u', 'n');
-
-        bool isVideo =
-            info->mMediaHandlerType == FOURCC('v', 'i', 'd', 'e');
-
-        if ((wantAudio && isAudio) || (!wantAudio && !isAudio)) {
-            if (info->mSampleDescs.empty()) {
-                break;
-            }
-
-            return i;
-        }
-    }
-
-    return -EWOULDBLOCK;
-}
-
-void FragmentedMP4Parser::onMessageReceived(const sp<AMessage> &msg) {
-    switch (msg->what()) {
-        case kWhatStart:
-        {
-            sp<RefBase> obj;
-            CHECK(msg->findObject("source", &obj));
-
-            mSource = static_cast<Source *>(obj.get());
-
-            mBuffer = new ABuffer(512 * 1024);
-            mBuffer->setRange(0, 0);
-
-            enter(0ll, 0, 0);
-
-            (new AMessage(kWhatProceed, id()))->post();
-            break;
-        }
-
-        case kWhatProceed:
-        {
-            CHECK(!mSuspended);
-
-            status_t err = onProceed();
-
-            if (err == OK) {
-                if (!mSuspended) {
-                    msg->post();
-                }
-            } else if (err != -EAGAIN) {
-                ALOGE("onProceed returned error %d", err);
-            }
-
-            break;
-        }
-
-        case kWhatReadMore:
-        {
-            size_t needed;
-            CHECK(msg->findSize("needed", &needed));
-
-            memmove(mBuffer->base(), mBuffer->data(), mBuffer->size());
-            mBufferPos += mBuffer->offset();
-            mBuffer->setRange(0, mBuffer->size());
-
-            size_t maxBytesToRead = mBuffer->capacity() - mBuffer->size();
-
-            if (maxBytesToRead < needed) {
-                ALOGV("resizing buffer.");
-
-                sp<ABuffer> newBuffer =
-                    new ABuffer((mBuffer->size() + needed + 1023) & ~1023);
-                memcpy(newBuffer->data(), mBuffer->data(), mBuffer->size());
-                newBuffer->setRange(0, mBuffer->size());
-
-                mBuffer = newBuffer;
-                maxBytesToRead = mBuffer->capacity() - mBuffer->size();
-            }
-
-            CHECK_GE(maxBytesToRead, needed);
-
-            ssize_t n = mSource->readAt(
-                    mBufferPos + mBuffer->size(),
-                    mBuffer->data() + mBuffer->size(), needed);
-
-            if (n < (ssize_t)needed) {
-                ALOGV("Reached EOF when reading %d @ %d + %d", needed, mBufferPos, mBuffer->size());
-                if (n < 0) {
-                    mFinalResult = n;
-                } else if (n == 0) {
-                    mFinalResult = ERROR_END_OF_STREAM;
-                } else {
-                    mFinalResult = ERROR_IO;
-                }
-            } else {
-                mBuffer->setRange(0, mBuffer->size() + n);
-                (new AMessage(kWhatProceed, id()))->post();
-            }
-
-            break;
-        }
-
-        case kWhatGetFormat:
-        {
-            int32_t wantAudio;
-            CHECK(msg->findInt32("audio", &wantAudio));
-
-            status_t err = -EWOULDBLOCK;
-            sp<AMessage> response = new AMessage;
-
-            ssize_t trackIndex = findTrack(wantAudio);
-
-            if (trackIndex < 0) {
-                err = trackIndex;
-            } else {
-                TrackInfo *info = &mTracks.editValueAt(trackIndex);
-
-                sp<AMessage> format = info->mSampleDescs.itemAt(0).mFormat;
-                if (info->mSidxDuration) {
-                    format->setInt64("durationUs", info->mSidxDuration);
-                } else {
-                    // this is probably going to be zero. Oh well...
-                    format->setInt64("durationUs",
-                                     1000000ll * info->mDuration / info->mMediaTimeScale);
-                }
-                response->setMessage(
-                        "format", format);
-
-                err = OK;
-            }
-
-            response->setInt32("err", err);
-
-            uint32_t replyID;
-            CHECK(msg->senderAwaitsResponse(&replyID));
-
-            response->postReply(replyID);
-            break;
-        }
-
-        case kWhatDequeueAccessUnit:
-        {
-            int32_t wantAudio;
-            CHECK(msg->findInt32("audio", &wantAudio));
-
-            status_t err = -EWOULDBLOCK;
-            sp<AMessage> response = new AMessage;
-
-            ssize_t trackIndex = findTrack(wantAudio);
-
-            if (trackIndex < 0) {
-                err = trackIndex;
-            } else {
-                sp<ABuffer> accessUnit;
-                err = onDequeueAccessUnit(trackIndex, &accessUnit);
-
-                if (err == OK) {
-                    response->setBuffer("accessUnit", accessUnit);
-                }
-            }
-
-            response->setInt32("err", err);
-
-            uint32_t replyID;
-            CHECK(msg->senderAwaitsResponse(&replyID));
-
-            response->postReply(replyID);
-            break;
-        }
-
-        case kWhatSeekTo:
-        {
-            ALOGV("kWhatSeekTo");
-            int32_t wantAudio;
-            CHECK(msg->findInt32("audio", &wantAudio));
-            int64_t position;
-            CHECK(msg->findInt64("position", &position));
-
-            status_t err = -EWOULDBLOCK;
-            sp<AMessage> response = new AMessage;
-
-            ssize_t trackIndex = findTrack(wantAudio);
-
-            if (trackIndex < 0) {
-                err = trackIndex;
-            } else {
-                err = onSeekTo(wantAudio, position);
-            }
-            response->setInt32("err", err);
-            uint32_t replyID;
-            CHECK(msg->senderAwaitsResponse(&replyID));
-            response->postReply(replyID);
-            break;
-        }
-        default:
-            TRESPASS();
-    }
-}
-
-status_t FragmentedMP4Parser::onProceed() {
-    status_t err;
-
-    if ((err = need(8)) != OK) {
-        return err;
-    }
-
-    uint64_t size = readU32(0);
-    uint32_t type = readU32(4);
-
-    size_t offset = 8;
-
-    if (size == 1) {
-        if ((err = need(16)) != OK) {
-            return err;
-        }
-
-        size = readU64(offset);
-        offset += 8;
-    }
-
-    uint8_t userType[16];
-
-    if (type == FOURCC('u', 'u', 'i', 'd')) {
-        if ((err = need(offset + 16)) != OK) {
-            return err;
-        }
-
-        memcpy(userType, mBuffer->data() + offset, 16);
-        offset += 16;
-    }
-
-    CHECK(!mStack.isEmpty());
-    uint32_t ptype = mStack.itemAt(mStack.size() - 1).mType;
-
-    static const size_t kNumDispatchers =
-        sizeof(kDispatchTable) / sizeof(kDispatchTable[0]);
-
-    size_t i;
-    for (i = 0; i < kNumDispatchers; ++i) {
-        if (kDispatchTable[i].mType == type
-                && kDispatchTable[i].mParentType == ptype) {
-            break;
-        }
-    }
-
-    // SampleEntry boxes are container boxes that start with a variable
-    // amount of data depending on the media handler type.
-    // We don't look inside 'hint' type SampleEntry boxes.
-
-    bool isSampleEntryBox =
-        (ptype == FOURCC('s', 't', 's', 'd'))
-        && editTrack(mCurrentTrackID)->mMediaHandlerType
-        != FOURCC('h', 'i', 'n', 't');
-
-    if ((i < kNumDispatchers && kDispatchTable[i].mHandler == 0)
-            || isSampleEntryBox || ptype == FOURCC('i', 'l', 's', 't')) {
-        // This is a container box.
-        if (type == FOURCC('m', 'o', 'o', 'f')) {
-            if (mFirstMoofOffset == 0) {
-                ALOGV("first moof @ %08x", mBufferPos + offset);
-                mFirstMoofOffset = mBufferPos + offset - 8; // point at the size
-            }
-        }
-        if (type == FOURCC('m', 'e', 't', 'a')) {
-            if ((err = need(offset + 4)) < OK) {
-                return err;
-            }
-
-            if (readU32(offset) != 0) {
-                return -EINVAL;
-            }
-
-            offset += 4;
-        } else if (type == FOURCC('s', 't', 's', 'd')) {
-            if ((err = need(offset + 8)) < OK) {
-                return err;
-            }
-
-            if (readU32(offset) != 0) {
-                return -EINVAL;
-            }
-
-            if (readU32(offset + 4) == 0) {
-                // We need at least some entries.
-                return -EINVAL;
-            }
-
-            offset += 8;
-        } else if (isSampleEntryBox) {
-            size_t headerSize;
-
-            switch (editTrack(mCurrentTrackID)->mMediaHandlerType) {
-                case FOURCC('v', 'i', 'd', 'e'):
-                {
-                    // 8 bytes SampleEntry + 70 bytes VisualSampleEntry
-                    headerSize = 78;
-                    break;
-                }
-
-                case FOURCC('s', 'o', 'u', 'n'):
-                {
-                    // 8 bytes SampleEntry + 20 bytes AudioSampleEntry
-                    headerSize = 28;
-                    break;
-                }
-
-                case FOURCC('m', 'e', 't', 'a'):
-                {
-                    headerSize = 8;  // 8 bytes SampleEntry
-                    break;
-                }
-
-                default:
-                    TRESPASS();
-            }
-
-            if (offset + headerSize > size) {
-                return -EINVAL;
-            }
-
-            if ((err = need(offset + headerSize)) != OK) {
-                return err;
-            }
-
-            switch (editTrack(mCurrentTrackID)->mMediaHandlerType) {
-                case FOURCC('v', 'i', 'd', 'e'):
-                {
-                    err = parseVisualSampleEntry(
-                            type, offset, offset + headerSize);
-                    break;
-                }
-
-                case FOURCC('s', 'o', 'u', 'n'):
-                {
-                    err = parseAudioSampleEntry(
-                            type, offset, offset + headerSize);
-                    break;
-                }
-
-                case FOURCC('m', 'e', 't', 'a'):
-                {
-                    err = OK;
-                    break;
-                }
-
-                default:
-                    TRESPASS();
-            }
-
-            if (err != OK) {
-                return err;
-            }
-
-            offset += headerSize;
-        }
-
-        skip(offset);
-
-        ALOGV("%sentering box of type '%s'",
-                IndentString(mStack.size()), Fourcc2String(type));
-
-        enter(mBufferPos - offset, type, size - offset);
-    } else {
-        if (!fitsContainer(size)) {
-            return -EINVAL;
-        }
-
-        if (i < kNumDispatchers && kDispatchTable[i].mHandler != 0) {
-            // We have a handler for this box type.
-
-            if ((err = need(size)) != OK) {
-                return err;
-            }
-
-            ALOGV("%sparsing box of type '%s'",
-                    IndentString(mStack.size()), Fourcc2String(type));
-
-            if ((err = (this->*kDispatchTable[i].mHandler)(
-                            type, offset, size)) != OK) {
-                return err;
-            }
-        } else {
-            // Unknown box type
-
-            ALOGV("%sskipping box of type '%s', size %llu",
-                    IndentString(mStack.size()),
-                    Fourcc2String(type), size);
-
-        }
-
-        skip(size);
-    }
-
-    return OK;
-}
-
-// static
-int FragmentedMP4Parser::CompareSampleLocation(
-        const SampleInfo &sample, const MediaDataInfo &mdatInfo) {
-    if (sample.mOffset + sample.mSize < mdatInfo.mOffset) {
-        return -1;
-    }
-
-    if (sample.mOffset >= mdatInfo.mOffset + mdatInfo.mBuffer->size()) {
-        return 1;
-    }
-
-    // Otherwise make sure the sample is completely contained within this
-    // media data block.
-
-    CHECK_GE(sample.mOffset, mdatInfo.mOffset);
-
-    CHECK_LE(sample.mOffset + sample.mSize,
-             mdatInfo.mOffset + mdatInfo.mBuffer->size());
-
-    return 0;
-}
-
-void FragmentedMP4Parser::resumeIfNecessary() {
-    if (!mSuspended) {
-        return;
-    }
-
-    ALOGV("resuming.");
-
-    mSuspended = false;
-    (new AMessage(kWhatProceed, id()))->post();
-}
-
-status_t FragmentedMP4Parser::getSample(
-        TrackInfo *info, sp<TrackFragment> *fragment, SampleInfo *sampleInfo) {
-    for (;;) {
-        if (info->mFragments.empty()) {
-            if (mFinalResult != OK) {
-                return mFinalResult;
-            }
-
-            resumeIfNecessary();
-            return -EWOULDBLOCK;
-        }
-
-        *fragment = *info->mFragments.begin();
-
-        status_t err = (*fragment)->getSample(sampleInfo);
-
-        if (err == OK) {
-            return OK;
-        } else if (err != ERROR_END_OF_STREAM) {
-            return err;
-        }
-
-        // Really, end of this fragment...
-
-        info->mFragments.erase(info->mFragments.begin());
-    }
-}
-
-status_t FragmentedMP4Parser::onDequeueAccessUnit(
-        size_t trackIndex, sp<ABuffer> *accessUnit) {
-    TrackInfo *info = &mTracks.editValueAt(trackIndex);
-
-    sp<TrackFragment> fragment;
-    SampleInfo sampleInfo;
-    status_t err = getSample(info, &fragment, &sampleInfo);
-
-    if (err == -EWOULDBLOCK) {
-        resumeIfNecessary();
-        return err;
-    } else if (err != OK) {
-        return err;
-    }
-
-    err = -EWOULDBLOCK;
-
-    bool checkDroppable = false;
-
-    for (size_t i = 0; i < mMediaData.size(); ++i) {
-        const MediaDataInfo &mdatInfo = mMediaData.itemAt(i);
-
-        int cmp = CompareSampleLocation(sampleInfo, mdatInfo);
-
-        if (cmp < 0 && !mSource->isSeekable()) {
-            return -EPIPE;
-        } else if (cmp == 0) {
-            if (i > 0) {
-                checkDroppable = true;
-            }
-
-            err = makeAccessUnit(info, sampleInfo, mdatInfo, accessUnit);
-            break;
-        }
-    }
-
-    if (err != OK) {
-        return err;
-    }
-
-    fragment->advance();
-
-    if (!mMediaData.empty() && checkDroppable) {
-        size_t numDroppable = 0;
-        bool done = false;
-
-        // XXX FIXME: if one of the tracks is not advanced (e.g. if you play an audio+video
-        // file with sf2), then mMediaData will not be pruned and keeps growing
-        for (size_t i = 0; !done && i < mMediaData.size(); ++i) {
-            const MediaDataInfo &mdatInfo = mMediaData.itemAt(i);
-
-            for (size_t j = 0; j < mTracks.size(); ++j) {
-                TrackInfo *info = &mTracks.editValueAt(j);
-
-                sp<TrackFragment> fragment;
-                SampleInfo sampleInfo;
-                err = getSample(info, &fragment, &sampleInfo);
-
-                if (err != OK) {
-                    done = true;
-                    break;
-                }
-
-                int cmp = CompareSampleLocation(sampleInfo, mdatInfo);
-
-                if (cmp <= 0) {
-                    done = true;
-                    break;
-                }
-            }
-
-            if (!done) {
-                ++numDroppable;
-            }
-        }
-
-        if (numDroppable > 0) {
-            mMediaData.removeItemsAt(0, numDroppable);
-
-            if (mMediaData.size() < 5) {
-                resumeIfNecessary();
-            }
-        }
-    }
-
-    return err;
-}
-
-static size_t parseNALSize(size_t nalLengthSize, const uint8_t *data) {
-    switch (nalLengthSize) {
-        case 1:
-            return *data;
-        case 2:
-            return U16_AT(data);
-        case 3:
-            return ((size_t)data[0] << 16) | U16_AT(&data[1]);
-        case 4:
-            return U32_AT(data);
-    }
-
-    // This cannot happen, mNALLengthSize springs to life by adding 1 to
-    // a 2-bit integer.
-    TRESPASS();
-
-    return 0;
-}
-
-status_t FragmentedMP4Parser::makeAccessUnit(
-        TrackInfo *info,
-        const SampleInfo &sample,
-        const MediaDataInfo &mdatInfo,
-        sp<ABuffer> *accessUnit) {
-    if (sample.mSampleDescIndex < 1
-            || sample.mSampleDescIndex > info->mSampleDescs.size()) {
-        return ERROR_MALFORMED;
-    }
-
-    int64_t presentationTimeUs =
-        1000000ll * sample.mPresentationTime / info->mMediaTimeScale;
-
-    const SampleDescription &sampleDesc =
-        info->mSampleDescs.itemAt(sample.mSampleDescIndex - 1);
-
-    size_t nalLengthSize;
-    if (!sampleDesc.mFormat->findSize("nal-length-size", &nalLengthSize)) {
-        *accessUnit = new ABuffer(sample.mSize);
-
-        memcpy((*accessUnit)->data(),
-               mdatInfo.mBuffer->data() + (sample.mOffset - mdatInfo.mOffset),
-               sample.mSize);
-
-        (*accessUnit)->meta()->setInt64("timeUs", presentationTimeUs);
-        if (IsIDR(*accessUnit)) {
-            (*accessUnit)->meta()->setInt32("is-sync-frame", 1);
-        }
-
-        return OK;
-    }
-
-    const uint8_t *srcPtr =
-        mdatInfo.mBuffer->data() + (sample.mOffset - mdatInfo.mOffset);
-
-    for (int i = 0; i < 2 ; ++i) {
-        size_t srcOffset = 0;
-        size_t dstOffset = 0;
-
-        while (srcOffset < sample.mSize) {
-            if (srcOffset + nalLengthSize > sample.mSize) {
-                return ERROR_MALFORMED;
-            }
-
-            size_t nalSize = parseNALSize(nalLengthSize, &srcPtr[srcOffset]);
-            srcOffset += nalLengthSize;
-
-            if (srcOffset + nalSize > sample.mSize) {
-                return ERROR_MALFORMED;
-            }
-
-            if (i == 1) {
-                memcpy((*accessUnit)->data() + dstOffset,
-                       "\x00\x00\x00\x01",
-                       4);
-
-                memcpy((*accessUnit)->data() + dstOffset + 4,
-                       srcPtr + srcOffset,
-                       nalSize);
-            }
-
-            srcOffset += nalSize;
-            dstOffset += nalSize + 4;
-        }
-
-        if (i == 0) {
-            (*accessUnit) = new ABuffer(dstOffset);
-            (*accessUnit)->meta()->setInt64(
-                    "timeUs", presentationTimeUs);
-        }
-    }
-    if (IsIDR(*accessUnit)) {
-        (*accessUnit)->meta()->setInt32("is-sync-frame", 1);
-    }
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::need(size_t size) {
-    if (!fitsContainer(size)) {
-        return -EINVAL;
-    }
-
-    if (size <= mBuffer->size()) {
-        return OK;
-    }
-
-    sp<AMessage> msg = new AMessage(kWhatReadMore, id());
-    msg->setSize("needed", size - mBuffer->size());
-    msg->post();
-
-    // ALOGV("need(%d) returning -EAGAIN, only have %d", size, mBuffer->size());
-
-    return -EAGAIN;
-}
-
-void FragmentedMP4Parser::enter(off64_t offset, uint32_t type, uint64_t size) {
-    Container container;
-    container.mOffset = offset;
-    container.mType = type;
-    container.mExtendsToEOF = (size == 0);
-    container.mBytesRemaining = size;
-
-    mStack.push(container);
-}
-
-bool FragmentedMP4Parser::fitsContainer(uint64_t size) const {
-    CHECK(!mStack.isEmpty());
-    const Container &container = mStack.itemAt(mStack.size() - 1);
-
-    return container.mExtendsToEOF || size <= container.mBytesRemaining;
-}
-
-uint16_t FragmentedMP4Parser::readU16(size_t offset) {
-    CHECK_LE(offset + 2, mBuffer->size());
-
-    const uint8_t *ptr = mBuffer->data() + offset;
-    return (ptr[0] << 8) | ptr[1];
-}
-
-uint32_t FragmentedMP4Parser::readU32(size_t offset) {
-    CHECK_LE(offset + 4, mBuffer->size());
-
-    const uint8_t *ptr = mBuffer->data() + offset;
-    return (ptr[0] << 24) | (ptr[1] << 16) | (ptr[2] << 8) | ptr[3];
-}
-
-uint64_t FragmentedMP4Parser::readU64(size_t offset) {
-    return (((uint64_t)readU32(offset)) << 32) | readU32(offset + 4);
-}
-
-void FragmentedMP4Parser::skip(off_t distance) {
-    CHECK(!mStack.isEmpty());
-    for (size_t i = mStack.size(); i-- > 0;) {
-        Container *container = &mStack.editItemAt(i);
-        if (!container->mExtendsToEOF) {
-            CHECK_LE(distance, (off_t)container->mBytesRemaining);
-
-            container->mBytesRemaining -= distance;
-
-            if (container->mBytesRemaining == 0) {
-                ALOGV("%sleaving box of type '%s'",
-                        IndentString(mStack.size() - 1),
-                        Fourcc2String(container->mType));
-
-#if 0
-                if (container->mType == FOURCC('s', 't', 's', 'd')) {
-                    TrackInfo *trackInfo = editTrack(mCurrentTrackID);
-                    for (size_t i = 0;
-                            i < trackInfo->mSampleDescs.size(); ++i) {
-                        ALOGI("format #%d: %s",
-                              i,
-                              trackInfo->mSampleDescs.itemAt(i)
-                                .mFormat->debugString().c_str());
-                    }
-                }
-#endif
-
-                if (container->mType == FOURCC('s', 't', 'b', 'l')) {
-                    TrackInfo *trackInfo = editTrack(mCurrentTrackID);
-
-                    trackInfo->mStaticFragment->signalCompletion();
-
-                    CHECK(trackInfo->mFragments.empty());
-                    trackInfo->mFragments.push_back(trackInfo->mStaticFragment);
-                    trackInfo->mStaticFragment.clear();
-                } else if (container->mType == FOURCC('t', 'r', 'a', 'f')) {
-                    TrackInfo *trackInfo =
-                        editTrack(mTrackFragmentHeaderInfo.mTrackID);
-
-                    const sp<TrackFragment> &fragment =
-                        *--trackInfo->mFragments.end();
-
-                    static_cast<DynamicTrackFragment *>(
-                            fragment.get())->signalCompletion();
-                } else if (container->mType == FOURCC('m', 'o', 'o', 'v')) {
-                    mDoneWithMoov = true;
-                }
-
-                container = NULL;
-                mStack.removeItemsAt(i);
-            }
-        }
-    }
-
-    if (distance < (off_t)mBuffer->size()) {
-        mBuffer->setRange(mBuffer->offset() + distance, mBuffer->size() - distance);
-        mBufferPos += distance;
-        return;
-    }
-
-    mBuffer->setRange(0, 0);
-    mBufferPos += distance;
-}
-
-status_t FragmentedMP4Parser::parseTrackHeader(
-        uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 4 > size) {
-        return -EINVAL;
-    }
-
-    uint32_t flags = readU32(offset);
-
-    uint32_t version = flags >> 24;
-    flags &= 0xffffff;
-
-    uint32_t trackID;
-    uint64_t duration;
-
-    if (version == 1) {
-        if (offset + 36 > size) {
-            return -EINVAL;
-        }
-
-        trackID = readU32(offset + 20);
-        duration = readU64(offset + 28);
-
-        offset += 36;
-    } else if (version == 0) {
-        if (offset + 24 > size) {
-            return -EINVAL;
-        }
-
-        trackID = readU32(offset + 12);
-        duration = readU32(offset + 20);
-
-        offset += 24;
-    } else {
-        return -EINVAL;
-    }
-
-    TrackInfo *info = editTrack(trackID, true /* createIfNecessary */);
-    info->mFlags = flags;
-    info->mDuration = duration;
-    if (info->mDuration == 0xffffffff) {
-        // ffmpeg sets this to -1, which is incorrect.
-        info->mDuration = 0;
-    }
-
-    info->mStaticFragment = new StaticTrackFragment;
-
-    mCurrentTrackID = trackID;
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::parseMediaHeader(
-        uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 4 > size) {
-        return -EINVAL;
-    }
-
-    uint32_t versionAndFlags = readU32(offset);
-
-    if (versionAndFlags & 0xffffff) {
-        return ERROR_MALFORMED;
-    }
-
-    uint32_t version = versionAndFlags >> 24;
-
-    TrackInfo *info = editTrack(mCurrentTrackID);
-
-    if (version == 1) {
-        if (offset + 4 + 32 > size) {
-            return -EINVAL;
-        }
-        info->mMediaTimeScale = U32_AT(mBuffer->data() + offset + 20);
-    } else if (version == 0) {
-        if (offset + 4 + 20 > size) {
-            return -EINVAL;
-        }
-        info->mMediaTimeScale = U32_AT(mBuffer->data() + offset + 12);
-    } else {
-        return ERROR_MALFORMED;
-    }
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::parseMediaHandler(
-        uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 12 > size) {
-        return -EINVAL;
-    }
-
-    if (readU32(offset) != 0) {
-        return -EINVAL;
-    }
-
-    uint32_t handlerType = readU32(offset + 8);
-
-    switch (handlerType) {
-        case FOURCC('v', 'i', 'd', 'e'):
-        case FOURCC('s', 'o', 'u', 'n'):
-        case FOURCC('h', 'i', 'n', 't'):
-        case FOURCC('m', 'e', 't', 'a'):
-            break;
-
-        default:
-            return -EINVAL;
-    }
-
-    editTrack(mCurrentTrackID)->mMediaHandlerType = handlerType;
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::parseVisualSampleEntry(
-        uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 78 > size) {
-        return -EINVAL;
-    }
-
-    TrackInfo *trackInfo = editTrack(mCurrentTrackID);
-
-    trackInfo->mSampleDescs.push();
-    SampleDescription *sampleDesc =
-        &trackInfo->mSampleDescs.editItemAt(
-                trackInfo->mSampleDescs.size() - 1);
-
-    sampleDesc->mType = type;
-    sampleDesc->mDataRefIndex = readU16(offset + 6);
-
-    sp<AMessage> format = new AMessage;
-
-    switch (type) {
-        case FOURCC('a', 'v', 'c', '1'):
-            format->setString("mime", MEDIA_MIMETYPE_VIDEO_AVC);
-            break;
-        case FOURCC('m', 'p', '4', 'v'):
-            format->setString("mime", MEDIA_MIMETYPE_VIDEO_MPEG4);
-            break;
-        case FOURCC('s', '2', '6', '3'):
-        case FOURCC('h', '2', '6', '3'):
-        case FOURCC('H', '2', '6', '3'):
-            format->setString("mime", MEDIA_MIMETYPE_VIDEO_H263);
-            break;
-        default:
-            format->setString("mime", "application/octet-stream");
-            break;
-    }
-
-    format->setInt32("width", readU16(offset + 8 + 16));
-    format->setInt32("height", readU16(offset + 8 + 18));
-
-    sampleDesc->mFormat = format;
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::parseAudioSampleEntry(
-        uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 28 > size) {
-        return -EINVAL;
-    }
-
-    TrackInfo *trackInfo = editTrack(mCurrentTrackID);
-
-    trackInfo->mSampleDescs.push();
-    SampleDescription *sampleDesc =
-        &trackInfo->mSampleDescs.editItemAt(
-                trackInfo->mSampleDescs.size() - 1);
-
-    sampleDesc->mType = type;
-    sampleDesc->mDataRefIndex = readU16(offset + 6);
-
-    sp<AMessage> format = new AMessage;
-
-    format->setInt32("channel-count", readU16(offset + 8 + 8));
-    format->setInt32("sample-size", readU16(offset + 8 + 10));
-    format->setInt32("sample-rate", readU32(offset + 8 + 16) / 65536.0f);
-
-    switch (type) {
-        case FOURCC('m', 'p', '4', 'a'):
-            format->setString("mime", MEDIA_MIMETYPE_AUDIO_AAC);
-            break;
-
-        case FOURCC('s', 'a', 'm', 'r'):
-            format->setString("mime", MEDIA_MIMETYPE_AUDIO_AMR_NB);
-            format->setInt32("channel-count", 1);
-            format->setInt32("sample-rate", 8000);
-            break;
-
-        case FOURCC('s', 'a', 'w', 'b'):
-            format->setString("mime", MEDIA_MIMETYPE_AUDIO_AMR_WB);
-            format->setInt32("channel-count", 1);
-            format->setInt32("sample-rate", 16000);
-            break;
-        default:
-            format->setString("mime", "application/octet-stream");
-            break;
-    }
-
-    sampleDesc->mFormat = format;
-
-    return OK;
-}
-
-static void addCodecSpecificData(
-        const sp<AMessage> &format, int32_t index,
-        const void *data, size_t size,
-        bool insertStartCode = false) {
-    sp<ABuffer> csd = new ABuffer(insertStartCode ? size + 4 : size);
-
-    memcpy(csd->data() + (insertStartCode ? 4 : 0), data, size);
-
-    if (insertStartCode) {
-        memcpy(csd->data(), "\x00\x00\x00\x01", 4);
-    }
-
-    csd->meta()->setInt32("csd", true);
-    csd->meta()->setInt64("timeUs", 0ll);
-
-    format->setBuffer(StringPrintf("csd-%d", index).c_str(), csd);
-}
-
-status_t FragmentedMP4Parser::parseSampleSizes(
-        uint32_t type, size_t offset, uint64_t size) {
-    return editTrack(mCurrentTrackID)->mStaticFragment->parseSampleSizes(
-            this, type, offset, size);
-}
-
-status_t FragmentedMP4Parser::parseCompactSampleSizes(
-        uint32_t type, size_t offset, uint64_t size) {
-    return editTrack(mCurrentTrackID)->mStaticFragment->parseCompactSampleSizes(
-            this, type, offset, size);
-}
-
-status_t FragmentedMP4Parser::parseSampleToChunk(
-        uint32_t type, size_t offset, uint64_t size) {
-    return editTrack(mCurrentTrackID)->mStaticFragment->parseSampleToChunk(
-            this, type, offset, size);
-}
-
-status_t FragmentedMP4Parser::parseChunkOffsets(
-        uint32_t type, size_t offset, uint64_t size) {
-    return editTrack(mCurrentTrackID)->mStaticFragment->parseChunkOffsets(
-            this, type, offset, size);
-}
-
-status_t FragmentedMP4Parser::parseChunkOffsets64(
-        uint32_t type, size_t offset, uint64_t size) {
-    return editTrack(mCurrentTrackID)->mStaticFragment->parseChunkOffsets64(
-            this, type, offset, size);
-}
-
-status_t FragmentedMP4Parser::parseAVCCodecSpecificData(
-        uint32_t type, size_t offset, uint64_t size) {
-    TrackInfo *trackInfo = editTrack(mCurrentTrackID);
-
-    SampleDescription *sampleDesc =
-        &trackInfo->mSampleDescs.editItemAt(
-                trackInfo->mSampleDescs.size() - 1);
-
-    if (sampleDesc->mType != FOURCC('a', 'v', 'c', '1')) {
-        return -EINVAL;
-    }
-
-    const uint8_t *ptr = mBuffer->data() + offset;
-
-    size -= offset;
-    offset = 0;
-
-    if (size < 7 || ptr[0] != 0x01) {
-        return ERROR_MALFORMED;
-    }
-
-    sampleDesc->mFormat->setSize("nal-length-size", 1 + (ptr[4] & 3));
-
-    size_t numSPS = ptr[5] & 31;
-
-    ptr += 6;
-    size -= 6;
-
-    for (size_t i = 0; i < numSPS; ++i) {
-        if (size < 2) {
-            return ERROR_MALFORMED;
-        }
-
-        size_t length = U16_AT(ptr);
-
-        ptr += 2;
-        size -= 2;
-
-        if (size < length) {
-            return ERROR_MALFORMED;
-        }
-
-        addCodecSpecificData(
-                sampleDesc->mFormat, i, ptr, length,
-                true /* insertStartCode */);
-
-        ptr += length;
-        size -= length;
-    }
-
-    if (size < 1) {
-        return ERROR_MALFORMED;
-    }
-
-    size_t numPPS = *ptr;
-    ++ptr;
-    --size;
-
-    for (size_t i = 0; i < numPPS; ++i) {
-        if (size < 2) {
-            return ERROR_MALFORMED;
-        }
-
-        size_t length = U16_AT(ptr);
-
-        ptr += 2;
-        size -= 2;
-
-        if (size < length) {
-            return ERROR_MALFORMED;
-        }
-
-        addCodecSpecificData(
-                sampleDesc->mFormat, numSPS + i, ptr, length,
-                true /* insertStartCode */);
-
-        ptr += length;
-        size -= length;
-    }
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::parseESDSCodecSpecificData(
-        uint32_t type, size_t offset, uint64_t size) {
-    TrackInfo *trackInfo = editTrack(mCurrentTrackID);
-
-    SampleDescription *sampleDesc =
-        &trackInfo->mSampleDescs.editItemAt(
-                trackInfo->mSampleDescs.size() - 1);
-
-    if (sampleDesc->mType != FOURCC('m', 'p', '4', 'a')
-            && sampleDesc->mType != FOURCC('m', 'p', '4', 'v')) {
-        return -EINVAL;
-    }
-
-    const uint8_t *ptr = mBuffer->data() + offset;
-
-    size -= offset;
-    offset = 0;
-
-    if (size < 4) {
-        return -EINVAL;
-    }
-
-    if (U32_AT(ptr) != 0) {
-        return -EINVAL;
-    }
-
-    ptr += 4;
-    size -=4;
-
-    ESDS esds(ptr, size);
-
-    uint8_t objectTypeIndication;
-    if (esds.getObjectTypeIndication(&objectTypeIndication) != OK) {
-        return ERROR_MALFORMED;
-    }
-
-    const uint8_t *csd;
-    size_t csd_size;
-    if (esds.getCodecSpecificInfo(
-                (const void **)&csd, &csd_size) != OK) {
-        return ERROR_MALFORMED;
-    }
-
-    addCodecSpecificData(sampleDesc->mFormat, 0, csd, csd_size);
-
-    if (sampleDesc->mType != FOURCC('m', 'p', '4', 'a')) {
-        return OK;
-    }
-
-    if (csd_size == 0) {
-        // There's no further information, i.e. no codec specific data
-        // Let's assume that the information provided in the mpeg4 headers
-        // is accurate and hope for the best.
-
-        return OK;
-    }
-
-    if (csd_size < 2) {
-        return ERROR_MALFORMED;
-    }
-
-    uint32_t objectType = csd[0] >> 3;
-
-    if (objectType == 31) {
-        return ERROR_UNSUPPORTED;
-    }
-
-    uint32_t freqIndex = (csd[0] & 7) << 1 | (csd[1] >> 7);
-    int32_t sampleRate = 0;
-    int32_t numChannels = 0;
-    if (freqIndex == 15) {
-        if (csd_size < 5) {
-            return ERROR_MALFORMED;
-        }
-
-        sampleRate = (csd[1] & 0x7f) << 17
-                        | csd[2] << 9
-                        | csd[3] << 1
-                        | (csd[4] >> 7);
-
-        numChannels = (csd[4] >> 3) & 15;
-    } else {
-        static uint32_t kSamplingRate[] = {
-            96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
-            16000, 12000, 11025, 8000, 7350
-        };
-
-        if (freqIndex == 13 || freqIndex == 14) {
-            return ERROR_MALFORMED;
-        }
-
-        sampleRate = kSamplingRate[freqIndex];
-        numChannels = (csd[1] >> 3) & 15;
-    }
-
-    if (numChannels == 0) {
-        return ERROR_UNSUPPORTED;
-    }
-
-    sampleDesc->mFormat->setInt32("sample-rate", sampleRate);
-    sampleDesc->mFormat->setInt32("channel-count", numChannels);
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::parseMediaData(
-        uint32_t type, size_t offset, uint64_t size) {
-    ALOGV("skipping 'mdat' chunk at offsets 0x%08lx-0x%08llx.",
-          mBufferPos + offset, mBufferPos + size);
-
-    sp<ABuffer> buffer = new ABuffer(size - offset);
-    memcpy(buffer->data(), mBuffer->data() + offset, size - offset);
-
-    mMediaData.push();
-    MediaDataInfo *info = &mMediaData.editItemAt(mMediaData.size() - 1);
-    info->mBuffer = buffer;
-    info->mOffset = mBufferPos + offset;
-
-    if (mMediaData.size() > 10) {
-        ALOGV("suspending for now.");
-        mSuspended = true;
-    }
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::parseSegmentIndex(
-        uint32_t type, size_t offset, uint64_t size) {
-    ALOGV("sidx box type %d, offset %d, size %d", type, int(offset), int(size));
-//    AString sidxstr;
-//    hexdump(mBuffer->data() + offset, size, 0 /* indent */, &sidxstr);
-//    ALOGV("raw sidx:");
-//    ALOGV("%s", sidxstr.c_str());
-    if (offset + 12 > size) {
-        return -EINVAL;
-    }
-
-    uint32_t flags = readU32(offset);
-
-    uint32_t version = flags >> 24;
-    flags &= 0xffffff;
-
-    ALOGV("sidx version %d", version);
-
-    uint32_t referenceId = readU32(offset + 4);
-    uint32_t timeScale = readU32(offset + 8);
-    ALOGV("sidx refid/timescale: %d/%d", referenceId, timeScale);
-
-    uint64_t earliestPresentationTime;
-    uint64_t firstOffset;
-
-    offset += 12;
-
-    if (version == 0) {
-        if (offset + 8 > size) {
-            return -EINVAL;
-        }
-        earliestPresentationTime = readU32(offset);
-        firstOffset = readU32(offset + 4);
-        offset += 8;
-    } else {
-        if (offset + 16 > size) {
-            return -EINVAL;
-        }
-        earliestPresentationTime = readU64(offset);
-        firstOffset = readU64(offset + 8);
-        offset += 16;
-    }
-    ALOGV("sidx pres/off: %Ld/%Ld", earliestPresentationTime, firstOffset);
-
-    if (offset + 4 > size) {
-        return -EINVAL;
-    }
-    if (readU16(offset) != 0) { // reserved
-        return -EINVAL;
-    }
-    int32_t referenceCount = readU16(offset + 2);
-    offset += 4;
-    ALOGV("refcount: %d", referenceCount);
-
-    if (offset + referenceCount * 12 > size) {
-        return -EINVAL;
-    }
-
-    TrackInfo *info = editTrack(mCurrentTrackID);
-    uint64_t total_duration = 0;
-    for (int i = 0; i < referenceCount; i++) {
-        uint32_t d1 = readU32(offset);
-        uint32_t d2 = readU32(offset + 4);
-        uint32_t d3 = readU32(offset + 8);
-
-        if (d1 & 0x80000000) {
-            ALOGW("sub-sidx boxes not supported yet");
-        }
-        bool sap = d3 & 0x80000000;
-        bool saptype = d3 >> 28;
-        if (!sap || saptype > 2) {
-            ALOGW("not a stream access point, or unsupported type");
-        }
-        total_duration += d2;
-        offset += 12;
-        ALOGV(" item %d, %08x %08x %08x", i, d1, d2, d3);
-        SidxEntry se;
-        se.mSize = d1 & 0x7fffffff;
-        se.mDurationUs = 1000000LL * d2 / timeScale;
-        info->mSidx.add(se);
-    }
-
-    info->mSidxDuration = total_duration * 1000000 / timeScale;
-    ALOGV("duration: %lld", info->mSidxDuration);
-    return OK;
-}
-
-status_t FragmentedMP4Parser::parseTrackExtends(
-        uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 24 > size) {
-        return -EINVAL;
-    }
-
-    if (readU32(offset) != 0) {
-        return -EINVAL;
-    }
-
-    uint32_t trackID = readU32(offset + 4);
-
-    TrackInfo *info = editTrack(trackID, true /* createIfNecessary */);
-    info->mDefaultSampleDescriptionIndex = readU32(offset + 8);
-    info->mDefaultSampleDuration = readU32(offset + 12);
-    info->mDefaultSampleSize = readU32(offset + 16);
-    info->mDefaultSampleFlags = readU32(offset + 20);
-
-    return OK;
-}
-
-FragmentedMP4Parser::TrackInfo *FragmentedMP4Parser::editTrack(
-        uint32_t trackID, bool createIfNecessary) {
-    ssize_t i = mTracks.indexOfKey(trackID);
-
-    if (i >= 0) {
-        return &mTracks.editValueAt(i);
-    }
-
-    if (!createIfNecessary) {
-        return NULL;
-    }
-
-    TrackInfo info;
-    info.mTrackID = trackID;
-    info.mFlags = 0;
-    info.mDuration = 0xffffffff;
-    info.mSidxDuration = 0;
-    info.mMediaTimeScale = 0;
-    info.mMediaHandlerType = 0;
-    info.mDefaultSampleDescriptionIndex = 0;
-    info.mDefaultSampleDuration = 0;
-    info.mDefaultSampleSize = 0;
-    info.mDefaultSampleFlags = 0;
-
-    info.mDecodingTime = 0;
-
-    mTracks.add(trackID, info);
-    return &mTracks.editValueAt(mTracks.indexOfKey(trackID));
-}
-
-status_t FragmentedMP4Parser::parseTrackFragmentHeader(
-        uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 8 > size) {
-        return -EINVAL;
-    }
-
-    uint32_t flags = readU32(offset);
-
-    if (flags & 0xff000000) {
-        return -EINVAL;
-    }
-
-    mTrackFragmentHeaderInfo.mFlags = flags;
-
-    mTrackFragmentHeaderInfo.mTrackID = readU32(offset + 4);
-    offset += 8;
-
-    if (flags & TrackFragmentHeaderInfo::kBaseDataOffsetPresent) {
-        if (offset + 8 > size) {
-            return -EINVAL;
-        }
-
-        mTrackFragmentHeaderInfo.mBaseDataOffset = readU64(offset);
-        offset += 8;
-    }
-
-    if (flags & TrackFragmentHeaderInfo::kSampleDescriptionIndexPresent) {
-        if (offset + 4 > size) {
-            return -EINVAL;
-        }
-
-        mTrackFragmentHeaderInfo.mSampleDescriptionIndex = readU32(offset);
-        offset += 4;
-    }
-
-    if (flags & TrackFragmentHeaderInfo::kDefaultSampleDurationPresent) {
-        if (offset + 4 > size) {
-            return -EINVAL;
-        }
-
-        mTrackFragmentHeaderInfo.mDefaultSampleDuration = readU32(offset);
-        offset += 4;
-    }
-
-    if (flags & TrackFragmentHeaderInfo::kDefaultSampleSizePresent) {
-        if (offset + 4 > size) {
-            return -EINVAL;
-        }
-
-        mTrackFragmentHeaderInfo.mDefaultSampleSize = readU32(offset);
-        offset += 4;
-    }
-
-    if (flags & TrackFragmentHeaderInfo::kDefaultSampleFlagsPresent) {
-        if (offset + 4 > size) {
-            return -EINVAL;
-        }
-
-        mTrackFragmentHeaderInfo.mDefaultSampleFlags = readU32(offset);
-        offset += 4;
-    }
-
-    if (!(flags & TrackFragmentHeaderInfo::kBaseDataOffsetPresent)) {
-        // This should point to the position of the first byte of the
-        // enclosing 'moof' container for the first track and
-        // the end of the data of the preceding fragment for subsequent
-        // tracks.
-
-        CHECK_GE(mStack.size(), 2u);
-
-        mTrackFragmentHeaderInfo.mBaseDataOffset =
-            mStack.itemAt(mStack.size() - 2).mOffset;
-
-        // XXX TODO: This does not do the right thing for the 2nd and
-        // subsequent tracks yet.
-    }
-
-    mTrackFragmentHeaderInfo.mDataOffset =
-        mTrackFragmentHeaderInfo.mBaseDataOffset;
-
-    TrackInfo *trackInfo = editTrack(mTrackFragmentHeaderInfo.mTrackID);
-
-    if (trackInfo->mFragments.empty()
-            || (*trackInfo->mFragments.begin())->complete()) {
-        trackInfo->mFragments.push_back(new DynamicTrackFragment);
-    }
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::parseTrackFragmentRun(
-        uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 8 > size) {
-        return -EINVAL;
-    }
-
-    enum {
-        kDataOffsetPresent                  = 0x01,
-        kFirstSampleFlagsPresent            = 0x04,
-        kSampleDurationPresent              = 0x100,
-        kSampleSizePresent                  = 0x200,
-        kSampleFlagsPresent                 = 0x400,
-        kSampleCompositionTimeOffsetPresent = 0x800,
-    };
-
-    uint32_t flags = readU32(offset);
-
-    if (flags & 0xff000000) {
-        return -EINVAL;
-    }
-
-    if ((flags & kFirstSampleFlagsPresent) && (flags & kSampleFlagsPresent)) {
-        // These two shall not be used together.
-        return -EINVAL;
-    }
-
-    uint32_t sampleCount = readU32(offset + 4);
-    offset += 8;
-
-    uint64_t dataOffset = mTrackFragmentHeaderInfo.mDataOffset;
-
-    uint32_t firstSampleFlags = 0;
-
-    if (flags & kDataOffsetPresent) {
-        if (offset + 4 > size) {
-            return -EINVAL;
-        }
-
-        int32_t dataOffsetDelta = (int32_t)readU32(offset);
-
-        dataOffset = mTrackFragmentHeaderInfo.mBaseDataOffset + dataOffsetDelta;
-
-        offset += 4;
-    }
-
-    if (flags & kFirstSampleFlagsPresent) {
-        if (offset + 4 > size) {
-            return -EINVAL;
-        }
-
-        firstSampleFlags = readU32(offset);
-        offset += 4;
-    }
-
-    TrackInfo *info = editTrack(mTrackFragmentHeaderInfo.mTrackID);
-
-    if (info == NULL) {
-        return -EINVAL;
-    }
-
-    uint32_t sampleDuration = 0, sampleSize = 0, sampleFlags = 0,
-             sampleCtsOffset = 0;
-
-    size_t bytesPerSample = 0;
-    if (flags & kSampleDurationPresent) {
-        bytesPerSample += 4;
-    } else if (mTrackFragmentHeaderInfo.mFlags
-            & TrackFragmentHeaderInfo::kDefaultSampleDurationPresent) {
-        sampleDuration = mTrackFragmentHeaderInfo.mDefaultSampleDuration;
-    } else {
-        sampleDuration = info->mDefaultSampleDuration;
-    }
-
-    if (flags & kSampleSizePresent) {
-        bytesPerSample += 4;
-    } else if (mTrackFragmentHeaderInfo.mFlags
-            & TrackFragmentHeaderInfo::kDefaultSampleSizePresent) {
-        sampleSize = mTrackFragmentHeaderInfo.mDefaultSampleSize;
-    } else {
-        sampleSize = info->mDefaultSampleSize;
-    }
-
-    if (flags & kSampleFlagsPresent) {
-        bytesPerSample += 4;
-    } else if (mTrackFragmentHeaderInfo.mFlags
-            & TrackFragmentHeaderInfo::kDefaultSampleFlagsPresent) {
-        sampleFlags = mTrackFragmentHeaderInfo.mDefaultSampleFlags;
-    } else {
-        sampleFlags = info->mDefaultSampleFlags;
-    }
-
-    if (flags & kSampleCompositionTimeOffsetPresent) {
-        bytesPerSample += 4;
-    } else {
-        sampleCtsOffset = 0;
-    }
-
-    if (offset + sampleCount * bytesPerSample > size) {
-        return -EINVAL;
-    }
-
-    uint32_t sampleDescIndex =
-        (mTrackFragmentHeaderInfo.mFlags
-            & TrackFragmentHeaderInfo::kSampleDescriptionIndexPresent)
-            ? mTrackFragmentHeaderInfo.mSampleDescriptionIndex
-            : info->mDefaultSampleDescriptionIndex;
-
-    for (uint32_t i = 0; i < sampleCount; ++i) {
-        if (flags & kSampleDurationPresent) {
-            sampleDuration = readU32(offset);
-            offset += 4;
-        }
-
-        if (flags & kSampleSizePresent) {
-            sampleSize = readU32(offset);
-            offset += 4;
-        }
-
-        if (flags & kSampleFlagsPresent) {
-            sampleFlags = readU32(offset);
-            offset += 4;
-        }
-
-        if (flags & kSampleCompositionTimeOffsetPresent) {
-            sampleCtsOffset = readU32(offset);
-            offset += 4;
-        }
-
-        ALOGV("adding sample at offset 0x%08llx, size %u, duration %u, "
-              "sampleDescIndex=%u, flags 0x%08x",
-                dataOffset, sampleSize, sampleDuration,
-                sampleDescIndex,
-                (flags & kFirstSampleFlagsPresent) && i == 0
-                    ? firstSampleFlags : sampleFlags);
-
-        const sp<TrackFragment> &fragment = *--info->mFragments.end();
-
-        uint32_t decodingTime = info->mDecodingTime;
-        info->mDecodingTime += sampleDuration;
-        uint32_t presentationTime = decodingTime + sampleCtsOffset;
-
-        static_cast<DynamicTrackFragment *>(
-                fragment.get())->addSample(
-                    dataOffset,
-                    sampleSize,
-                    presentationTime,
-                    sampleDescIndex,
-                    ((flags & kFirstSampleFlagsPresent) && i == 0)
-                        ? firstSampleFlags : sampleFlags);
-
-        dataOffset += sampleSize;
-    }
-
-    mTrackFragmentHeaderInfo.mDataOffset = dataOffset;
-
-    return OK;
-}
-
-void FragmentedMP4Parser::copyBuffer(
-        sp<ABuffer> *dst, size_t offset, uint64_t size) const {
-    sp<ABuffer> buf = new ABuffer(size);
-    memcpy(buf->data(), mBuffer->data() + offset, size);
-
-    *dst = buf;
-}
-
-}  // namespace android

media/libstagefright/mp4/TrackFragment.cpp

diff --git a/media/libstagefright/mp4/TrackFragment.cpp b/media/libstagefright/mp4/TrackFragment.cpp
deleted file mode 100644
index 3699038..0000000
--- a/media/libstagefright/mp4/TrackFragment.cpp
+++ /dev/null
@@ -1,364 +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.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "TrackFragment"
-#include <utils/Log.h>
-
-#include "TrackFragment.h"
-
-#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/Utils.h>
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/hexdump.h>
-
-namespace android {
-
-FragmentedMP4Parser::DynamicTrackFragment::DynamicTrackFragment()
-    : mComplete(false),
-      mSampleIndex(0) {
-}
-
-FragmentedMP4Parser::DynamicTrackFragment::~DynamicTrackFragment() {
-}
-
-status_t FragmentedMP4Parser::DynamicTrackFragment::getSample(SampleInfo *info) {
-    if (mSampleIndex >= mSamples.size()) {
-        return mComplete ? ERROR_END_OF_STREAM : -EWOULDBLOCK;
-    }
-
-    *info = mSamples.itemAt(mSampleIndex);
-
-    return OK;
-}
-
-void FragmentedMP4Parser::DynamicTrackFragment::advance() {
-    ++mSampleIndex;
-}
-
-void FragmentedMP4Parser::DynamicTrackFragment::addSample(
-        off64_t dataOffset, size_t sampleSize,
-        uint32_t presentationTime,
-        size_t sampleDescIndex,
-        uint32_t flags) {
-    mSamples.push();
-    SampleInfo *sampleInfo = &mSamples.editItemAt(mSamples.size() - 1);
-
-    sampleInfo->mOffset = dataOffset;
-    sampleInfo->mSize = sampleSize;
-    sampleInfo->mPresentationTime = presentationTime;
-    sampleInfo->mSampleDescIndex = sampleDescIndex;
-    sampleInfo->mFlags = flags;
-}
-
-status_t FragmentedMP4Parser::DynamicTrackFragment::signalCompletion() {
-    mComplete = true;
-
-    return OK;
-}
-
-bool FragmentedMP4Parser::DynamicTrackFragment::complete() const {
-    return mComplete;
-}
-
-////////////////////////////////////////////////////////////////////////////////
-
-FragmentedMP4Parser::StaticTrackFragment::StaticTrackFragment()
-    : mSampleIndex(0),
-      mSampleCount(0),
-      mChunkIndex(0),
-      mSampleToChunkIndex(-1),
-      mSampleToChunkRemaining(0),
-      mPrevChunkIndex(0xffffffff),
-      mNextSampleOffset(0) {
-}
-
-FragmentedMP4Parser::StaticTrackFragment::~StaticTrackFragment() {
-}
-
-status_t FragmentedMP4Parser::StaticTrackFragment::getSample(SampleInfo *info) {
-    if (mSampleIndex >= mSampleCount) {
-        return ERROR_END_OF_STREAM;
-    }
-
-    *info = mSampleInfo;
-
-    ALOGV("returning sample %d at [0x%08llx, 0x%08llx)",
-          mSampleIndex,
-          info->mOffset, info->mOffset + info->mSize);
-
-    return OK;
-}
-
-void FragmentedMP4Parser::StaticTrackFragment::updateSampleInfo() {
-    if (mSampleIndex >= mSampleCount) {
-        return;
-    }
-
-    if (mSampleSizes != NULL) {
-        uint32_t defaultSampleSize = U32_AT(mSampleSizes->data() + 4);
-        if (defaultSampleSize > 0) {
-            mSampleInfo.mSize = defaultSampleSize;
-        } else {
-            mSampleInfo.mSize= U32_AT(mSampleSizes->data() + 12 + 4 * mSampleIndex);
-        }
-    } else {
-        CHECK(mCompactSampleSizes != NULL);
-
-        uint32_t fieldSize = U32_AT(mCompactSampleSizes->data() + 4);
-
-        switch (fieldSize) {
-            case 4:
-            {
-                unsigned byte = mCompactSampleSizes->data()[12 + mSampleIndex / 2];
-                mSampleInfo.mSize = (mSampleIndex & 1) ? byte & 0x0f : byte >> 4;
-                break;
-            }
-
-            case 8:
-            {
-                mSampleInfo.mSize = mCompactSampleSizes->data()[12 + mSampleIndex];
-                break;
-            }
-
-            default:
-            {
-                CHECK_EQ(fieldSize, 16);
-                mSampleInfo.mSize =
-                    U16_AT(mCompactSampleSizes->data() + 12 + mSampleIndex * 2);
-                break;
-            }
-        }
-    }
-
-    CHECK_GT(mSampleToChunkRemaining, 0);
-
-    // The sample desc index is 1-based... XXX
-    mSampleInfo.mSampleDescIndex =
-        U32_AT(mSampleToChunk->data() + 8 + 12 * mSampleToChunkIndex + 8);
-
-    if (mChunkIndex != mPrevChunkIndex) {
-        mPrevChunkIndex = mChunkIndex;
-
-        if (mChunkOffsets != NULL) {
-            uint32_t entryCount = U32_AT(mChunkOffsets->data() + 4);
-
-            if (mChunkIndex >= entryCount) {
-                mSampleIndex = mSampleCount;
-                return;
-            }
-
-            mNextSampleOffset =
-                U32_AT(mChunkOffsets->data() + 8 + 4 * mChunkIndex);
-        } else {
-            CHECK(mChunkOffsets64 != NULL);
-
-            uint32_t entryCount = U32_AT(mChunkOffsets64->data() + 4);
-
-            if (mChunkIndex >= entryCount) {
-                mSampleIndex = mSampleCount;
-                return;
-            }
-
-            mNextSampleOffset =
-                U64_AT(mChunkOffsets64->data() + 8 + 8 * mChunkIndex);
-        }
-    }
-
-    mSampleInfo.mOffset = mNextSampleOffset;
-
-    mSampleInfo.mPresentationTime = 0;
-    mSampleInfo.mFlags = 0;
-}
-
-void FragmentedMP4Parser::StaticTrackFragment::advance() {
-    mNextSampleOffset += mSampleInfo.mSize;
-
-    ++mSampleIndex;
-    if (--mSampleToChunkRemaining == 0) {
-        ++mChunkIndex;
-
-        uint32_t entryCount = U32_AT(mSampleToChunk->data() + 4);
-
-        // If this is the last entry in the sample to chunk table, we will
-        // stay on this entry.
-        if ((uint32_t)(mSampleToChunkIndex + 1) < entryCount) {
-            uint32_t nextChunkIndex =
-                U32_AT(mSampleToChunk->data() + 8 + 12 * (mSampleToChunkIndex + 1));
-
-            CHECK_GE(nextChunkIndex, 1u);
-            --nextChunkIndex;
-
-            if (mChunkIndex >= nextChunkIndex) {
-                CHECK_EQ(mChunkIndex, nextChunkIndex);
-                ++mSampleToChunkIndex;
-            }
-        }
-
-        mSampleToChunkRemaining =
-            U32_AT(mSampleToChunk->data() + 8 + 12 * mSampleToChunkIndex + 4);
-    }
-
-    updateSampleInfo();
-}
-
-static void setU32At(uint8_t *ptr, uint32_t x) {
-    ptr[0] = x >> 24;
-    ptr[1] = (x >> 16) & 0xff;
-    ptr[2] = (x >> 8) & 0xff;
-    ptr[3] = x & 0xff;
-}
-
-status_t FragmentedMP4Parser::StaticTrackFragment::signalCompletion() {
-    mSampleToChunkIndex = 0;
-
-    mSampleToChunkRemaining =
-        (mSampleToChunk == NULL)
-            ? 0
-            : U32_AT(mSampleToChunk->data() + 8 + 12 * mSampleToChunkIndex + 4);
-
-    updateSampleInfo();
-
-    return OK;
-}
-
-bool FragmentedMP4Parser::StaticTrackFragment::complete() const {
-    return true;
-}
-
-status_t FragmentedMP4Parser::StaticTrackFragment::parseSampleSizes(
-        FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 12 > size) {
-        return ERROR_MALFORMED;
-    }
-
-    if (parser->readU32(offset) != 0) {
-        return ERROR_MALFORMED;
-    }
-
-    uint32_t sampleSize = parser->readU32(offset + 4);
-    uint32_t sampleCount = parser->readU32(offset + 8);
-
-    if (sampleSize == 0 && offset + 12 + sampleCount * 4 != size) {
-        return ERROR_MALFORMED;
-    }
-
-    parser->copyBuffer(&mSampleSizes, offset, size);
-
-    mSampleCount = sampleCount;
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::StaticTrackFragment::parseCompactSampleSizes(
-        FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 12 > size) {
-        return ERROR_MALFORMED;
-    }
-
-    if (parser->readU32(offset) != 0) {
-        return ERROR_MALFORMED;
-    }
-
-    uint32_t fieldSize = parser->readU32(offset + 4);
-
-    if (fieldSize != 4 && fieldSize != 8 && fieldSize != 16) {
-        return ERROR_MALFORMED;
-    }
-
-    uint32_t sampleCount = parser->readU32(offset + 8);
-
-    if (offset + 12 + (sampleCount * fieldSize + 4) / 8 != size) {
-        return ERROR_MALFORMED;
-    }
-
-    parser->copyBuffer(&mCompactSampleSizes, offset, size);
-
-    mSampleCount = sampleCount;
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::StaticTrackFragment::parseSampleToChunk(
-        FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 8 > size) {
-        return ERROR_MALFORMED;
-    }
-
-    if (parser->readU32(offset) != 0) {
-        return ERROR_MALFORMED;
-    }
-
-    uint32_t entryCount = parser->readU32(offset + 4);
-
-    if (entryCount == 0) {
-        return OK;
-    }
-
-    if (offset + 8 + entryCount * 12 != size) {
-        return ERROR_MALFORMED;
-    }
-
-    parser->copyBuffer(&mSampleToChunk, offset, size);
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::StaticTrackFragment::parseChunkOffsets(
-        FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 8 > size) {
-        return ERROR_MALFORMED;
-    }
-
-    if (parser->readU32(offset) != 0) {
-        return ERROR_MALFORMED;
-    }
-
-    uint32_t entryCount = parser->readU32(offset + 4);
-
-    if (offset + 8 + entryCount * 4 != size) {
-        return ERROR_MALFORMED;
-    }
-
-    parser->copyBuffer(&mChunkOffsets, offset, size);
-
-    return OK;
-}
-
-status_t FragmentedMP4Parser::StaticTrackFragment::parseChunkOffsets64(
-        FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size) {
-    if (offset + 8 > size) {
-        return ERROR_MALFORMED;
-    }
-
-    if (parser->readU32(offset) != 0) {
-        return ERROR_MALFORMED;
-    }
-
-    uint32_t entryCount = parser->readU32(offset + 4);
-
-    if (offset + 8 + entryCount * 8 != size) {
-        return ERROR_MALFORMED;
-    }
-
-    parser->copyBuffer(&mChunkOffsets64, offset, size);
-
-    return OK;
-}
-
-}  // namespace android
-

media/libstagefright/mp4/TrackFragment.h

diff --git a/media/libstagefright/mp4/TrackFragment.h b/media/libstagefright/mp4/TrackFragment.h
deleted file mode 100644
index e1ad46e..0000000
--- a/media/libstagefright/mp4/TrackFragment.h
+++ /dev/null
@@ -1,122 +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 TRACK_FRAGMENT_H_
-
-#define TRACK_FRAGMENT_H_
-
-#include "include/FragmentedMP4Parser.h"
-
-namespace android {
-
-struct FragmentedMP4Parser::TrackFragment : public RefBase {
-    TrackFragment() {}
-
-    virtual status_t getSample(SampleInfo *info) = 0;
-    virtual void advance() = 0;
-
-    virtual status_t signalCompletion() = 0;
-    virtual bool complete() const = 0;
-
-protected:
-    virtual ~TrackFragment() {}
-
-private:
-    DISALLOW_EVIL_CONSTRUCTORS(TrackFragment);
-};
-
-struct FragmentedMP4Parser::DynamicTrackFragment : public FragmentedMP4Parser::TrackFragment {
-    DynamicTrackFragment();
-
-    virtual status_t getSample(SampleInfo *info);
-    virtual void advance();
-
-    void addSample(
-            off64_t dataOffset, size_t sampleSize,
-            uint32_t presentationTime,
-            size_t sampleDescIndex,
-            uint32_t flags);
-
-    // No more samples will be added to this fragment.
-    virtual status_t signalCompletion();
-
-    virtual bool complete() const;
-
-protected:
-    virtual ~DynamicTrackFragment();
-
-private:
-    bool mComplete;
-    size_t mSampleIndex;
-    Vector<SampleInfo> mSamples;
-
-    DISALLOW_EVIL_CONSTRUCTORS(DynamicTrackFragment);
-};
-
-struct FragmentedMP4Parser::StaticTrackFragment : public FragmentedMP4Parser::TrackFragment {
-    StaticTrackFragment();
-
-    virtual status_t getSample(SampleInfo *info);
-    virtual void advance();
-
-    virtual status_t signalCompletion();
-    virtual bool complete() const;
-
-    status_t parseSampleSizes(
-            FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseCompactSampleSizes(
-            FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseSampleToChunk(
-            FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseChunkOffsets(
-            FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size);
-
-    status_t parseChunkOffsets64(
-            FragmentedMP4Parser *parser, uint32_t type, size_t offset, uint64_t size);
-
-protected:
-    virtual ~StaticTrackFragment();
-
-private:
-    size_t mSampleIndex;
-    size_t mSampleCount;
-    uint32_t mChunkIndex;
-
-    SampleInfo mSampleInfo;
-
-    sp<ABuffer> mSampleSizes;
-    sp<ABuffer> mCompactSampleSizes;
-
-    sp<ABuffer> mSampleToChunk;
-    ssize_t mSampleToChunkIndex;
-    size_t mSampleToChunkRemaining;
-
-    sp<ABuffer> mChunkOffsets;
-    sp<ABuffer> mChunkOffsets64;
-    uint32_t mPrevChunkIndex;
-    uint64_t mNextSampleOffset;
-
-    void updateSampleInfo();
-
-    DISALLOW_EVIL_CONSTRUCTORS(StaticTrackFragment);
-};
-
-}  // namespace android
-
-#endif  // TRACK_FRAGMENT_H_

media/libstagefright/mpeg2ts/ATSParser.cpp

diff --git a/media/libstagefright/mpeg2ts/ATSParser.cpp b/media/libstagefright/mpeg2ts/ATSParser.cpp
index 9850a46..d039f7d 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.cpp
+++ b/media/libstagefright/mpeg2ts/ATSParser.cpp
@@ -506,6 +506,11 @@
                     ElementaryStreamQueue::PCM_AUDIO);
             break;
 
+        case STREAMTYPE_AC3:
+            mQueue = new ElementaryStreamQueue(
+                    ElementaryStreamQueue::AC3);
+            break;
+
         default:
             break;
     }
@@ -614,6 +619,7 @@
         case STREAMTYPE_MPEG2_AUDIO:
         case STREAMTYPE_MPEG2_AUDIO_ADTS:
         case STREAMTYPE_PCM_AUDIO:
+        case STREAMTYPE_AC3:
             return true;
 
         default:
@@ -856,7 +862,7 @@
 }
 
 void ATSParser::Stream::onPayloadData(
-        unsigned PTS_DTS_flags, uint64_t PTS, uint64_t DTS,
+        unsigned PTS_DTS_flags, uint64_t PTS, uint64_t /* DTS */,
         const uint8_t *data, size_t size) {
 #if 0
     ALOGI("payload streamType 0x%02x, PTS = 0x%016llx, dPTS = %lld",
@@ -1193,7 +1199,10 @@
     unsigned sync_byte = br->getBits(8);
     CHECK_EQ(sync_byte, 0x47u);
 
-    MY_LOGV("transport_error_indicator = %u", br->getBits(1));
+    if (br->getBits(1)) {  // transport_error_indicator
+        // silently ignore.
+        return OK;
+    }
 
     unsigned payload_unit_start_indicator = br->getBits(1);
     ALOGV("payload_unit_start_indicator = %u", payload_unit_start_indicator);
@@ -1258,7 +1267,7 @@
 }
 
 void ATSParser::updatePCR(
-        unsigned PID, uint64_t PCR, size_t byteOffsetFromStart) {
+        unsigned /* PID */, uint64_t PCR, size_t byteOffsetFromStart) {
     ALOGV("PCR 0x%016llx @ %d", PCR, byteOffsetFromStart);
 
     if (mNumPCRs == 2) {

media/libstagefright/mpeg2ts/ATSParser.h

diff --git a/media/libstagefright/mpeg2ts/ATSParser.h b/media/libstagefright/mpeg2ts/ATSParser.h
index a10edc9..d4e30b4 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.h
+++ b/media/libstagefright/mpeg2ts/ATSParser.h
@@ -88,6 +88,10 @@
         STREAMTYPE_MPEG2_AUDIO_ADTS     = 0x0f,
         STREAMTYPE_MPEG4_VIDEO          = 0x10,
         STREAMTYPE_H264                 = 0x1b,
+
+        // From ATSC A/53 Part 3:2009, 6.7.1
+        STREAMTYPE_AC3                  = 0x81,
+
         STREAMTYPE_PCM_AUDIO            = 0x83,
     };
 

media/libstagefright/mpeg2ts/Android.mk

diff --git a/media/libstagefright/mpeg2ts/Android.mk b/media/libstagefright/mpeg2ts/Android.mk
index c1a7a9d..c17a0b7 100644
--- a/media/libstagefright/mpeg2ts/Android.mk
+++ b/media/libstagefright/mpeg2ts/Android.mk
@@ -13,6 +13,8 @@
 	$(TOP)/frameworks/av/media/libstagefright \
 	$(TOP)/frameworks/native/include/media/openmax
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_MODULE:= libstagefright_mpeg2ts
 
 ifeq ($(TARGET_ARCH),arm)

media/libstagefright/mpeg2ts/AnotherPacketSource.cpp

diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
index 3153c8b..6dfaa94 100644
--- a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
+++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
@@ -34,7 +34,8 @@
     : mIsAudio(false),
       mFormat(NULL),
       mLastQueuedTimeUs(0),
-      mEOSResult(OK) {
+      mEOSResult(OK),
+      mLatestEnqueuedMeta(NULL) {
     setFormat(meta);
 }
 
@@ -61,7 +62,7 @@
 AnotherPacketSource::~AnotherPacketSource() {
 }
 
-status_t AnotherPacketSource::start(MetaData *params) {
+status_t AnotherPacketSource::start(MetaData * /* params */) {
     return OK;
 }
 
@@ -70,7 +71,27 @@
 }
 
 sp<MetaData> AnotherPacketSource::getFormat() {
-    return mFormat;
+    Mutex::Autolock autoLock(mLock);
+    if (mFormat != NULL) {
+        return mFormat;
+    }
+
+    List<sp<ABuffer> >::iterator it = mBuffers.begin();
+    while (it != mBuffers.end()) {
+        sp<ABuffer> buffer = *it;
+        int32_t discontinuity;
+        if (buffer->meta()->findInt32("discontinuity", &discontinuity)) {
+            break;
+        }
+
+        sp<RefBase> object;
+        if (buffer->meta()->findObject("format", &object)) {
+            return static_cast<MetaData*>(object.get());
+        }
+
+        ++it;
+    }
+    return NULL;
 }
 
 status_t AnotherPacketSource::dequeueAccessUnit(sp<ABuffer> *buffer) {
@@ -94,6 +115,11 @@
             return INFO_DISCONTINUITY;
         }
 
+        sp<RefBase> object;
+        if ((*buffer)->meta()->findObject("format", &object)) {
+            mFormat = static_cast<MetaData*>(object.get());
+        }
+
         return OK;
     }
 
@@ -120,17 +146,22 @@
             }
 
             return INFO_DISCONTINUITY;
-        } else {
-            int64_t timeUs;
-            CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
-
-            MediaBuffer *mediaBuffer = new MediaBuffer(buffer);
-
-            mediaBuffer->meta_data()->setInt64(kKeyTime, timeUs);
-
-            *out = mediaBuffer;
-            return OK;
         }
+
+        sp<RefBase> object;
+        if (buffer->meta()->findObject("format", &object)) {
+            mFormat = static_cast<MetaData*>(object.get());
+        }
+
+        int64_t timeUs;
+        CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
+
+        MediaBuffer *mediaBuffer = new MediaBuffer(buffer);
+
+        mediaBuffer->meta_data()->setInt64(kKeyTime, timeUs);
+
+        *out = mediaBuffer;
+        return OK;
     }
 
     return mEOSResult;
@@ -152,12 +183,24 @@
         return;
     }
 
-    CHECK(buffer->meta()->findInt64("timeUs", &mLastQueuedTimeUs));
+    int64_t lastQueuedTimeUs;
+    CHECK(buffer->meta()->findInt64("timeUs", &lastQueuedTimeUs));
+    mLastQueuedTimeUs = lastQueuedTimeUs;
     ALOGV("queueAccessUnit timeUs=%lld us (%.2f secs)", mLastQueuedTimeUs, mLastQueuedTimeUs / 1E6);
 
     Mutex::Autolock autoLock(mLock);
     mBuffers.push_back(buffer);
     mCondition.signal();
+
+    if (!mLatestEnqueuedMeta.get()) {
+        mLatestEnqueuedMeta = buffer->meta();
+    } else {
+        int64_t latestTimeUs = 0;
+        CHECK(mLatestEnqueuedMeta->findInt64("timeUs", &latestTimeUs));
+        if (lastQueuedTimeUs > latestTimeUs) {
+            mLatestEnqueuedMeta = buffer->meta();
+        }
+    }
 }
 
 void AnotherPacketSource::clear() {
@@ -167,6 +210,7 @@
     mEOSResult = OK;
 
     mFormat = NULL;
+    mLatestEnqueuedMeta = NULL;
 }
 
 void AnotherPacketSource::queueDiscontinuity(
@@ -191,6 +235,7 @@
 
     mEOSResult = OK;
     mLastQueuedTimeUs = 0;
+    mLatestEnqueuedMeta = NULL;
 
     sp<ABuffer> buffer = new ABuffer(0);
     buffer->meta()->setInt32("discontinuity", static_cast<int32_t>(type));
@@ -278,4 +323,9 @@
     return (mEOSResult != OK);
 }
 
+sp<AMessage> AnotherPacketSource::getLatestMeta() {
+    Mutex::Autolock autoLock(mLock);
+    return mLatestEnqueuedMeta;
+}
+
 }  // namespace android

media/libstagefright/mpeg2ts/AnotherPacketSource.h

diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.h b/media/libstagefright/mpeg2ts/AnotherPacketSource.h
index e16cf78..9b193a2 100644
--- a/media/libstagefright/mpeg2ts/AnotherPacketSource.h
+++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.h
@@ -62,6 +62,8 @@
 
     bool isFinished(int64_t duration) const;
 
+    sp<AMessage> getLatestMeta();
+
 protected:
     virtual ~AnotherPacketSource();
 
@@ -74,6 +76,7 @@
     int64_t mLastQueuedTimeUs;
     List<sp<ABuffer> > mBuffers;
     status_t mEOSResult;
+    sp<AMessage> mLatestEnqueuedMeta;
 
     bool wasFormatChange(int32_t discontinuityType) const;
 

media/libstagefright/mpeg2ts/ESQueue.cpp

diff --git a/media/libstagefright/mpeg2ts/ESQueue.cpp b/media/libstagefright/mpeg2ts/ESQueue.cpp
index 8f9c9c8..c0c9717 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.cpp
+++ b/media/libstagefright/mpeg2ts/ESQueue.cpp
@@ -56,6 +56,122 @@
     }
 }
 
+// Parse AC3 header assuming the current ptr is start position of syncframe,
+// update metadata only applicable, and return the payload size
+static unsigned parseAC3SyncFrame(
+        const uint8_t *ptr, size_t size, sp<MetaData> *metaData) {
+    static const unsigned channelCountTable[] = {2, 1, 2, 3, 3, 4, 4, 5};
+    static const unsigned samplingRateTable[] = {48000, 44100, 32000};
+    static const unsigned rates[] = {32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256,
+            320, 384, 448, 512, 576, 640};
+
+    static const unsigned frameSizeTable[19][3] = {
+        { 64, 69, 96 },
+        { 80, 87, 120 },
+        { 96, 104, 144 },
+        { 112, 121, 168 },
+        { 128, 139, 192 },
+        { 160, 174, 240 },
+        { 192, 208, 288 },
+        { 224, 243, 336 },
+        { 256, 278, 384 },
+        { 320, 348, 480 },
+        { 384, 417, 576 },
+        { 448, 487, 672 },
+        { 512, 557, 768 },
+        { 640, 696, 960 },
+        { 768, 835, 1152 },
+        { 896, 975, 1344 },
+        { 1024, 1114, 1536 },
+        { 1152, 1253, 1728 },
+        { 1280, 1393, 1920 },
+    };
+
+    ABitReader bits(ptr, size);
+    unsigned syncStartPos = 0;  // in bytes
+    if (bits.numBitsLeft() < 16) {
+        return 0;
+    }
+    if (bits.getBits(16) != 0x0B77) {
+        return 0;
+    }
+
+    if (bits.numBitsLeft() < 16 + 2 + 6 + 5 + 3 + 3) {
+        ALOGV("Not enough bits left for further parsing");
+        return 0;
+    }
+    bits.skipBits(16);  // crc1
+
+    unsigned fscod = bits.getBits(2);
+    if (fscod == 3) {
+        ALOGW("Incorrect fscod in AC3 header");
+        return 0;
+    }
+
+    unsigned frmsizecod = bits.getBits(6);
+    if (frmsizecod > 37) {
+        ALOGW("Incorrect frmsizecod in AC3 header");
+        return 0;
+    }
+
+    unsigned bsid = bits.getBits(5);
+    if (bsid > 8) {
+        ALOGW("Incorrect bsid in AC3 header. Possibly E-AC-3?");
+        return 0;
+    }
+
+    unsigned bsmod = bits.getBits(3);
+    unsigned acmod = bits.getBits(3);
+    unsigned cmixlev = 0;
+    unsigned surmixlev = 0;
+    unsigned dsurmod = 0;
+
+    if ((acmod & 1) > 0 && acmod != 1) {
+        if (bits.numBitsLeft() < 2) {
+            return 0;
+        }
+        cmixlev = bits.getBits(2);
+    }
+    if ((acmod & 4) > 0) {
+        if (bits.numBitsLeft() < 2) {
+            return 0;
+        }
+        surmixlev = bits.getBits(2);
+    }
+    if (acmod == 2) {
+        if (bits.numBitsLeft() < 2) {
+            return 0;
+        }
+        dsurmod = bits.getBits(2);
+    }
+
+    if (bits.numBitsLeft() < 1) {
+        return 0;
+    }
+    unsigned lfeon = bits.getBits(1);
+
+    unsigned samplingRate = samplingRateTable[fscod];
+    unsigned payloadSize = frameSizeTable[frmsizecod >> 1][fscod];
+    if (fscod == 1) {
+        payloadSize += frmsizecod & 1;
+    }
+    payloadSize <<= 1;  // convert from 16-bit words to bytes
+
+    unsigned channelCount = channelCountTable[acmod] + lfeon;
+
+    if (metaData != NULL) {
+        (*metaData)->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AC3);
+        (*metaData)->setInt32(kKeyChannelCount, channelCount);
+        (*metaData)->setInt32(kKeySampleRate, samplingRate);
+    }
+
+    return payloadSize;
+}
+
+static bool IsSeeminglyValidAC3Header(const uint8_t *ptr, size_t size) {
+    return parseAC3SyncFrame(ptr, size, NULL) > 0;
+}
+
 static bool IsSeeminglyValidADTSHeader(const uint8_t *ptr, size_t size) {
     if (size < 3) {
         // Not enough data to verify header.
@@ -224,6 +340,33 @@
                 break;
             }
 
+            case AC3:
+            {
+                uint8_t *ptr = (uint8_t *)data;
+
+                ssize_t startOffset = -1;
+                for (size_t i = 0; i < size; ++i) {
+                    if (IsSeeminglyValidAC3Header(&ptr[i], size - i)) {
+                        startOffset = i;
+                        break;
+                    }
+                }
+
+                if (startOffset < 0) {
+                    return ERROR_MALFORMED;
+                }
+
+                if (startOffset > 0) {
+                    ALOGI("found something resembling an AC3 syncword at "
+                          "offset %d",
+                          startOffset);
+                }
+
+                data = &ptr[startOffset];
+                size -= startOffset;
+                break;
+            }
+
             case MPEG_AUDIO:
             {
                 uint8_t *ptr = (uint8_t *)data;
@@ -328,6 +471,8 @@
             return dequeueAccessUnitH264();
         case AAC:
             return dequeueAccessUnitAAC();
+        case AC3:
+            return dequeueAccessUnitAC3();
         case MPEG_VIDEO:
             return dequeueAccessUnitMPEGVideo();
         case MPEG4_VIDEO:
@@ -340,6 +485,51 @@
     }
 }
 
+sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitAC3() {
+    unsigned syncStartPos = 0;  // in bytes
+    unsigned payloadSize = 0;
+    sp<MetaData> format = new MetaData;
+    while (true) {
+        if (syncStartPos + 2 >= mBuffer->size()) {
+            return NULL;
+        }
+
+        payloadSize = parseAC3SyncFrame(
+                mBuffer->data() + syncStartPos,
+                mBuffer->size() - syncStartPos,
+                &format);
+        if (payloadSize > 0) {
+            break;
+        }
+        ++syncStartPos;
+    }
+
+    if (mBuffer->size() < syncStartPos + payloadSize) {
+        ALOGV("Not enough buffer size for AC3");
+        return NULL;
+    }
+
+    if (mFormat == NULL) {
+        mFormat = format;
+    }
+
+    sp<ABuffer> accessUnit = new ABuffer(syncStartPos + payloadSize);
+    memcpy(accessUnit->data(), mBuffer->data(), syncStartPos + payloadSize);
+
+    int64_t timeUs = fetchTimestamp(syncStartPos + payloadSize);
+    CHECK_GE(timeUs, 0ll);
+    accessUnit->meta()->setInt64("timeUs", timeUs);
+
+    memmove(
+            mBuffer->data(),
+            mBuffer->data() + syncStartPos + payloadSize,
+            mBuffer->size() - syncStartPos - payloadSize);
+
+    mBuffer->setRange(0, mBuffer->size() - syncStartPos - payloadSize);
+
+    return accessUnit;
+}
+
 sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitPCMAudio() {
     if (mBuffer->size() < 4) {
         return NULL;
@@ -604,7 +794,9 @@
                 dstOffset += pos.nalSize + 4;
             }
 
+#if !LOG_NDEBUG
             ALOGV("accessUnit contains nal types %s", out.c_str());
+#endif
 
             const NALPosition &pos = nals.itemAt(nals.size() - 1);
             size_t nextScan = pos.nalOffset + pos.nalSize;

media/libstagefright/mpeg2ts/ESQueue.h

diff --git a/media/libstagefright/mpeg2ts/ESQueue.h b/media/libstagefright/mpeg2ts/ESQueue.h
index 66a8087..a2cca77 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.h
+++ b/media/libstagefright/mpeg2ts/ESQueue.h
@@ -32,6 +32,7 @@
     enum Mode {
         H264,
         AAC,
+        AC3,
         MPEG_AUDIO,
         MPEG_VIDEO,
         MPEG4_VIDEO,
@@ -67,6 +68,7 @@
 
     sp<ABuffer> dequeueAccessUnitH264();
     sp<ABuffer> dequeueAccessUnitAAC();
+    sp<ABuffer> dequeueAccessUnitAC3();
     sp<ABuffer> dequeueAccessUnitMPEGAudio();
     sp<ABuffer> dequeueAccessUnitMPEGVideo();
     sp<ABuffer> dequeueAccessUnitMPEG4Video();

media/libstagefright/mpeg2ts/MPEG2PSExtractor.cpp

diff --git a/media/libstagefright/mpeg2ts/MPEG2PSExtractor.cpp b/media/libstagefright/mpeg2ts/MPEG2PSExtractor.cpp
index dd714c99..bc2a16d 100644
--- a/media/libstagefright/mpeg2ts/MPEG2PSExtractor.cpp
+++ b/media/libstagefright/mpeg2ts/MPEG2PSExtractor.cpp
@@ -130,7 +130,8 @@
     return new WrappedTrack(this, mTracks.valueAt(index));
 }
 
-sp<MetaData> MPEG2PSExtractor::getTrackMetaData(size_t index, uint32_t flags) {
+sp<MetaData> MPEG2PSExtractor::getTrackMetaData(
+        size_t index, uint32_t /* flags */) {
     if (index >= mTracks.size()) {
         return NULL;
     }
@@ -625,7 +626,7 @@
 
 status_t MPEG2PSExtractor::Track::appendPESData(
         unsigned PTS_DTS_flags,
-        uint64_t PTS, uint64_t DTS,
+        uint64_t PTS, uint64_t /* DTS */,
         const uint8_t *data, size_t size) {
     if (mQueue == NULL) {
         return OK;

media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp

diff --git a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
index d449c34..35ca118 100644
--- a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
+++ b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
@@ -141,7 +141,7 @@
 }
 
 sp<MetaData> MPEG2TSExtractor::getTrackMetaData(
-        size_t index, uint32_t flags) {
+        size_t index, uint32_t /* flags */) {
     return index < mSourceImpls.size()
         ? mSourceImpls.editItemAt(index)->getFormat() : NULL;
 }

media/libstagefright/omx/GraphicBufferSource.cpp

diff --git a/media/libstagefright/omx/GraphicBufferSource.cpp b/media/libstagefright/omx/GraphicBufferSource.cpp
index cf43e94..b81b116 100644
--- a/media/libstagefright/omx/GraphicBufferSource.cpp
+++ b/media/libstagefright/omx/GraphicBufferSource.cpp
@@ -41,12 +41,21 @@
     mNumFramesAvailable(0),
     mEndOfStream(false),
     mEndOfStreamSent(false),
+    mMaxTimestampGapUs(-1ll),
+    mPrevOriginalTimeUs(-1ll),
+    mPrevModifiedTimeUs(-1ll),
+    mSkipFramesBeforeNs(-1ll),
     mRepeatAfterUs(-1ll),
     mRepeatLastFrameGeneration(0),
+    mRepeatLastFrameTimestamp(-1ll),
     mLatestSubmittedBufferId(-1),
     mLatestSubmittedBufferFrameNum(0),
     mLatestSubmittedBufferUseCount(0),
-    mRepeatBufferDeferred(false) {
+    mRepeatBufferDeferred(false),
+    mTimePerCaptureUs(-1ll),
+    mTimePerFrameUs(-1ll),
+    mPrevCaptureUs(-1ll),
+    mPrevFrameUs(-1ll) {
 
     ALOGV("GraphicBufferSource w=%u h=%u c=%u",
             bufferWidth, bufferHeight, bufferCount);
@@ -148,6 +157,18 @@
     }
 }
 
+void GraphicBufferSource::omxIdle() {
+    ALOGV("omxIdle");
+
+    Mutex::Autolock autoLock(mMutex);
+
+    if (mExecuting) {
+        // We are only interested in the transition from executing->idle,
+        // not loaded->idle.
+        mExecuting = false;
+    }
+}
+
 void GraphicBufferSource::omxLoaded(){
     Mutex::Autolock autoLock(mMutex);
     if (!mExecuting) {
@@ -194,7 +215,9 @@
 void GraphicBufferSource::codecBufferEmptied(OMX_BUFFERHEADERTYPE* header) {
     Mutex::Autolock autoLock(mMutex);
 
-    CHECK(mExecuting);  // could this happen if app stop()s early?
+    if (!mExecuting) {
+        return;
+    }
 
     int cbi = findMatchingCodecBuffer_l(header);
     if (cbi < 0) {
@@ -213,7 +236,12 @@
     // see if the GraphicBuffer reference was null, which should only ever
     // happen for EOS.
     if (codecBuffer.mGraphicBuffer == NULL) {
-        CHECK(mEndOfStream && mEndOfStreamSent);
+        if (!(mEndOfStream && mEndOfStreamSent)) {
+            // This can happen when broken code sends us the same buffer
+            // twice in a row.
+            ALOGE("ERROR: codecBufferEmptied on non-EOS null buffer "
+                    "(buffer emptied twice?)");
+        }
         // No GraphicBuffer to deal with, no additional input or output is
         // expected, so just return.
         return;
@@ -280,6 +308,32 @@
     return;
 }
 
+void GraphicBufferSource::codecBufferFilled(OMX_BUFFERHEADERTYPE* header) {
+    Mutex::Autolock autoLock(mMutex);
+
+    if (mMaxTimestampGapUs > 0ll
+            && !(header->nFlags & OMX_BUFFERFLAG_CODECCONFIG)) {
+        ssize_t index = mOriginalTimeUs.indexOfKey(header->nTimeStamp);
+        if (index >= 0) {
+            ALOGV("OUT timestamp: %lld -> %lld",
+                    header->nTimeStamp, mOriginalTimeUs[index]);
+            header->nTimeStamp = mOriginalTimeUs[index];
+            mOriginalTimeUs.removeItemsAt(index);
+        } else {
+            // giving up the effort as encoder doesn't appear to preserve pts
+            ALOGW("giving up limiting timestamp gap (pts = %lld)",
+                    header->nTimeStamp);
+            mMaxTimestampGapUs = -1ll;
+        }
+        if (mOriginalTimeUs.size() > BufferQueue::NUM_BUFFER_SLOTS) {
+            // something terribly wrong must have happened, giving up...
+            ALOGE("mOriginalTimeUs has too many entries (%d)",
+                    mOriginalTimeUs.size());
+            mMaxTimestampGapUs = -1ll;
+        }
+    }
+}
+
 void GraphicBufferSource::suspend(bool suspend) {
     Mutex::Autolock autoLock(mMutex);
 
@@ -365,7 +419,18 @@
         mBufferSlot[item.mBuf] = item.mGraphicBuffer;
     }
 
-    err = submitBuffer_l(item, cbi);
+    err = UNKNOWN_ERROR;
+
+    // only submit sample if start time is unspecified, or sample
+    // is queued after the specified start time
+    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);
+    }
+
     if (err != OK) {
         ALOGV("submitBuffer_l failed, releasing bq buf %d", item.mBuf);
         mBufferQueue->releaseBuffer(item.mBuf, item.mFrameNumber,
@@ -384,6 +449,23 @@
     if (mLatestSubmittedBufferId < 0 || mSuspended) {
         return false;
     }
+    if (mBufferSlot[mLatestSubmittedBufferId] == 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");
+        mBufferQueue->releaseBuffer(
+                mLatestSubmittedBufferId,
+                mLatestSubmittedBufferFrameNum,
+                EGL_NO_DISPLAY,
+                EGL_NO_SYNC_KHR,
+                Fence::NO_FENCE);
+        mLatestSubmittedBufferId = -1;
+        mLatestSubmittedBufferFrameNum = 0;
+        return false;
+    }
 
     int cbi = findAvailableCodecBuffer_l();
     if (cbi < 0) {
@@ -395,6 +477,7 @@
     BufferQueue::BufferItem item;
     item.mBuf = mLatestSubmittedBufferId;
     item.mFrameNumber = mLatestSubmittedBufferFrameNum;
+    item.mTimestamp = mRepeatLastFrameTimestamp;
 
     status_t err = submitBuffer_l(item, cbi);
 
@@ -404,6 +487,20 @@
 
     ++mLatestSubmittedBufferUseCount;
 
+    /* repeat last frame up to kRepeatLastFrameCount times.
+     * in case of static scene, a single repeat might not get rid of encoder
+     * ghosting completely, refresh a couple more times to get better quality
+     */
+    if (--mRepeatLastFrameCount > 0) {
+        mRepeatLastFrameTimestamp = item.mTimestamp + mRepeatAfterUs * 1000;
+
+        if (mReflector != NULL) {
+            sp<AMessage> msg = new AMessage(kWhatRepeatLastFrame, mReflector->id());
+            msg->setInt32("generation", ++mRepeatLastFrameGeneration);
+            msg->post(mRepeatAfterUs);
+        }
+    }
+
     return true;
 }
 
@@ -424,8 +521,11 @@
 
     mLatestSubmittedBufferId = item.mBuf;
     mLatestSubmittedBufferFrameNum = item.mFrameNumber;
+    mRepeatLastFrameTimestamp = item.mTimestamp + mRepeatAfterUs * 1000;
+
     mLatestSubmittedBufferUseCount = 1;
     mRepeatBufferDeferred = false;
+    mRepeatLastFrameCount = kRepeatLastFrameCount;
 
     if (mReflector != NULL) {
         sp<AMessage> msg = new AMessage(kWhatRepeatLastFrame, mReflector->id());
@@ -461,9 +561,71 @@
     return OK;
 }
 
+int64_t GraphicBufferSource::getTimestamp(const BufferQueue::BufferItem &item) {
+    int64_t timeUs = item.mTimestamp / 1000;
+
+    if (mTimePerCaptureUs > 0ll) {
+        // Time lapse or slow motion mode
+        if (mPrevCaptureUs < 0ll) {
+            // first capture
+            mPrevCaptureUs = timeUs;
+            mPrevFrameUs = timeUs;
+        } else {
+            // snap to nearest capture point
+            int64_t nFrames = (timeUs + mTimePerCaptureUs / 2 - mPrevCaptureUs)
+                    / mTimePerCaptureUs;
+            if (nFrames <= 0) {
+                // skip this frame as it's too close to previous capture
+                ALOGV("skipping frame, timeUs %lld", timeUs);
+                return -1;
+            }
+            mPrevCaptureUs = mPrevCaptureUs + nFrames * mTimePerCaptureUs;
+            mPrevFrameUs += mTimePerFrameUs * nFrames;
+        }
+
+        ALOGV("timeUs %lld, captureUs %lld, frameUs %lld",
+                timeUs, mPrevCaptureUs, mPrevFrameUs);
+
+        return mPrevFrameUs;
+    } else if (mMaxTimestampGapUs > 0ll) {
+        /* Cap timestamp gap between adjacent frames to specified max
+         *
+         * In the scenario of cast mirroring, encoding could be suspended for
+         * prolonged periods. Limiting the pts gap to workaround the problem
+         * where encoder's rate control logic produces huge frames after a
+         * long period of suspension.
+         */
+
+        int64_t originalTimeUs = timeUs;
+        if (mPrevOriginalTimeUs >= 0ll) {
+            if (originalTimeUs < mPrevOriginalTimeUs) {
+                // Drop the frame if it's going backward in time. Bad timestamp
+                // could disrupt encoder's rate control completely.
+                ALOGW("Dropping frame that's going backward in time");
+                return -1;
+            }
+            int64_t timestampGapUs = originalTimeUs - mPrevOriginalTimeUs;
+            timeUs = (timestampGapUs < mMaxTimestampGapUs ?
+                    timestampGapUs : mMaxTimestampGapUs) + mPrevModifiedTimeUs;
+        }
+        mPrevOriginalTimeUs = originalTimeUs;
+        mPrevModifiedTimeUs = timeUs;
+        mOriginalTimeUs.add(timeUs, originalTimeUs);
+        ALOGV("IN  timestamp: %lld -> %lld", originalTimeUs, timeUs);
+    }
+
+    return timeUs;
+}
+
 status_t GraphicBufferSource::submitBuffer_l(
         const BufferQueue::BufferItem &item, int cbi) {
     ALOGV("submitBuffer_l cbi=%d", cbi);
+
+    int64_t timeUs = getTimestamp(item);
+    if (timeUs < 0ll) {
+        return UNKNOWN_ERROR;
+    }
+
     CodecBuffer& codecBuffer(mCodecBuffers.editItemAt(cbi));
     codecBuffer.mGraphicBuffer = mBufferSlot[item.mBuf];
     codecBuffer.mBuf = item.mBuf;
@@ -479,7 +641,7 @@
 
     status_t err = mNodeInstance->emptyDirectBuffer(header, 0,
             4 + sizeof(buffer_handle_t), OMX_BUFFERFLAG_ENDOFFRAME,
-            item.mTimestamp / 1000);
+            timeUs);
     if (err != OK) {
         ALOGW("WARNING: emptyDirectBuffer failed: 0x%x", err);
         codecBuffer.mGraphicBuffer = NULL;
@@ -573,6 +735,12 @@
         BufferQueue::BufferItem item;
         status_t err = mBufferQueue->acquireBuffer(&item, 0);
         if (err == OK) {
+            // If this is the first time we're seeing this buffer, add it to our
+            // slot table.
+            if (item.mGraphicBuffer != NULL) {
+                ALOGV("onFrameAvailable: setting mBufferSlot %d", item.mBuf);
+                mBufferSlot[item.mBuf] = item.mGraphicBuffer;
+            }
             mBufferQueue->releaseBuffer(item.mBuf, item.mFrameNumber,
                     EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, item.mFence);
         }
@@ -622,6 +790,38 @@
     return OK;
 }
 
+status_t GraphicBufferSource::setMaxTimestampGapUs(int64_t maxGapUs) {
+    Mutex::Autolock autoLock(mMutex);
+
+    if (mExecuting || maxGapUs <= 0ll) {
+        return INVALID_OPERATION;
+    }
+
+    mMaxTimestampGapUs = maxGapUs;
+
+    return OK;
+}
+
+void GraphicBufferSource::setSkipFramesBeforeUs(int64_t skipFramesBeforeUs) {
+    Mutex::Autolock autoLock(mMutex);
+
+    mSkipFramesBeforeNs =
+            (skipFramesBeforeUs > 0) ? (skipFramesBeforeUs * 1000) : -1ll;
+}
+
+status_t GraphicBufferSource::setTimeLapseUs(int64_t* data) {
+    Mutex::Autolock autoLock(mMutex);
+
+    if (mExecuting || data[0] <= 0ll || data[1] <= 0ll) {
+        return INVALID_OPERATION;
+    }
+
+    mTimePerFrameUs = data[0];
+    mTimePerCaptureUs = data[1];
+
+    return OK;
+}
+
 void GraphicBufferSource::onMessageReceived(const sp<AMessage> &msg) {
     switch (msg->what()) {
         case kWhatRepeatLastFrame:

media/libstagefright/omx/GraphicBufferSource.h

diff --git a/media/libstagefright/omx/GraphicBufferSource.h b/media/libstagefright/omx/GraphicBufferSource.h
index 244a843..fba42b7 100644
--- a/media/libstagefright/omx/GraphicBufferSource.h
+++ b/media/libstagefright/omx/GraphicBufferSource.h
@@ -69,6 +69,11 @@
     // sitting in the BufferQueue, this will send them to the codec.
     void omxExecuting();
 
+    // This is called when OMX transitions to OMX_StateIdle, indicating that
+    // the codec is meant to return all buffers back to the client for them
+    // to be freed. Do NOT submit any more buffers to the component.
+    void omxIdle();
+
     // This is called when OMX transitions to OMX_StateLoaded, indicating that
     // we are shutting down.
     void omxLoaded();
@@ -82,6 +87,10 @@
     // fill it with a new frame of data; otherwise, just mark it as available.
     void codecBufferEmptied(OMX_BUFFERHEADERTYPE* header);
 
+    // Called when omx_message::FILL_BUFFER_DONE is received. (Currently the
+    // buffer source will fix timestamp in the header if needed.)
+    void codecBufferFilled(OMX_BUFFERHEADERTYPE* header);
+
     // This is called after the last input frame has been submitted.  We
     // need to submit an empty buffer with the EOS flag set.  If we don't
     // have a codec buffer ready, we just set the mEndOfStream flag.
@@ -100,6 +109,26 @@
     // state and once this behaviour is specified it cannot be reset.
     status_t setRepeatPreviousFrameDelayUs(int64_t repeatAfterUs);
 
+    // When set, the timestamp fed to the encoder will be modified such that
+    // the gap between two adjacent frames is capped at maxGapUs. Timestamp
+    // will be restored to the original when the encoded frame is returned to
+    // the client.
+    // This is to solve a problem in certain real-time streaming case, where
+    // encoder's rate control logic produces huge frames after a long period
+    // of suspension on input.
+    status_t setMaxTimestampGapUs(int64_t maxGapUs);
+
+    // 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
+    // When set, the sample's timestamp will be modified to playback framerate,
+    // and capture timestamp will be modified to capture rate.
+    status_t setTimeLapseUs(int64_t* data);
+
+    // Sets the start time us (in system time), samples before which should
+    // be dropped and not submitted to encoder
+    void setSkipFramesBeforeUs(int64_t startTimeUs);
+
 protected:
     // BufferQueue::ConsumerListener interface, called when a new frame of
     // data is available.  If we're executing and a codec buffer is
@@ -160,6 +189,7 @@
 
     void setLatestSubmittedBuffer_l(const BufferQueue::BufferItem &item);
     bool repeatLatestSubmittedBuffer_l();
+    int64_t getTimestamp(const BufferQueue::BufferItem &item);
 
     // Lock, covers all member variables.
     mutable Mutex mMutex;
@@ -201,13 +231,23 @@
     enum {
         kWhatRepeatLastFrame,
     };
+    enum {
+        kRepeatLastFrameCount = 10,
+    };
 
-    int64_t mRepeatAfterUs;
+    KeyedVector<int64_t, int64_t> mOriginalTimeUs;
+    int64_t mMaxTimestampGapUs;
+    int64_t mPrevOriginalTimeUs;
+    int64_t mPrevModifiedTimeUs;
+    int64_t mSkipFramesBeforeNs;
 
     sp<ALooper> mLooper;
     sp<AHandlerReflector<GraphicBufferSource> > mReflector;
 
+    int64_t mRepeatAfterUs;
     int32_t mRepeatLastFrameGeneration;
+    int64_t mRepeatLastFrameTimestamp;
+    int32_t mRepeatLastFrameCount;
 
     int mLatestSubmittedBufferId;
     uint64_t mLatestSubmittedBufferFrameNum;
@@ -217,6 +257,12 @@
     // no codec buffer was available at the time.
     bool mRepeatBufferDeferred;
 
+    // Time lapse / slow motion configuration
+    int64_t mTimePerCaptureUs;
+    int64_t mTimePerFrameUs;
+    int64_t mPrevCaptureUs;
+    int64_t mPrevFrameUs;
+
     void onMessageReceived(const sp<AMessage> &msg);
 
     DISALLOW_EVIL_CONSTRUCTORS(GraphicBufferSource);

media/libstagefright/omx/OMX.cpp

diff --git a/media/libstagefright/omx/OMX.cpp b/media/libstagefright/omx/OMX.cpp
index aaa9f89..a608479 100644
--- a/media/libstagefright/omx/OMX.cpp
+++ b/media/libstagefright/omx/OMX.cpp
@@ -185,7 +185,7 @@
     instance->onObserverDied(mMaster);
 }
 
-bool OMX::livesLocally(node_id node, pid_t pid) {
+bool OMX::livesLocally(node_id /* node */, pid_t pid) {
     return pid == getpid();
 }
 
@@ -331,6 +331,13 @@
     return findInstance(node)->storeMetaDataInBuffers(port_index, enable);
 }
 
+status_t OMX::prepareForAdaptivePlayback(
+        node_id node, OMX_U32 portIndex, OMX_BOOL enable,
+        OMX_U32 maxFrameWidth, OMX_U32 maxFrameHeight) {
+    return findInstance(node)->prepareForAdaptivePlayback(
+            portIndex, enable, maxFrameWidth, maxFrameHeight);
+}
+
 status_t OMX::useBuffer(
         node_id node, OMX_U32 port_index, const sp<IMemory> &params,
         buffer_id *buffer) {
@@ -417,7 +424,7 @@
         OMX_IN OMX_EVENTTYPE eEvent,
         OMX_IN OMX_U32 nData1,
         OMX_IN OMX_U32 nData2,
-        OMX_IN OMX_PTR pEventData) {
+        OMX_IN OMX_PTR /* pEventData */) {
     ALOGV("OnEvent(%d, %ld, %ld)", eEvent, nData1, nData2);
 
     // Forward to OMXNodeInstance.
@@ -472,7 +479,7 @@
 OMX::node_id OMX::makeNodeID(OMXNodeInstance *instance) {
     // mLock is already held.
 
-    node_id node = (node_id)++mNodeCounter;
+    node_id node = (node_id)(uintptr_t)++mNodeCounter;
     mNodeIDToInstance.add(node, instance);
 
     return node;

media/libstagefright/omx/OMXNodeInstance.cpp

diff --git a/media/libstagefright/omx/OMXNodeInstance.cpp b/media/libstagefright/omx/OMXNodeInstance.cpp
index ef683a0..0fb38fa 100644
--- a/media/libstagefright/omx/OMXNodeInstance.cpp
+++ b/media/libstagefright/omx/OMXNodeInstance.cpp
@@ -243,13 +243,18 @@
 status_t OMXNodeInstance::sendCommand(
         OMX_COMMANDTYPE cmd, OMX_S32 param) {
     const sp<GraphicBufferSource>& bufferSource(getGraphicBufferSource());
-    if (bufferSource != NULL
-            && cmd == OMX_CommandStateSet
-            && param == OMX_StateLoaded) {
-        // Initiating transition from Executing -> Loaded
-        // Buffers are about to be freed.
-        bufferSource->omxLoaded();
-        setGraphicBufferSource(NULL);
+    if (bufferSource != NULL && cmd == OMX_CommandStateSet) {
+        if (param == OMX_StateIdle) {
+            // Initiating transition from Executing -> Idle
+            // ACodec is waiting for all buffers to be returned, do NOT
+            // submit any more buffers to the codec.
+            bufferSource->omxIdle();
+        } else if (param == OMX_StateLoaded) {
+            // Initiating transition from Idle/Executing -> Loaded
+            // Buffers are about to be freed.
+            bufferSource->omxLoaded();
+            setGraphicBufferSource(NULL);
+        }
 
         // fall through
     }
@@ -261,7 +266,7 @@
 }
 
 status_t OMXNodeInstance::getParameter(
-        OMX_INDEXTYPE index, void *params, size_t size) {
+        OMX_INDEXTYPE index, void *params, size_t /* size */) {
     Mutex::Autolock autoLock(mLock);
 
     OMX_ERRORTYPE err = OMX_GetParameter(mHandle, index, params);
@@ -270,7 +275,7 @@
 }
 
 status_t OMXNodeInstance::setParameter(
-        OMX_INDEXTYPE index, const void *params, size_t size) {
+        OMX_INDEXTYPE index, const void *params, size_t /* size */) {
     Mutex::Autolock autoLock(mLock);
 
     OMX_ERRORTYPE err = OMX_SetParameter(
@@ -280,7 +285,7 @@
 }
 
 status_t OMXNodeInstance::getConfig(
-        OMX_INDEXTYPE index, void *params, size_t size) {
+        OMX_INDEXTYPE index, void *params, size_t /* size */) {
     Mutex::Autolock autoLock(mLock);
 
     OMX_ERRORTYPE err = OMX_GetConfig(mHandle, index, params);
@@ -288,7 +293,7 @@
 }
 
 status_t OMXNodeInstance::setConfig(
-        OMX_INDEXTYPE index, const void *params, size_t size) {
+        OMX_INDEXTYPE index, const void *params, size_t /* size */) {
     Mutex::Autolock autoLock(mLock);
 
     OMX_ERRORTYPE err = OMX_SetConfig(
@@ -417,6 +422,40 @@
     return err;
 }
 
+status_t OMXNodeInstance::prepareForAdaptivePlayback(
+        OMX_U32 portIndex, OMX_BOOL enable, OMX_U32 maxFrameWidth,
+        OMX_U32 maxFrameHeight) {
+    Mutex::Autolock autolock(mLock);
+
+    OMX_INDEXTYPE index;
+    OMX_STRING name = const_cast<OMX_STRING>(
+            "OMX.google.android.index.prepareForAdaptivePlayback");
+
+    OMX_ERRORTYPE err = OMX_GetExtensionIndex(mHandle, name, &index);
+    if (err != OMX_ErrorNone) {
+        ALOGW_IF(enable, "OMX_GetExtensionIndex %s failed", name);
+        return StatusFromOMXError(err);
+    }
+
+    PrepareForAdaptivePlaybackParams params;
+    params.nSize = sizeof(params);
+    params.nVersion.s.nVersionMajor = 1;
+    params.nVersion.s.nVersionMinor = 0;
+    params.nVersion.s.nRevision = 0;
+    params.nVersion.s.nStep = 0;
+
+    params.nPortIndex = portIndex;
+    params.bEnable = enable;
+    params.nMaxFrameWidth = maxFrameWidth;
+    params.nMaxFrameHeight = maxFrameHeight;
+    if ((err = OMX_SetParameter(mHandle, index, &params)) != OMX_ErrorNone) {
+        ALOGW("OMX_SetParameter failed for PrepareForAdaptivePlayback "
+              "with error %d (0x%08x)", err, err);
+        return UNKNOWN_ERROR;
+    }
+    return err;
+}
+
 status_t OMXNodeInstance::useBuffer(
         OMX_U32 portIndex, const sp<IMemory> &params,
         OMX::buffer_id *buffer) {
@@ -571,7 +610,7 @@
 }
 
 status_t OMXNodeInstance::updateGraphicBufferInMeta(
-        OMX_U32 portIndex, const sp<GraphicBuffer>& graphicBuffer,
+        OMX_U32 /* portIndex */, const sp<GraphicBuffer>& graphicBuffer,
         OMX::buffer_id buffer) {
     Mutex::Autolock autoLock(mLock);
 
@@ -810,6 +849,9 @@
     switch (type) {
         case IOMX::INTERNAL_OPTION_SUSPEND:
         case IOMX::INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY:
+        case IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP:
+        case IOMX::INTERNAL_OPTION_START_TIME:
+        case IOMX::INTERNAL_OPTION_TIME_LAPSE:
         {
             const sp<GraphicBufferSource> &bufferSource =
                 getGraphicBufferSource();
@@ -825,7 +867,8 @@
 
                 bool suspend = *(bool *)data;
                 bufferSource->suspend(suspend);
-            } else {
+            } else if (type ==
+                    IOMX::INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY){
                 if (size != sizeof(int64_t)) {
                     return INVALID_OPERATION;
                 }
@@ -833,6 +876,29 @@
                 int64_t delayUs = *(int64_t *)data;
 
                 return bufferSource->setRepeatPreviousFrameDelayUs(delayUs);
+            } else if (type ==
+                    IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP){
+                if (size != sizeof(int64_t)) {
+                    return INVALID_OPERATION;
+                }
+
+                int64_t maxGapUs = *(int64_t *)data;
+
+                return bufferSource->setMaxTimestampGapUs(maxGapUs);
+            } else if (type == IOMX::INTERNAL_OPTION_START_TIME) {
+                if (size != sizeof(int64_t)) {
+                    return INVALID_OPERATION;
+                }
+
+                int64_t skipFramesBeforeUs = *(int64_t *)data;
+
+                bufferSource->setSkipFramesBeforeUs(skipFramesBeforeUs);
+            } else { // IOMX::INTERNAL_OPTION_TIME_LAPSE
+                if (size != sizeof(int64_t) * 2) {
+                    return INVALID_OPERATION;
+                }
+
+                bufferSource->setTimeLapseUs((int64_t *)data);
             }
 
             return OK;
@@ -844,6 +910,8 @@
 }
 
 void OMXNodeInstance::onMessage(const omx_message &msg) {
+    const sp<GraphicBufferSource>& bufferSource(getGraphicBufferSource());
+
     if (msg.type == omx_message::FILL_BUFFER_DONE) {
         OMX_BUFFERHEADERTYPE *buffer =
             static_cast<OMX_BUFFERHEADERTYPE *>(
@@ -853,10 +921,18 @@
             static_cast<BufferMeta *>(buffer->pAppPrivate);
 
         buffer_meta->CopyFromOMX(buffer);
-    } else if (msg.type == omx_message::EMPTY_BUFFER_DONE) {
-        const sp<GraphicBufferSource>& bufferSource(getGraphicBufferSource());
 
         if (bufferSource != NULL) {
+            // fix up the buffer info (especially timestamp) if needed
+            bufferSource->codecBufferFilled(buffer);
+
+            omx_message newMsg = msg;
+            newMsg.u.extended_buffer_data.timestamp = buffer->nTimeStamp;
+            mObserver->onMessage(newMsg);
+            return;
+        }
+    } else if (msg.type == omx_message::EMPTY_BUFFER_DONE) {
+        if (bufferSource != NULL) {
             // This is one of the buffers used exclusively by
             // GraphicBufferSource.
             // Don't dispatch a message back to ACodec, since it doesn't
@@ -902,7 +978,7 @@
 
 // static
 OMX_ERRORTYPE OMXNodeInstance::OnEvent(
-        OMX_IN OMX_HANDLETYPE hComponent,
+        OMX_IN OMX_HANDLETYPE /* hComponent */,
         OMX_IN OMX_PTR pAppData,
         OMX_IN OMX_EVENTTYPE eEvent,
         OMX_IN OMX_U32 nData1,
@@ -918,7 +994,7 @@
 
 // static
 OMX_ERRORTYPE OMXNodeInstance::OnEmptyBufferDone(
-        OMX_IN OMX_HANDLETYPE hComponent,
+        OMX_IN OMX_HANDLETYPE /* hComponent */,
         OMX_IN OMX_PTR pAppData,
         OMX_IN OMX_BUFFERHEADERTYPE* pBuffer) {
     OMXNodeInstance *instance = static_cast<OMXNodeInstance *>(pAppData);
@@ -930,7 +1006,7 @@
 
 // static
 OMX_ERRORTYPE OMXNodeInstance::OnFillBufferDone(
-        OMX_IN OMX_HANDLETYPE hComponent,
+        OMX_IN OMX_HANDLETYPE /* hComponent */,
         OMX_IN OMX_PTR pAppData,
         OMX_IN OMX_BUFFERHEADERTYPE* pBuffer) {
     OMXNodeInstance *instance = static_cast<OMXNodeInstance *>(pAppData);

media/libstagefright/omx/SoftOMXComponent.cpp

diff --git a/media/libstagefright/omx/SoftOMXComponent.cpp b/media/libstagefright/omx/SoftOMXComponent.cpp
index b1c34dc..646cd32 100644
--- a/media/libstagefright/omx/SoftOMXComponent.cpp
+++ b/media/libstagefright/omx/SoftOMXComponent.cpp
@@ -257,69 +257,69 @@
 ////////////////////////////////////////////////////////////////////////////////
 
 OMX_ERRORTYPE SoftOMXComponent::sendCommand(
-        OMX_COMMANDTYPE cmd, OMX_U32 param, OMX_PTR data) {
+        OMX_COMMANDTYPE /* cmd */, OMX_U32 /* param */, OMX_PTR /* data */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::getParameter(
-        OMX_INDEXTYPE index, OMX_PTR params) {
+        OMX_INDEXTYPE /* index */, OMX_PTR /* params */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::setParameter(
-        OMX_INDEXTYPE index, const OMX_PTR params) {
+        OMX_INDEXTYPE /* index */, const OMX_PTR /* params */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::getConfig(
-        OMX_INDEXTYPE index, OMX_PTR params) {
+        OMX_INDEXTYPE /* index */, OMX_PTR /* params */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::setConfig(
-        OMX_INDEXTYPE index, const OMX_PTR params) {
+        OMX_INDEXTYPE /* index */, const OMX_PTR /* params */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::getExtensionIndex(
-        const char *name, OMX_INDEXTYPE *index) {
+        const char * /* name */, OMX_INDEXTYPE * /* index */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::useBuffer(
-        OMX_BUFFERHEADERTYPE **buffer,
-        OMX_U32 portIndex,
-        OMX_PTR appPrivate,
-        OMX_U32 size,
-        OMX_U8 *ptr) {
+        OMX_BUFFERHEADERTYPE ** /* buffer */,
+        OMX_U32 /* portIndex */,
+        OMX_PTR /* appPrivate */,
+        OMX_U32 /* size */,
+        OMX_U8 * /* ptr */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::allocateBuffer(
-        OMX_BUFFERHEADERTYPE **buffer,
-        OMX_U32 portIndex,
-        OMX_PTR appPrivate,
-        OMX_U32 size) {
+        OMX_BUFFERHEADERTYPE ** /* buffer */,
+        OMX_U32 /* portIndex */,
+        OMX_PTR /* appPrivate */,
+        OMX_U32 /* size */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::freeBuffer(
-        OMX_U32 portIndex,
-        OMX_BUFFERHEADERTYPE *buffer) {
+        OMX_U32 /* portIndex */,
+        OMX_BUFFERHEADERTYPE * /* buffer */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::emptyThisBuffer(
-        OMX_BUFFERHEADERTYPE *buffer) {
+        OMX_BUFFERHEADERTYPE * /* buffer */) {
     return OMX_ErrorUndefined;
 }
 
 OMX_ERRORTYPE SoftOMXComponent::fillThisBuffer(
-        OMX_BUFFERHEADERTYPE *buffer) {
+        OMX_BUFFERHEADERTYPE * /* buffer */) {
     return OMX_ErrorUndefined;
 }
 
-OMX_ERRORTYPE SoftOMXComponent::getState(OMX_STATETYPE *state) {
+OMX_ERRORTYPE SoftOMXComponent::getState(OMX_STATETYPE * /* state */) {
     return OMX_ErrorUndefined;
 }
 

media/libstagefright/omx/SoftOMXPlugin.cpp

diff --git a/media/libstagefright/omx/SoftOMXPlugin.cpp b/media/libstagefright/omx/SoftOMXPlugin.cpp
index d6cde73..65f5404 100644
--- a/media/libstagefright/omx/SoftOMXPlugin.cpp
+++ b/media/libstagefright/omx/SoftOMXPlugin.cpp
@@ -50,6 +50,7 @@
     { "OMX.google.mpeg4.encoder", "mpeg4enc", "video_encoder.mpeg4" },
     { "OMX.google.mp3.decoder", "mp3dec", "audio_decoder.mp3" },
     { "OMX.google.vorbis.decoder", "vorbisdec", "audio_decoder.vorbis" },
+    { "OMX.google.opus.decoder", "opusdec", "audio_decoder.opus" },
     { "OMX.google.vp8.decoder", "vpxdec", "video_decoder.vp8" },
     { "OMX.google.vp9.decoder", "vpxdec", "video_decoder.vp9" },
     { "OMX.google.vp8.encoder", "vpxenc", "video_encoder.vp8" },
@@ -154,7 +155,7 @@
 
 OMX_ERRORTYPE SoftOMXPlugin::enumerateComponents(
         OMX_STRING name,
-        size_t size,
+        size_t /* size */,
         OMX_U32 index) {
     if (index >= kNumComponents) {
         return OMX_ErrorNoMore;

media/libstagefright/omx/tests/Android.mk

diff --git a/media/libstagefright/omx/tests/Android.mk b/media/libstagefright/omx/tests/Android.mk
index 1061c39..8b79af4 100644
--- a/media/libstagefright/omx/tests/Android.mk
+++ b/media/libstagefright/omx/tests/Android.mk
@@ -11,6 +11,8 @@
 	$(TOP)/frameworks/av/media/libstagefright \
 	$(TOP)/frameworks/native/include/media/openmax
 
+LOCAL_CFLAGS += -Werror
+
 LOCAL_MODULE := omx_tests
 
 LOCAL_MODULE_TAGS := tests

media/libstagefright/omx/tests/OMXHarness.cpp

diff --git a/media/libstagefright/omx/tests/OMXHarness.cpp b/media/libstagefright/omx/tests/OMXHarness.cpp
index 4bee808..f4dfd6b 100644
--- a/media/libstagefright/omx/tests/OMXHarness.cpp
+++ b/media/libstagefright/omx/tests/OMXHarness.cpp
@@ -16,6 +16,7 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "OMXHarness"
+#include <inttypes.h>
 #include <utils/Log.h>
 
 #include "OMXHarness.h"
@@ -25,6 +26,7 @@
 #include <binder/ProcessState.h>
 #include <binder/IServiceManager.h>
 #include <binder/MemoryDealer.h>
+#include <media/IMediaHTTPService.h>
 #include <media/IMediaPlayerService.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/ALooper.h>
@@ -241,7 +243,8 @@
 };
 
 static sp<MediaExtractor> CreateExtractorFromURI(const char *uri) {
-    sp<DataSource> source = DataSource::CreateFromURI(uri);
+    sp<DataSource> source =
+        DataSource::CreateFromURI(NULL /* httpService */, uri);
 
     if (source == NULL) {
         return NULL;
@@ -460,6 +463,7 @@
         { "audio_decoder.aac", "audio/mp4a-latm" },
         { "audio_decoder.mp3", "audio/mpeg" },
         { "audio_decoder.vorbis", "audio/vorbis" },
+        { "audio_decoder.opus", "audio/opus" },
         { "audio_decoder.g711alaw", MEDIA_MIMETYPE_AUDIO_G711_ALAW },
         { "audio_decoder.g711mlaw", MEDIA_MIMETYPE_AUDIO_G711_MLAW },
     };
@@ -492,6 +496,7 @@
         { "audio/mpeg",
           "file:///sdcard/media_api/music/MP3_48KHz_128kbps_s_1_17_CBR.mp3" },
         { "audio/vorbis", NULL },
+        { "audio/opus", NULL },
         { "video/x-vnd.on2.vp8",
           "file:///sdcard/media_api/video/big-buck-bunny_trailer.webm" },
         { MEDIA_MIMETYPE_AUDIO_G711_ALAW, "file:///sdcard/M1F1-Alaw-AFsp.wav" },
@@ -711,11 +716,11 @@
             int64_t bufferTimeUs;
             CHECK(buffer->meta_data()->findInt64(kKeyTime, &bufferTimeUs));
             if (!CloseEnough(bufferTimeUs, actualSeekTimeUs)) {
-                printf("\n  * Attempted seeking to %lld us (%.2f secs)",
+                printf("\n  * Attempted seeking to %" PRId64 " us (%.2f secs)",
                        requestedSeekTimeUs, requestedSeekTimeUs / 1E6);
-                printf("\n  * Nearest keyframe is at %lld us (%.2f secs)",
+                printf("\n  * Nearest keyframe is at %" PRId64 " us (%.2f secs)",
                        actualSeekTimeUs, actualSeekTimeUs / 1E6);
-                printf("\n  * Returned buffer was at %lld us (%.2f secs)\n\n",
+                printf("\n  * Returned buffer was at %" PRId64 " us (%.2f secs)\n\n",
                        bufferTimeUs, bufferTimeUs / 1E6);
 
                 buffer->release();

media/libstagefright/rtsp/AMPEG2TSAssembler.cpp

diff --git a/media/libstagefright/rtsp/AMPEG2TSAssembler.cpp b/media/libstagefright/rtsp/AMPEG2TSAssembler.cpp
index 4c9bf5b..dca5c89 100644
--- a/media/libstagefright/rtsp/AMPEG2TSAssembler.cpp
+++ b/media/libstagefright/rtsp/AMPEG2TSAssembler.cpp
@@ -34,7 +34,9 @@
 namespace android {
 
 AMPEG2TSAssembler::AMPEG2TSAssembler(
-        const sp<AMessage> &notify, const char *desc, const AString &params)
+        const sp<AMessage> &notify,
+        const char * /* desc */,
+        const AString & /* params */)
     : mNotifyMsg(notify),
       mNextExpectedSeqNoValid(false),
       mNextExpectedSeqNo(0) {

media/libstagefright/rtsp/APacketSource.cpp

diff --git a/media/libstagefright/rtsp/APacketSource.cpp b/media/libstagefright/rtsp/APacketSource.cpp
index 462c384..09f52bc 100644
--- a/media/libstagefright/rtsp/APacketSource.cpp
+++ b/media/libstagefright/rtsp/APacketSource.cpp
@@ -23,7 +23,7 @@
 #include "ARawAudioAssembler.h"
 #include "ASessionDescription.h"
 
-#include "avc_utils.h"
+#include "include/avc_utils.h"
 
 #include <ctype.h>
 

media/libstagefright/rtsp/ARTPWriter.cpp

diff --git a/media/libstagefright/rtsp/ARTPWriter.cpp b/media/libstagefright/rtsp/ARTPWriter.cpp
index 0d07043..c46d16f 100644
--- a/media/libstagefright/rtsp/ARTPWriter.cpp
+++ b/media/libstagefright/rtsp/ARTPWriter.cpp
@@ -114,7 +114,7 @@
     return (mFlags & kFlagEOS) != 0;
 }
 
-status_t ARTPWriter::start(MetaData *params) {
+status_t ARTPWriter::start(MetaData * /* params */) {
     Mutex::Autolock autoLock(mLock);
     if (mFlags & kFlagStarted) {
         return INVALID_OPERATION;

media/libstagefright/rtsp/ARTSPConnection.cpp

diff --git a/media/libstagefright/rtsp/ARTSPConnection.cpp b/media/libstagefright/rtsp/ARTSPConnection.cpp
index efde7a9..4054da6 100644
--- a/media/libstagefright/rtsp/ARTSPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTSPConnection.cpp
@@ -33,7 +33,7 @@
 #include <openssl/md5.h>
 #include <sys/socket.h>
 
-#include "HTTPBase.h"
+#include "include/HTTPBase.h"
 
 namespace android {
 

media/libstagefright/rtsp/ARawAudioAssembler.cpp

diff --git a/media/libstagefright/rtsp/ARawAudioAssembler.cpp b/media/libstagefright/rtsp/ARawAudioAssembler.cpp
index 0da5dd2..167f7a4 100644
--- a/media/libstagefright/rtsp/ARawAudioAssembler.cpp
+++ b/media/libstagefright/rtsp/ARawAudioAssembler.cpp
@@ -34,7 +34,9 @@
 namespace android {
 
 ARawAudioAssembler::ARawAudioAssembler(
-        const sp<AMessage> &notify, const char *desc, const AString &params)
+        const sp<AMessage> &notify,
+        const char * /* desc */,
+        const AString & /* params */)
     : mNotifyMsg(notify),
       mNextExpectedSeqNoValid(false),
       mNextExpectedSeqNo(0) {

media/libstagefright/rtsp/Android.mk

diff --git a/media/libstagefright/rtsp/Android.mk b/media/libstagefright/rtsp/Android.mk
index e77c69c..39eedc0 100644
--- a/media/libstagefright/rtsp/Android.mk
+++ b/media/libstagefright/rtsp/Android.mk
@@ -20,7 +20,7 @@
         SDPLoader.cpp               \
 
 LOCAL_C_INCLUDES:= \
-	$(TOP)/frameworks/av/media/libstagefright/include \
+	$(TOP)/frameworks/av/media/libstagefright \
 	$(TOP)/frameworks/native/include/media/openmax \
 	$(TOP)/external/openssl/include
 
@@ -30,6 +30,8 @@
     LOCAL_CFLAGS += -Wno-psabi
 endif
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_STATIC_LIBRARY)
 
 ################################################################################

media/libstagefright/rtsp/MyHandler.h

diff --git a/media/libstagefright/rtsp/MyHandler.h b/media/libstagefright/rtsp/MyHandler.h
index cd77aa0..45470a3 100644
--- a/media/libstagefright/rtsp/MyHandler.h
+++ b/media/libstagefright/rtsp/MyHandler.h
@@ -19,7 +19,11 @@
 #define MY_HANDLER_H_
 
 //#define LOG_NDEBUG 0
+
+#ifndef LOG_TAG
 #define LOG_TAG "MyHandler"
+#endif
+
 #include <utils/Log.h>
 
 #include "APacketSource.h"
@@ -42,6 +46,12 @@
 
 #include "HTTPBase.h"
 
+#if LOG_NDEBUG
+#define UNUSED_UNLESS_VERBOSE(x) (void)(x)
+#else
+#define UNUSED_UNLESS_VERBOSE(x)
+#endif
+
 // If no access units are received within 5 secs, assume that the rtp
 // stream has ended and signal end of stream.
 static int64_t kAccessUnitTimeoutUs = 10000000ll;
@@ -178,7 +188,7 @@
         mConn->connect(mOriginalSessionURL.c_str(), reply);
     }
 
-    AString getControlURL(sp<ASessionDescription> desc) {
+    AString getControlURL() {
         AString sessionLevelControlURL;
         if (mSessionDesc->findAttribute(
                 0,
@@ -545,7 +555,7 @@
                                 mBaseURL = tmp;
                             }
 
-                            mControlURL = getControlURL(mSessionDesc);
+                            mControlURL = getControlURL();
 
                             if (mSessionDesc->countTracks() < 2) {
                                 // There's no actual tracks in this session.
@@ -591,7 +601,7 @@
 
                         mSeekable = !isLiveStream(mSessionDesc);
 
-                        mControlURL = getControlURL(mSessionDesc);
+                        mControlURL = getControlURL();
 
                         if (mSessionDesc->countTracks() < 2) {
                             // There's no actual tracks in this session.
@@ -1805,6 +1815,8 @@
     bool addMediaTimestamp(
             int32_t trackIndex, const TrackInfo *track,
             const sp<ABuffer> &accessUnit) {
+        UNUSED_UNLESS_VERBOSE(trackIndex);
+
         uint32_t rtpTime;
         CHECK(accessUnit->meta()->findInt32(
                     "rtp-time", (int32_t *)&rtpTime));

media/libstagefright/rtsp/SDPLoader.cpp

diff --git a/media/libstagefright/rtsp/SDPLoader.cpp b/media/libstagefright/rtsp/SDPLoader.cpp
index ed3fa7e..ce1e89d 100644
--- a/media/libstagefright/rtsp/SDPLoader.cpp
+++ b/media/libstagefright/rtsp/SDPLoader.cpp
@@ -18,11 +18,13 @@
 #define LOG_TAG "SDPLoader"
 #include <utils/Log.h>
 
-#include "SDPLoader.h"
+#include "include/SDPLoader.h"
 
 #include "ASessionDescription.h"
-#include "HTTPBase.h"
 
+#include <media/IMediaHTTPConnection.h>
+#include <media/IMediaHTTPService.h>
+#include <media/stagefright/MediaHTTP.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
 
@@ -30,22 +32,15 @@
 
 namespace android {
 
-SDPLoader::SDPLoader(const sp<AMessage> &notify, uint32_t flags, bool uidValid, uid_t uid)
+SDPLoader::SDPLoader(
+        const sp<AMessage> &notify,
+        uint32_t flags,
+        const sp<IMediaHTTPService> &httpService)
     : mNotify(notify),
       mFlags(flags),
-      mUIDValid(uidValid),
-      mUID(uid),
       mNetLooper(new ALooper),
       mCancelled(false),
-      mHTTPDataSource(
-              HTTPBase::Create(
-                  (mFlags & kFlagIncognito)
-                    ? HTTPBase::kFlagIncognito
-                    : 0)) {
-    if (mUIDValid) {
-        mHTTPDataSource->setUID(mUID);
-    }
-
+      mHTTPDataSource(new MediaHTTP(httpService->makeHTTPConnection())) {
     mNetLooper->setName("sdp net");
     mNetLooper->start(false /* runOnCallingThread */,
                       false /* canCallJava */,
@@ -130,7 +125,7 @@
         ssize_t readSize = mHTTPDataSource->readAt(0, buffer->data(), sdpSize);
 
         if (readSize < 0) {
-            ALOGE("Failed to read SDP, error code = %ld", readSize);
+            ALOGE("Failed to read SDP, error code = %d", readSize);
             err = UNKNOWN_ERROR;
         } else {
             desc = new ASessionDescription;

media/libstagefright/tests/DummyRecorder.cpp

diff --git a/media/libstagefright/tests/DummyRecorder.cpp b/media/libstagefright/tests/DummyRecorder.cpp
index ac37b28..8f17088 100644
--- a/media/libstagefright/tests/DummyRecorder.cpp
+++ b/media/libstagefright/tests/DummyRecorder.cpp
@@ -61,7 +61,7 @@
     mSource->stop();
     void *dummy;
     pthread_join(mThread, &dummy);
-    status_t err = (status_t) dummy;
+    status_t err = static_cast<status_t>(reinterpret_cast<uintptr_t>(dummy));
 
     ALOGV("Ending the reading thread");
     return err;

media/libstagefright/tests/SurfaceMediaSource_test.cpp

diff --git a/media/libstagefright/tests/SurfaceMediaSource_test.cpp b/media/libstagefright/tests/SurfaceMediaSource_test.cpp
index 49ffcd6..a3093d0 100644
--- a/media/libstagefright/tests/SurfaceMediaSource_test.cpp
+++ b/media/libstagefright/tests/SurfaceMediaSource_test.cpp
@@ -35,7 +35,6 @@
 #include <gui/SurfaceComposerClient.h>
 
 #include <binder/ProcessState.h>
-#include <ui/FramebufferNativeWindow.h>
 
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/MediaBufferGroup.h>
@@ -527,7 +526,8 @@
 }
 
 // Dequeuing and queuing the buffer without really filling it in.
-void SurfaceMediaSourceTest::oneBufferPassNoFill(int width, int height ) {
+void SurfaceMediaSourceTest::oneBufferPassNoFill(
+        int /* width */, int /* height  */) {
     ANativeWindowBuffer* anb;
     ASSERT_EQ(NO_ERROR, native_window_dequeue_buffer_and_wait(mANW.get(), &anb));
     ASSERT_TRUE(anb != NULL);
@@ -746,9 +746,8 @@
     CHECK(fd >= 0);
 
     sp<MediaRecorder> mr = SurfaceMediaSourceGLTest::setUpMediaRecorder(fd,
-            VIDEO_SOURCE_GRALLOC_BUFFER,
-            OUTPUT_FORMAT_MPEG_4, VIDEO_ENCODER_H264, mYuvTexWidth,
-            mYuvTexHeight, 30);
+            VIDEO_SOURCE_SURFACE, OUTPUT_FORMAT_MPEG_4, VIDEO_ENCODER_H264,
+            mYuvTexWidth, mYuvTexHeight, 30);
     // get the reference to the surfacemediasource living in
     // mediaserver that is created by stagefrightrecorder
     sp<IGraphicBufferProducer> iST = mr->querySurfaceMediaSourceFromMediaServer();
@@ -880,7 +879,7 @@
     }
     CHECK(fd >= 0);
 
-    sp<MediaRecorder> mr = setUpMediaRecorder(fd, VIDEO_SOURCE_GRALLOC_BUFFER,
+    sp<MediaRecorder> mr = setUpMediaRecorder(fd, VIDEO_SOURCE_SURFACE,
             OUTPUT_FORMAT_MPEG_4, VIDEO_ENCODER_H264, mYuvTexWidth, mYuvTexHeight, 30);
 
     // get the reference to the surfacemediasource living in
@@ -923,7 +922,7 @@
     }
     CHECK(fd >= 0);
 
-    sp<MediaRecorder> mr = setUpMediaRecorder(fd, VIDEO_SOURCE_GRALLOC_BUFFER,
+    sp<MediaRecorder> mr = setUpMediaRecorder(fd, VIDEO_SOURCE_SURFACE,
             OUTPUT_FORMAT_MPEG_4, VIDEO_ENCODER_H264, mYuvTexWidth, mYuvTexHeight, 30);
 
     // get the reference to the surfacemediasource living in

media/libstagefright/timedtext/Android.mk

diff --git a/media/libstagefright/timedtext/Android.mk b/media/libstagefright/timedtext/Android.mk
index f099bbd..6a8b9fc 100644
--- a/media/libstagefright/timedtext/Android.mk
+++ b/media/libstagefright/timedtext/Android.mk
@@ -9,7 +9,8 @@
         TimedTextSRTSource.cpp    \
         TimedTextPlayer.cpp
 
-LOCAL_CFLAGS += -Wno-multichar
+LOCAL_CFLAGS += -Wno-multichar -Werror
+
 LOCAL_C_INCLUDES:= \
         $(TOP)/frameworks/av/include/media/stagefright/timedtext \
         $(TOP)/frameworks/av/media/libstagefright

media/libstagefright/timedtext/TimedTextDriver.cpp

diff --git a/media/libstagefright/timedtext/TimedTextDriver.cpp b/media/libstagefright/timedtext/TimedTextDriver.cpp
index 12fd7f4..71aa21e 100644
--- a/media/libstagefright/timedtext/TimedTextDriver.cpp
+++ b/media/libstagefright/timedtext/TimedTextDriver.cpp
@@ -20,6 +20,7 @@
 
 #include <binder/IPCThreadState.h>
 
+#include <media/IMediaHTTPService.h>
 #include <media/mediaplayer.h>
 #include <media/MediaPlayerInterface.h>
 #include <media/stagefright/DataSource.h>
@@ -40,9 +41,11 @@
 namespace android {
 
 TimedTextDriver::TimedTextDriver(
-        const wp<MediaPlayerBase> &listener)
+        const wp<MediaPlayerBase> &listener,
+        const sp<IMediaHTTPService> &httpService)
     : mLooper(new ALooper),
       mListener(listener),
+      mHTTPService(httpService),
       mState(UNINITIALIZED),
       mCurrentTrackIndex(UINT_MAX) {
     mLooper->setName("TimedTextDriver");
@@ -207,7 +210,7 @@
     }
 
     sp<DataSource> dataSource =
-            DataSource::CreateFromURI(uri);
+            DataSource::CreateFromURI(mHTTPService, uri);
     return createOutOfBandTextSource(trackIndex, mimeType, dataSource);
 }
 

media/libstagefright/timedtext/TimedTextSource.h

diff --git a/media/libstagefright/timedtext/TimedTextSource.h b/media/libstagefright/timedtext/TimedTextSource.h
index 756cc31..8c1c1cd 100644
--- a/media/libstagefright/timedtext/TimedTextSource.h
+++ b/media/libstagefright/timedtext/TimedTextSource.h
@@ -47,7 +47,7 @@
           int64_t *endTimeUs,
           Parcel *parcel,
           const MediaSource::ReadOptions *options = NULL) = 0;
-  virtual status_t extractGlobalDescriptions(Parcel *parcel) {
+  virtual status_t extractGlobalDescriptions(Parcel * /* parcel */) {
       return INVALID_OPERATION;
   }
   virtual sp<MetaData> getFormat();

media/libstagefright/wifi-display/Android.mk

diff --git a/media/libstagefright/wifi-display/Android.mk b/media/libstagefright/wifi-display/Android.mk
index c7d107e..f70454a 100644
--- a/media/libstagefright/wifi-display/Android.mk
+++ b/media/libstagefright/wifi-display/Android.mk
@@ -3,16 +3,9 @@
 include $(CLEAR_VARS)
 
 LOCAL_SRC_FILES:= \
-        MediaReceiver.cpp               \
         MediaSender.cpp                 \
         Parameters.cpp                  \
-        rtp/RTPAssembler.cpp            \
-        rtp/RTPReceiver.cpp             \
         rtp/RTPSender.cpp               \
-        sink/DirectRenderer.cpp         \
-        sink/WifiDisplaySink.cpp        \
-        SNTPClient.cpp                  \
-        TimeSyncer.cpp                  \
         source/Converter.cpp            \
         source/MediaPuller.cpp          \
         source/PlaybackSession.cpp      \
@@ -42,87 +35,3 @@
 LOCAL_MODULE_TAGS:= optional
 
 include $(BUILD_SHARED_LIBRARY)
-
-################################################################################
-
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES:= \
-        wfd.cpp                 \
-
-LOCAL_SHARED_LIBRARIES:= \
-        libbinder                       \
-        libgui                          \
-        libmedia                        \
-        libstagefright                  \
-        libstagefright_foundation       \
-        libstagefright_wfd              \
-        libutils                        \
-        liblog                          \
-
-LOCAL_MODULE:= wfd
-
-include $(BUILD_EXECUTABLE)
-
-################################################################################
-
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES:= \
-        udptest.cpp                 \
-
-LOCAL_SHARED_LIBRARIES:= \
-        libbinder                       \
-        libgui                          \
-        libmedia                        \
-        libstagefright                  \
-        libstagefright_foundation       \
-        libstagefright_wfd              \
-        libutils                        \
-        liblog                          \
-
-LOCAL_MODULE:= udptest
-
-include $(BUILD_EXECUTABLE)
-
-################################################################################
-
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES:= \
-        rtptest.cpp                 \
-
-LOCAL_SHARED_LIBRARIES:= \
-        libbinder                       \
-        libgui                          \
-        libmedia                        \
-        libstagefright                  \
-        libstagefright_foundation       \
-        libstagefright_wfd              \
-        libutils                        \
-        liblog                          \
-
-LOCAL_MODULE:= rtptest
-
-include $(BUILD_EXECUTABLE)
-
-################################################################################
-
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES:= \
-        nettest.cpp                     \
-
-LOCAL_SHARED_LIBRARIES:= \
-        libbinder                       \
-        libgui                          \
-        libmedia                        \
-        libstagefright                  \
-        libstagefright_foundation       \
-        libstagefright_wfd              \
-        libutils                        \
-        liblog                          \
-
-LOCAL_MODULE:= nettest
-
-include $(BUILD_EXECUTABLE)

media/libstagefright/wifi-display/MediaReceiver.cpp

diff --git a/media/libstagefright/wifi-display/MediaReceiver.cpp b/media/libstagefright/wifi-display/MediaReceiver.cpp
deleted file mode 100644
index 5524235..0000000
--- a/media/libstagefright/wifi-display/MediaReceiver.cpp
+++ /dev/null
@@ -1,328 +0,0 @@
-/*
- * Copyright 2013, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "MediaReceiver"
-#include <utils/Log.h>
-
-#include "MediaReceiver.h"
-
-#include "AnotherPacketSource.h"
-#include "rtp/RTPReceiver.h"
-
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/foundation/ANetworkSession.h>
-#include <media/stagefright/MetaData.h>
-#include <media/stagefright/Utils.h>
-
-namespace android {
-
-MediaReceiver::MediaReceiver(
-        const sp<ANetworkSession> &netSession,
-        const sp<AMessage> &notify)
-    : mNetSession(netSession),
-      mNotify(notify),
-      mMode(MODE_UNDEFINED),
-      mGeneration(0),
-      mInitStatus(OK),
-      mInitDoneCount(0) {
-}
-
-MediaReceiver::~MediaReceiver() {
-}
-
-ssize_t MediaReceiver::addTrack(
-        RTPReceiver::TransportMode rtpMode,
-        RTPReceiver::TransportMode rtcpMode,
-        int32_t *localRTPPort) {
-    if (mMode != MODE_UNDEFINED) {
-        return INVALID_OPERATION;
-    }
-
-    size_t trackIndex = mTrackInfos.size();
-
-    TrackInfo info;
-
-    sp<AMessage> notify = new AMessage(kWhatReceiverNotify, id());
-    notify->setInt32("generation", mGeneration);
-    notify->setSize("trackIndex", trackIndex);
-
-    info.mReceiver = new RTPReceiver(mNetSession, notify);
-    looper()->registerHandler(info.mReceiver);
-
-    info.mReceiver->registerPacketType(
-            33, RTPReceiver::PACKETIZATION_TRANSPORT_STREAM);
-
-    info.mReceiver->registerPacketType(
-            96, RTPReceiver::PACKETIZATION_AAC);
-
-    info.mReceiver->registerPacketType(
-            97, RTPReceiver::PACKETIZATION_H264);
-
-    status_t err = info.mReceiver->initAsync(
-            rtpMode,
-            rtcpMode,
-            localRTPPort);
-
-    if (err != OK) {
-        looper()->unregisterHandler(info.mReceiver->id());
-        info.mReceiver.clear();
-
-        return err;
-    }
-
-    mTrackInfos.push_back(info);
-
-    return trackIndex;
-}
-
-status_t MediaReceiver::connectTrack(
-        size_t trackIndex,
-        const char *remoteHost,
-        int32_t remoteRTPPort,
-        int32_t remoteRTCPPort) {
-    if (trackIndex >= mTrackInfos.size()) {
-        return -ERANGE;
-    }
-
-    TrackInfo *info = &mTrackInfos.editItemAt(trackIndex);
-    return info->mReceiver->connect(remoteHost, remoteRTPPort, remoteRTCPPort);
-}
-
-status_t MediaReceiver::initAsync(Mode mode) {
-    if ((mode == MODE_TRANSPORT_STREAM || mode == MODE_TRANSPORT_STREAM_RAW)
-            && mTrackInfos.size() > 1) {
-        return INVALID_OPERATION;
-    }
-
-    sp<AMessage> msg = new AMessage(kWhatInit, id());
-    msg->setInt32("mode", mode);
-    msg->post();
-
-    return OK;
-}
-
-void MediaReceiver::onMessageReceived(const sp<AMessage> &msg) {
-    switch (msg->what()) {
-        case kWhatInit:
-        {
-            int32_t mode;
-            CHECK(msg->findInt32("mode", &mode));
-
-            CHECK_EQ(mMode, MODE_UNDEFINED);
-            mMode = (Mode)mode;
-
-            if (mInitStatus != OK || mInitDoneCount == mTrackInfos.size()) {
-                notifyInitDone(mInitStatus);
-            }
-
-            mTSParser = new ATSParser(
-                    ATSParser::ALIGNED_VIDEO_DATA
-                        | ATSParser::TS_TIMESTAMPS_ARE_ABSOLUTE);
-
-            mFormatKnownMask = 0;
-            break;
-        }
-
-        case kWhatReceiverNotify:
-        {
-            int32_t generation;
-            CHECK(msg->findInt32("generation", &generation));
-            if (generation != mGeneration) {
-                break;
-            }
-
-            onReceiverNotify(msg);
-            break;
-        }
-
-        default:
-            TRESPASS();
-    }
-}
-
-void MediaReceiver::onReceiverNotify(const sp<AMessage> &msg) {
-    int32_t what;
-    CHECK(msg->findInt32("what", &what));
-
-    switch (what) {
-        case RTPReceiver::kWhatInitDone:
-        {
-            ++mInitDoneCount;
-
-            int32_t err;
-            CHECK(msg->findInt32("err", &err));
-
-            if (err != OK) {
-                mInitStatus = err;
-                ++mGeneration;
-            }
-
-            if (mMode != MODE_UNDEFINED) {
-                if (mInitStatus != OK || mInitDoneCount == mTrackInfos.size()) {
-                    notifyInitDone(mInitStatus);
-                }
-            }
-            break;
-        }
-
-        case RTPReceiver::kWhatError:
-        {
-            int32_t err;
-            CHECK(msg->findInt32("err", &err));
-
-            notifyError(err);
-            break;
-        }
-
-        case RTPReceiver::kWhatAccessUnit:
-        {
-            size_t trackIndex;
-            CHECK(msg->findSize("trackIndex", &trackIndex));
-
-            sp<ABuffer> accessUnit;
-            CHECK(msg->findBuffer("accessUnit", &accessUnit));
-
-            int32_t followsDiscontinuity;
-            if (!msg->findInt32(
-                        "followsDiscontinuity", &followsDiscontinuity)) {
-                followsDiscontinuity = 0;
-            }
-
-            if (mMode == MODE_TRANSPORT_STREAM) {
-                if (followsDiscontinuity) {
-                    mTSParser->signalDiscontinuity(
-                            ATSParser::DISCONTINUITY_TIME, NULL /* extra */);
-                }
-
-                for (size_t offset = 0;
-                        offset < accessUnit->size(); offset += 188) {
-                    status_t err = mTSParser->feedTSPacket(
-                             accessUnit->data() + offset, 188);
-
-                    if (err != OK) {
-                        notifyError(err);
-                        break;
-                    }
-                }
-
-                drainPackets(0 /* trackIndex */, ATSParser::VIDEO);
-                drainPackets(1 /* trackIndex */, ATSParser::AUDIO);
-            } else {
-                postAccessUnit(trackIndex, accessUnit, NULL);
-            }
-            break;
-        }
-
-        case RTPReceiver::kWhatPacketLost:
-        {
-            notifyPacketLost();
-            break;
-        }
-
-        default:
-            TRESPASS();
-    }
-}
-
-void MediaReceiver::drainPackets(
-        size_t trackIndex, ATSParser::SourceType type) {
-    sp<AnotherPacketSource> source =
-        static_cast<AnotherPacketSource *>(
-                mTSParser->getSource(type).get());
-
-    if (source == NULL) {
-        return;
-    }
-
-    sp<AMessage> format;
-    if (!(mFormatKnownMask & (1ul << trackIndex))) {
-        sp<MetaData> meta = source->getFormat();
-        CHECK(meta != NULL);
-
-        CHECK_EQ((status_t)OK, convertMetaDataToMessage(meta, &format));
-
-        mFormatKnownMask |= 1ul << trackIndex;
-    }
-
-    status_t finalResult;
-    while (source->hasBufferAvailable(&finalResult)) {
-        sp<ABuffer> accessUnit;
-        status_t err = source->dequeueAccessUnit(&accessUnit);
-        if (err == OK) {
-            postAccessUnit(trackIndex, accessUnit, format);
-            format.clear();
-        } else if (err != INFO_DISCONTINUITY) {
-            notifyError(err);
-        }
-    }
-
-    if (finalResult != OK) {
-        notifyError(finalResult);
-    }
-}
-
-void MediaReceiver::notifyInitDone(status_t err) {
-    sp<AMessage> notify = mNotify->dup();
-    notify->setInt32("what", kWhatInitDone);
-    notify->setInt32("err", err);
-    notify->post();
-}
-
-void MediaReceiver::notifyError(status_t err) {
-    sp<AMessage> notify = mNotify->dup();
-    notify->setInt32("what", kWhatError);
-    notify->setInt32("err", err);
-    notify->post();
-}
-
-void MediaReceiver::notifyPacketLost() {
-    sp<AMessage> notify = mNotify->dup();
-    notify->setInt32("what", kWhatPacketLost);
-    notify->post();
-}
-
-void MediaReceiver::postAccessUnit(
-        size_t trackIndex,
-        const sp<ABuffer> &accessUnit,
-        const sp<AMessage> &format) {
-    sp<AMessage> notify = mNotify->dup();
-    notify->setInt32("what", kWhatAccessUnit);
-    notify->setSize("trackIndex", trackIndex);
-    notify->setBuffer("accessUnit", accessUnit);
-
-    if (format != NULL) {
-        notify->setMessage("format", format);
-    }
-
-    notify->post();
-}
-
-status_t MediaReceiver::informSender(
-        size_t trackIndex, const sp<AMessage> &params) {
-    if (trackIndex >= mTrackInfos.size()) {
-        return -ERANGE;
-    }
-
-    TrackInfo *info = &mTrackInfos.editItemAt(trackIndex);
-    return info->mReceiver->informSender(params);
-}
-
-}  // namespace android
-
-

media/libstagefright/wifi-display/MediaReceiver.h

diff --git a/media/libstagefright/wifi-display/MediaReceiver.h b/media/libstagefright/wifi-display/MediaReceiver.h
deleted file mode 100644
index afbb407..0000000
--- a/media/libstagefright/wifi-display/MediaReceiver.h
+++ /dev/null
@@ -1,111 +0,0 @@
-/*
- * Copyright 2013, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <media/stagefright/foundation/AHandler.h>
-
-#include "ATSParser.h"
-#include "rtp/RTPReceiver.h"
-
-namespace android {
-
-struct ABuffer;
-struct ANetworkSession;
-struct AMessage;
-struct ATSParser;
-
-// This class facilitates receiving of media data for one or more tracks
-// over RTP. Either a 1:1 track to RTP channel mapping is used or a single
-// RTP channel provides the data for a transport stream that is consequently
-// demuxed and its track's data provided to the observer.
-struct MediaReceiver : public AHandler {
-    enum {
-        kWhatInitDone,
-        kWhatError,
-        kWhatAccessUnit,
-        kWhatPacketLost,
-    };
-
-    MediaReceiver(
-            const sp<ANetworkSession> &netSession,
-            const sp<AMessage> &notify);
-
-    ssize_t addTrack(
-            RTPReceiver::TransportMode rtpMode,
-            RTPReceiver::TransportMode rtcpMode,
-            int32_t *localRTPPort);
-
-    status_t connectTrack(
-            size_t trackIndex,
-            const char *remoteHost,
-            int32_t remoteRTPPort,
-            int32_t remoteRTCPPort);
-
-    enum Mode {
-        MODE_UNDEFINED,
-        MODE_TRANSPORT_STREAM,
-        MODE_TRANSPORT_STREAM_RAW,
-        MODE_ELEMENTARY_STREAMS,
-    };
-    status_t initAsync(Mode mode);
-
-    status_t informSender(size_t trackIndex, const sp<AMessage> &params);
-
-protected:
-    virtual void onMessageReceived(const sp<AMessage> &msg);
-    virtual ~MediaReceiver();
-
-private:
-    enum {
-        kWhatInit,
-        kWhatReceiverNotify,
-    };
-
-    struct TrackInfo {
-        sp<RTPReceiver> mReceiver;
-    };
-
-    sp<ANetworkSession> mNetSession;
-    sp<AMessage> mNotify;
-
-    Mode mMode;
-    int32_t mGeneration;
-
-    Vector<TrackInfo> mTrackInfos;
-
-    status_t mInitStatus;
-    size_t mInitDoneCount;
-
-    sp<ATSParser> mTSParser;
-    uint32_t mFormatKnownMask;
-
-    void onReceiverNotify(const sp<AMessage> &msg);
-
-    void drainPackets(size_t trackIndex, ATSParser::SourceType type);
-
-    void notifyInitDone(status_t err);
-    void notifyError(status_t err);
-    void notifyPacketLost();
-
-    void postAccessUnit(
-            size_t trackIndex,
-            const sp<ABuffer> &accessUnit,
-            const sp<AMessage> &format);
-
-    DISALLOW_EVIL_CONSTRUCTORS(MediaReceiver);
-};
-
-}  // namespace android
-

media/libstagefright/wifi-display/SNTPClient.cpp

diff --git a/media/libstagefright/wifi-display/SNTPClient.cpp b/media/libstagefright/wifi-display/SNTPClient.cpp
deleted file mode 100644
index 5c0af6a..0000000
--- a/media/libstagefright/wifi-display/SNTPClient.cpp
+++ /dev/null
@@ -1,174 +0,0 @@
-/*
- * Copyright 2013, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include "SNTPClient.h"
-
-#include <media/stagefright/foundation/ALooper.h>
-#include <media/stagefright/Utils.h>
-
-#include <arpa/inet.h>
-#include <netdb.h>
-#include <netinet/in.h>
-#include <sys/socket.h>
-#include <unistd.h>
-
-namespace android {
-
-SNTPClient::SNTPClient() {
-}
-
-status_t SNTPClient::requestTime(const char *host) {
-    struct hostent *ent;
-    int64_t requestTimeNTP, requestTimeUs;
-    ssize_t n;
-    int64_t responseTimeUs, responseTimeNTP;
-    int64_t originateTimeNTP, receiveTimeNTP, transmitTimeNTP;
-    int64_t roundTripTimeNTP, clockOffsetNTP;
-
-    status_t err = UNKNOWN_ERROR;
-
-    int s = socket(AF_INET, SOCK_DGRAM, 0);
-
-    if (s < 0) {
-        err = -errno;
-
-        goto bail;
-    }
-
-    ent = gethostbyname(host);
-
-    if (ent == NULL) {
-        err = -ENOENT;
-        goto bail2;
-    }
-
-    struct sockaddr_in hostAddr;
-    memset(hostAddr.sin_zero, 0, sizeof(hostAddr.sin_zero));
-    hostAddr.sin_family = AF_INET;
-    hostAddr.sin_port = htons(kNTPPort);
-    hostAddr.sin_addr.s_addr = *(in_addr_t *)ent->h_addr;
-
-    uint8_t packet[kNTPPacketSize];
-    memset(packet, 0, sizeof(packet));
-
-    packet[0] = kNTPModeClient | (kNTPVersion << 3);
-
-    requestTimeNTP = getNowNTP();
-    requestTimeUs = ALooper::GetNowUs();
-    writeTimeStamp(&packet[kNTPTransmitTimeOffset], requestTimeNTP);
-
-    n = sendto(
-            s, packet, sizeof(packet), 0,
-            (const struct sockaddr *)&hostAddr, sizeof(hostAddr));
-
-    if (n < 0) {
-        err = -errno;
-        goto bail2;
-    }
-
-    memset(packet, 0, sizeof(packet));
-
-    do {
-        n = recv(s, packet, sizeof(packet), 0);
-    } while (n < 0 && errno == EINTR);
-
-    if (n < 0) {
-        err = -errno;
-        goto bail2;
-    }
-
-    responseTimeUs = ALooper::GetNowUs();
-
-    responseTimeNTP = requestTimeNTP + makeNTP(responseTimeUs - requestTimeUs);
-
-    originateTimeNTP = readTimeStamp(&packet[kNTPOriginateTimeOffset]);
-    receiveTimeNTP = readTimeStamp(&packet[kNTPReceiveTimeOffset]);
-    transmitTimeNTP = readTimeStamp(&packet[kNTPTransmitTimeOffset]);
-
-    roundTripTimeNTP =
-        makeNTP(responseTimeUs - requestTimeUs)
-            - (transmitTimeNTP - receiveTimeNTP);
-
-    clockOffsetNTP =
-        ((receiveTimeNTP - originateTimeNTP)
-            + (transmitTimeNTP - responseTimeNTP)) / 2;
-
-    mTimeReferenceNTP = responseTimeNTP + clockOffsetNTP;
-    mTimeReferenceUs = responseTimeUs;
-    mRoundTripTimeNTP = roundTripTimeNTP;
-
-    err = OK;
-
-bail2:
-    close(s);
-    s = -1;
-
-bail:
-    return err;
-}
-
-int64_t SNTPClient::adjustTimeUs(int64_t timeUs) const {
-    uint64_t nowNTP =
-        mTimeReferenceNTP + makeNTP(timeUs - mTimeReferenceUs);
-
-    int64_t nowUs =
-        (nowNTP >> 32) * 1000000ll
-        + ((nowNTP & 0xffffffff) * 1000000ll) / (1ll << 32);
-
-    nowUs -= ((70ll * 365 + 17) * 24) * 60 * 60 * 1000000ll;
-
-    return nowUs;
-}
-
-// static
-void SNTPClient::writeTimeStamp(uint8_t *dst, uint64_t ntpTime) {
-    *dst++ = (ntpTime >> 56) & 0xff;
-    *dst++ = (ntpTime >> 48) & 0xff;
-    *dst++ = (ntpTime >> 40) & 0xff;
-    *dst++ = (ntpTime >> 32) & 0xff;
-    *dst++ = (ntpTime >> 24) & 0xff;
-    *dst++ = (ntpTime >> 16) & 0xff;
-    *dst++ = (ntpTime >> 8) & 0xff;
-    *dst++ = ntpTime & 0xff;
-}
-
-// static
-uint64_t SNTPClient::readTimeStamp(const uint8_t *dst) {
-    return U64_AT(dst);
-}
-
-// static
-uint64_t SNTPClient::getNowNTP() {
-    struct timeval tv;
-    gettimeofday(&tv, NULL /* time zone */);
-
-    uint64_t nowUs = tv.tv_sec * 1000000ll + tv.tv_usec;
-
-    nowUs += ((70ll * 365 + 17) * 24) * 60 * 60 * 1000000ll;
-
-    return makeNTP(nowUs);
-}
-
-// static
-uint64_t SNTPClient::makeNTP(uint64_t deltaUs) {
-    uint64_t hi = deltaUs / 1000000ll;
-    uint64_t lo = ((1ll << 32) * (deltaUs % 1000000ll)) / 1000000ll;
-
-    return (hi << 32) | lo;
-}
-
-}  // namespace android
-

media/libstagefright/wifi-display/SNTPClient.h

diff --git a/media/libstagefright/wifi-display/SNTPClient.h b/media/libstagefright/wifi-display/SNTPClient.h
deleted file mode 100644
index 967d1fc..0000000
--- a/media/libstagefright/wifi-display/SNTPClient.h
+++ /dev/null
@@ -1,62 +0,0 @@
-/*
- * Copyright 2013, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef SNTP_CLIENT_H_
-
-#define SNTP_CLIENT_H_
-
-#include <media/stagefright/foundation/ABase.h>
-#include <utils/Errors.h>
-
-namespace android {
-
-// Implementation of the SNTP (Simple Network Time Protocol)
-struct SNTPClient {
-    SNTPClient();
-
-    status_t requestTime(const char *host);
-
-    // given a time obtained from ALooper::GetNowUs()
-    // return the number of us elapsed since Jan 1 1970 00:00:00 (UTC).
-    int64_t adjustTimeUs(int64_t timeUs) const;
-
-private:
-    enum {
-        kNTPPort = 123,
-        kNTPPacketSize = 48,
-        kNTPModeClient = 3,
-        kNTPVersion = 3,
-        kNTPTransmitTimeOffset = 40,
-        kNTPOriginateTimeOffset = 24,
-        kNTPReceiveTimeOffset = 32,
-    };
-
-    uint64_t mTimeReferenceNTP;
-    int64_t mTimeReferenceUs;
-    int64_t mRoundTripTimeNTP;
-
-    static void writeTimeStamp(uint8_t *dst, uint64_t ntpTime);
-    static uint64_t readTimeStamp(const uint8_t *dst);
-
-    static uint64_t getNowNTP();
-    static uint64_t makeNTP(uint64_t deltaUs);
-
-    DISALLOW_EVIL_CONSTRUCTORS(SNTPClient);
-};
-
-}  // namespace android
-
-#endif  // SNTP_CLIENT_H_

media/libstagefright/wifi-display/TimeSyncer.cpp

diff --git a/media/libstagefright/wifi-display/TimeSyncer.cpp b/media/libstagefright/wifi-display/TimeSyncer.cpp
deleted file mode 100644
index 0f4d93a..0000000
--- a/media/libstagefright/wifi-display/TimeSyncer.cpp
+++ /dev/null
@@ -1,337 +0,0 @@
-/*
- * Copyright 2013, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NEBUG 0
-#define LOG_TAG "TimeSyncer"
-#include <utils/Log.h>
-
-#include "TimeSyncer.h"
-
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AHandler.h>
-#include <media/stagefright/foundation/ALooper.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/foundation/ANetworkSession.h>
-#include <media/stagefright/Utils.h>
-
-namespace android {
-
-TimeSyncer::TimeSyncer(
-        const sp<ANetworkSession> &netSession, const sp<AMessage> &notify)
-    : mNetSession(netSession),
-      mNotify(notify),
-      mIsServer(false),
-      mConnected(false),
-      mUDPSession(0),
-      mSeqNo(0),
-      mTotalTimeUs(0.0),
-      mPendingT1(0ll),
-      mTimeoutGeneration(0) {
-}
-
-TimeSyncer::~TimeSyncer() {
-}
-
-void TimeSyncer::startServer(unsigned localPort) {
-    sp<AMessage> msg = new AMessage(kWhatStartServer, id());
-    msg->setInt32("localPort", localPort);
-    msg->post();
-}
-
-void TimeSyncer::startClient(const char *remoteHost, unsigned remotePort) {
-    sp<AMessage> msg = new AMessage(kWhatStartClient, id());
-    msg->setString("remoteHost", remoteHost);
-    msg->setInt32("remotePort", remotePort);
-    msg->post();
-}
-
-void TimeSyncer::onMessageReceived(const sp<AMessage> &msg) {
-    switch (msg->what()) {
-        case kWhatStartClient:
-        {
-            AString remoteHost;
-            CHECK(msg->findString("remoteHost", &remoteHost));
-
-            int32_t remotePort;
-            CHECK(msg->findInt32("remotePort", &remotePort));
-
-            sp<AMessage> notify = new AMessage(kWhatUDPNotify, id());
-
-            CHECK_EQ((status_t)OK,
-                     mNetSession->createUDPSession(
-                         0 /* localPort */,
-                         remoteHost.c_str(),
-                         remotePort,
-                         notify,
-                         &mUDPSession));
-
-            postSendPacket();
-            break;
-        }
-
-        case kWhatStartServer:
-        {
-            mIsServer = true;
-
-            int32_t localPort;
-            CHECK(msg->findInt32("localPort", &localPort));
-
-            sp<AMessage> notify = new AMessage(kWhatUDPNotify, id());
-
-            CHECK_EQ((status_t)OK,
-                     mNetSession->createUDPSession(
-                         localPort, notify, &mUDPSession));
-
-            break;
-        }
-
-        case kWhatSendPacket:
-        {
-            if (mHistory.size() == 0) {
-                ALOGI("starting batch");
-            }
-
-            TimeInfo ti;
-            memset(&ti, 0, sizeof(ti));
-
-            ti.mT1 = ALooper::GetNowUs();
-
-            CHECK_EQ((status_t)OK,
-                     mNetSession->sendRequest(
-                         mUDPSession, &ti, sizeof(ti)));
-
-            mPendingT1 = ti.mT1;
-            postTimeout();
-            break;
-        }
-
-        case kWhatTimedOut:
-        {
-            int32_t generation;
-            CHECK(msg->findInt32("generation", &generation));
-
-            if (generation != mTimeoutGeneration) {
-                break;
-            }
-
-            ALOGI("timed out, sending another request");
-            postSendPacket();
-            break;
-        }
-
-        case kWhatUDPNotify:
-        {
-            int32_t reason;
-            CHECK(msg->findInt32("reason", &reason));
-
-            switch (reason) {
-                case ANetworkSession::kWhatError:
-                {
-                    int32_t sessionID;
-                    CHECK(msg->findInt32("sessionID", &sessionID));
-
-                    int32_t err;
-                    CHECK(msg->findInt32("err", &err));
-
-                    AString detail;
-                    CHECK(msg->findString("detail", &detail));
-
-                    ALOGE("An error occurred in session %d (%d, '%s/%s').",
-                          sessionID,
-                          err,
-                          detail.c_str(),
-                          strerror(-err));
-
-                    mNetSession->destroySession(sessionID);
-
-                    cancelTimeout();
-
-                    notifyError(err);
-                    break;
-                }
-
-                case ANetworkSession::kWhatDatagram:
-                {
-                    int32_t sessionID;
-                    CHECK(msg->findInt32("sessionID", &sessionID));
-
-                    sp<ABuffer> packet;
-                    CHECK(msg->findBuffer("data", &packet));
-
-                    int64_t arrivalTimeUs;
-                    CHECK(packet->meta()->findInt64(
-                                "arrivalTimeUs", &arrivalTimeUs));
-
-                    CHECK_EQ(packet->size(), sizeof(TimeInfo));
-
-                    TimeInfo *ti = (TimeInfo *)packet->data();
-
-                    if (mIsServer) {
-                        if (!mConnected) {
-                            AString fromAddr;
-                            CHECK(msg->findString("fromAddr", &fromAddr));
-
-                            int32_t fromPort;
-                            CHECK(msg->findInt32("fromPort", &fromPort));
-
-                            CHECK_EQ((status_t)OK,
-                                     mNetSession->connectUDPSession(
-                                         mUDPSession, fromAddr.c_str(), fromPort));
-
-                            mConnected = true;
-                        }
-
-                        ti->mT2 = arrivalTimeUs;
-                        ti->mT3 = ALooper::GetNowUs();
-
-                        CHECK_EQ((status_t)OK,
-                                 mNetSession->sendRequest(
-                                     mUDPSession, ti, sizeof(*ti)));
-                    } else {
-                        if (ti->mT1 != mPendingT1) {
-                            break;
-                        }
-
-                        cancelTimeout();
-                        mPendingT1 = 0;
-
-                        ti->mT4 = arrivalTimeUs;
-
-                        // One way delay for a packet to travel from client
-                        // to server or back (assumed to be the same either way).
-                        int64_t delay =
-                            (ti->mT2 - ti->mT1 + ti->mT4 - ti->mT3) / 2;
-
-                        // Offset between the client clock (T1, T4) and the
-                        // server clock (T2, T3) timestamps.
-                        int64_t offset =
-                            (ti->mT2 - ti->mT1 - ti->mT4 + ti->mT3) / 2;
-
-                        mHistory.push_back(*ti);
-
-                        ALOGV("delay = %lld us,\toffset %lld us",
-                               delay,
-                               offset);
-
-                        if (mHistory.size() < kNumPacketsPerBatch) {
-                            postSendPacket(1000000ll / 30);
-                        } else {
-                            notifyOffset();
-
-                            ALOGI("batch done");
-
-                            mHistory.clear();
-                            postSendPacket(kBatchDelayUs);
-                        }
-                    }
-                    break;
-                }
-
-                default:
-                    TRESPASS();
-            }
-
-            break;
-        }
-
-        default:
-            TRESPASS();
-    }
-}
-
-void TimeSyncer::postSendPacket(int64_t delayUs) {
-    (new AMessage(kWhatSendPacket, id()))->post(delayUs);
-}
-
-void TimeSyncer::postTimeout() {
-    sp<AMessage> msg = new AMessage(kWhatTimedOut, id());
-    msg->setInt32("generation", mTimeoutGeneration);
-    msg->post(kTimeoutDelayUs);
-}
-
-void TimeSyncer::cancelTimeout() {
-    ++mTimeoutGeneration;
-}
-
-void TimeSyncer::notifyError(status_t err) {
-    if (mNotify == NULL) {
-        looper()->stop();
-        return;
-    }
-
-    sp<AMessage> notify = mNotify->dup();
-    notify->setInt32("what", kWhatError);
-    notify->setInt32("err", err);
-    notify->post();
-}
-
-// static
-int TimeSyncer::CompareRountripTime(const TimeInfo *ti1, const TimeInfo *ti2) {
-    int64_t rt1 = ti1->mT4 - ti1->mT1;
-    int64_t rt2 = ti2->mT4 - ti2->mT1;
-
-    if (rt1 < rt2) {
-        return -1;
-    } else if (rt1 > rt2) {
-        return 1;
-    }
-
-    return 0;
-}
-
-void TimeSyncer::notifyOffset() {
-    mHistory.sort(CompareRountripTime);
-
-    int64_t sum = 0ll;
-    size_t count = 0;
-
-    // Only consider the third of the information associated with the best
-    // (smallest) roundtrip times.
-    for (size_t i = 0; i < mHistory.size() / 3; ++i) {
-        const TimeInfo *ti = &mHistory[i];
-
-#if 0
-        // One way delay for a packet to travel from client
-        // to server or back (assumed to be the same either way).
-        int64_t delay =
-            (ti->mT2 - ti->mT1 + ti->mT4 - ti->mT3) / 2;
-#endif
-
-        // Offset between the client clock (T1, T4) and the
-        // server clock (T2, T3) timestamps.
-        int64_t offset =
-            (ti->mT2 - ti->mT1 - ti->mT4 + ti->mT3) / 2;
-
-        ALOGV("(%d) RT: %lld us, offset: %lld us",
-              i, ti->mT4 - ti->mT1, offset);
-
-        sum += offset;
-        ++count;
-    }
-
-    if (mNotify == NULL) {
-        ALOGI("avg. offset is %lld", sum / count);
-        return;
-    }
-
-    sp<AMessage> notify = mNotify->dup();
-    notify->setInt32("what", kWhatTimeOffset);
-    notify->setInt64("offset", sum / count);
-    notify->post();
-}
-
-}  // namespace android

media/libstagefright/wifi-display/TimeSyncer.h

diff --git a/media/libstagefright/wifi-display/TimeSyncer.h b/media/libstagefright/wifi-display/TimeSyncer.h
deleted file mode 100644
index 4e7571f..0000000
--- a/media/libstagefright/wifi-display/TimeSyncer.h
+++ /dev/null
@@ -1,109 +0,0 @@
-/*
- * Copyright 2013, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef TIME_SYNCER_H_
-
-#define TIME_SYNCER_H_
-
-#include <media/stagefright/foundation/AHandler.h>
-
-namespace android {
-
-struct ANetworkSession;
-
-/*
-   TimeSyncer allows us to synchronize time between a client and a server.
-   The client sends a UDP packet containing its send-time to the server,
-   the server sends that packet back to the client amended with information
-   about when it was received as well as the time the reply was sent back.
-   Finally the client receives the reply and has now enough information to
-   compute the clock offset between client and server assuming that packet
-   exchange is symmetric, i.e. time for a packet client->server and
-   server->client is roughly equal.
-   This exchange is repeated a number of times and the average offset computed
-   over the 30% of packets that had the lowest roundtrip times.
-   The offset is determined every 10 secs to account for slight differences in
-   clock frequency.
-*/
-struct TimeSyncer : public AHandler {
-    enum {
-        kWhatError,
-        kWhatTimeOffset,
-    };
-    TimeSyncer(
-            const sp<ANetworkSession> &netSession,
-            const sp<AMessage> &notify);
-
-    void startServer(unsigned localPort);
-    void startClient(const char *remoteHost, unsigned remotePort);
-
-protected:
-    virtual ~TimeSyncer();
-
-    virtual void onMessageReceived(const sp<AMessage> &msg);
-
-private:
-    enum {
-        kWhatStartServer,
-        kWhatStartClient,
-        kWhatUDPNotify,
-        kWhatSendPacket,
-        kWhatTimedOut,
-    };
-
-    struct TimeInfo {
-        int64_t mT1;  // client timestamp at send
-        int64_t mT2;  // server timestamp at receive
-        int64_t mT3;  // server timestamp at send
-        int64_t mT4;  // client timestamp at receive
-    };
-
-    enum {
-        kNumPacketsPerBatch = 30,
-    };
-    static const int64_t kTimeoutDelayUs = 500000ll;
-    static const int64_t kBatchDelayUs = 60000000ll;  // every minute
-
-    sp<ANetworkSession> mNetSession;
-    sp<AMessage> mNotify;
-
-    bool mIsServer;
-    bool mConnected;
-    int32_t mUDPSession;
-    uint32_t mSeqNo;
-    double mTotalTimeUs;
-
-    Vector<TimeInfo> mHistory;
-
-    int64_t mPendingT1;
-    int32_t mTimeoutGeneration;
-
-    void postSendPacket(int64_t delayUs = 0ll);
-
-    void postTimeout();
-    void cancelTimeout();
-
-    void notifyError(status_t err);
-    void notifyOffset();
-
-    static int CompareRountripTime(const TimeInfo *ti1, const TimeInfo *ti2);
-
-    DISALLOW_EVIL_CONSTRUCTORS(TimeSyncer);
-};
-
-}  // namespace android
-
-#endif  // TIME_SYNCER_H_

media/libstagefright/wifi-display/nettest.cpp

diff --git a/media/libstagefright/wifi-display/nettest.cpp b/media/libstagefright/wifi-display/nettest.cpp
deleted file mode 100644
index 73c0d80..0000000
--- a/media/libstagefright/wifi-display/nettest.cpp
+++ /dev/null
@@ -1,400 +0,0 @@
-/*
- * Copyright 2013, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NEBUG 0
-#define LOG_TAG "nettest"
-#include <utils/Log.h>
-
-#include "TimeSyncer.h"
-
-#include <binder/ProcessState.h>
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AHandler.h>
-#include <media/stagefright/foundation/ALooper.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/foundation/ANetworkSession.h>
-#include <media/stagefright/foundation/hexdump.h>
-#include <media/stagefright/DataSource.h>
-#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/NuMediaExtractor.h>
-#include <media/stagefright/Utils.h>
-
-namespace android {
-
-struct TestHandler : public AHandler {
-    TestHandler(const sp<ANetworkSession> &netSession);
-
-    void listen(int32_t port);
-    void connect(const char *host, int32_t port);
-
-protected:
-    virtual ~TestHandler();
-    virtual void onMessageReceived(const sp<AMessage> &msg);
-
-private:
-    enum {
-        kTimeSyncerPort = 8123,
-    };
-
-    enum {
-        kWhatListen,
-        kWhatConnect,
-        kWhatTimeSyncerNotify,
-        kWhatNetNotify,
-        kWhatSendMore,
-        kWhatStop,
-    };
-
-    sp<ANetworkSession> mNetSession;
-    sp<TimeSyncer> mTimeSyncer;
-
-    int32_t mServerSessionID;
-    int32_t mSessionID;
-
-    int64_t mTimeOffsetUs;
-    bool mTimeOffsetValid;
-
-    int32_t mCounter;
-
-    int64_t mMaxDelayMs;
-
-    void dumpDelay(int32_t counter, int64_t delayMs);
-
-    DISALLOW_EVIL_CONSTRUCTORS(TestHandler);
-};
-
-TestHandler::TestHandler(const sp<ANetworkSession> &netSession)
-    : mNetSession(netSession),
-      mServerSessionID(0),
-      mSessionID(0),
-      mTimeOffsetUs(-1ll),
-      mTimeOffsetValid(false),
-      mCounter(0),
-      mMaxDelayMs(-1ll) {
-}
-
-TestHandler::~TestHandler() {
-}
-
-void TestHandler::listen(int32_t port) {
-    sp<AMessage> msg = new AMessage(kWhatListen, id());
-    msg->setInt32("port", port);
-    msg->post();
-}
-
-void TestHandler::connect(const char *host, int32_t port) {
-    sp<AMessage> msg = new AMessage(kWhatConnect, id());
-    msg->setString("host", host);
-    msg->setInt32("port", port);
-    msg->post();
-}
-
-void TestHandler::dumpDelay(int32_t counter, int64_t delayMs) {
-    static const int64_t kMinDelayMs = 0;
-    static const int64_t kMaxDelayMs = 300;
-
-    const char *kPattern = "########################################";
-    size_t kPatternSize = strlen(kPattern);
-
-    int n = (kPatternSize * (delayMs - kMinDelayMs))
-                / (kMaxDelayMs - kMinDelayMs);
-
-    if (n < 0) {
-        n = 0;
-    } else if ((size_t)n > kPatternSize) {
-        n = kPatternSize;
-    }
-
-    if (delayMs > mMaxDelayMs) {
-        mMaxDelayMs = delayMs;
-    }
-
-    ALOGI("[%d] (%4lld ms / %4lld ms) %s",
-          counter,
-          delayMs,
-          mMaxDelayMs,
-          kPattern + kPatternSize - n);
-}
-
-void TestHandler::onMessageReceived(const sp<AMessage> &msg) {
-    switch (msg->what()) {
-        case kWhatListen:
-        {
-            sp<AMessage> notify = new AMessage(kWhatTimeSyncerNotify, id());
-            mTimeSyncer = new TimeSyncer(mNetSession, notify);
-            looper()->registerHandler(mTimeSyncer);
-
-            notify = new AMessage(kWhatNetNotify, id());
-
-            int32_t port;
-            CHECK(msg->findInt32("port", &port));
-
-            struct in_addr ifaceAddr;
-            ifaceAddr.s_addr = INADDR_ANY;
-
-            CHECK_EQ((status_t)OK,
-                     mNetSession->createTCPDatagramSession(
-                         ifaceAddr,
-                         port,
-                         notify,
-                         &mServerSessionID));
-            break;
-        }
-
-        case kWhatConnect:
-        {
-            sp<AMessage> notify = new AMessage(kWhatTimeSyncerNotify, id());
-            mTimeSyncer = new TimeSyncer(mNetSession, notify);
-            looper()->registerHandler(mTimeSyncer);
-            mTimeSyncer->startServer(kTimeSyncerPort);
-
-            AString host;
-            CHECK(msg->findString("host", &host));
-
-            int32_t port;
-            CHECK(msg->findInt32("port", &port));
-
-            notify = new AMessage(kWhatNetNotify, id());
-
-            CHECK_EQ((status_t)OK,
-                     mNetSession->createTCPDatagramSession(
-                         0 /* localPort */,
-                         host.c_str(),
-                         port,
-                         notify,
-                         &mSessionID));
-            break;
-        }
-
-        case kWhatNetNotify:
-        {
-            int32_t reason;
-            CHECK(msg->findInt32("reason", &reason));
-
-            switch (reason) {
-                case ANetworkSession::kWhatConnected:
-                {
-                    ALOGI("kWhatConnected");
-
-                    (new AMessage(kWhatSendMore, id()))->post();
-                    break;
-                }
-
-                case ANetworkSession::kWhatClientConnected:
-                {
-                    ALOGI("kWhatClientConnected");
-
-                    CHECK_EQ(mSessionID, 0);
-                    CHECK(msg->findInt32("sessionID", &mSessionID));
-
-                    AString clientIP;
-                    CHECK(msg->findString("client-ip", &clientIP));
-
-                    mTimeSyncer->startClient(clientIP.c_str(), kTimeSyncerPort);
-                    break;
-                }
-
-                case ANetworkSession::kWhatDatagram:
-                {
-                    sp<ABuffer> packet;
-                    CHECK(msg->findBuffer("data", &packet));
-
-                    CHECK_EQ(packet->size(), 12u);
-
-                    int32_t counter = U32_AT(packet->data());
-                    int64_t timeUs = U64_AT(packet->data() + 4);
-
-                    if (mTimeOffsetValid) {
-                        timeUs -= mTimeOffsetUs;
-                        int64_t nowUs = ALooper::GetNowUs();
-                        int64_t delayMs = (nowUs - timeUs) / 1000ll;
-
-                        dumpDelay(counter, delayMs);
-                    } else {
-                        ALOGI("received %d", counter);
-                    }
-                    break;
-                }
-
-                case ANetworkSession::kWhatError:
-                {
-                    ALOGE("kWhatError");
-                    break;
-                }
-
-                default:
-                    TRESPASS();
-            }
-            break;
-        }
-
-        case kWhatTimeSyncerNotify:
-        {
-            CHECK(msg->findInt64("offset", &mTimeOffsetUs));
-            mTimeOffsetValid = true;
-            break;
-        }
-
-        case kWhatSendMore:
-        {
-            uint8_t buffer[4 + 8];
-            buffer[0] = mCounter >> 24;
-            buffer[1] = (mCounter >> 16) & 0xff;
-            buffer[2] = (mCounter >> 8) & 0xff;
-            buffer[3] = mCounter & 0xff;
-
-            int64_t nowUs = ALooper::GetNowUs();
-
-            buffer[4] = nowUs >> 56;
-            buffer[5] = (nowUs >> 48) & 0xff;
-            buffer[6] = (nowUs >> 40) & 0xff;
-            buffer[7] = (nowUs >> 32) & 0xff;
-            buffer[8] = (nowUs >> 24) & 0xff;
-            buffer[9] = (nowUs >> 16) & 0xff;
-            buffer[10] = (nowUs >> 8) & 0xff;
-            buffer[11] = nowUs & 0xff;
-
-            ++mCounter;
-
-            CHECK_EQ((status_t)OK,
-                     mNetSession->sendRequest(
-                         mSessionID,
-                         buffer,
-                         sizeof(buffer),
-                         true /* timeValid */,
-                         nowUs));
-
-            msg->post(100000ll);
-            break;
-        }
-
-        case kWhatStop:
-        {
-            if (mSessionID != 0) {
-                mNetSession->destroySession(mSessionID);
-                mSessionID = 0;
-            }
-
-            if (mServerSessionID != 0) {
-                mNetSession->destroySession(mServerSessionID);
-                mServerSessionID = 0;
-            }
-
-            looper()->stop();
-            break;
-        }
-
-        default:
-            TRESPASS();
-    }
-}
-
-}  // namespace android
-
-static void usage(const char *me) {
-    fprintf(stderr,
-            "usage: %s -c host:port\tconnect to remote host\n"
-            "               -l port   \tlisten\n",
-            me);
-}
-
-int main(int argc, char **argv) {
-    using namespace android;
-
-    // srand(time(NULL));
-
-    ProcessState::self()->startThreadPool();
-
-    DataSource::RegisterDefaultSniffers();
-
-    int32_t connectToPort = -1;
-    AString connectToHost;
-
-    int32_t listenOnPort = -1;
-
-    int res;
-    while ((res = getopt(argc, argv, "hc:l:")) >= 0) {
-        switch (res) {
-            case 'c':
-            {
-                const char *colonPos = strrchr(optarg, ':');
-
-                if (colonPos == NULL) {
-                    usage(argv[0]);
-                    exit(1);
-                }
-
-                connectToHost.setTo(optarg, colonPos - optarg);
-
-                char *end;
-                connectToPort = strtol(colonPos + 1, &end, 10);
-
-                if (*end != '\0' || end == colonPos + 1
-                        || connectToPort < 0 || connectToPort > 65535) {
-                    fprintf(stderr, "Illegal port specified.\n");
-                    exit(1);
-                }
-                break;
-            }
-
-            case 'l':
-            {
-                char *end;
-                listenOnPort = strtol(optarg, &end, 10);
-
-                if (*end != '\0' || end == optarg
-                        || listenOnPort < 0 || listenOnPort > 65535) {
-                    fprintf(stderr, "Illegal port specified.\n");
-                    exit(1);
-                }
-                break;
-            }
-
-            case '?':
-            case 'h':
-                usage(argv[0]);
-                exit(1);
-        }
-    }
-
-    if ((listenOnPort < 0 && connectToPort < 0)
-            || (listenOnPort >= 0 && connectToPort >= 0)) {
-        fprintf(stderr,
-                "You need to select either client or server mode.\n");
-        exit(1);
-    }
-
-    sp<ANetworkSession> netSession = new ANetworkSession;
-    netSession->start();
-
-    sp<ALooper> looper = new ALooper;
-
-    sp<TestHandler> handler = new TestHandler(netSession);
-    looper->registerHandler(handler);
-
-    if (listenOnPort) {
-        handler->listen(listenOnPort);
-    }
-
-    if (connectToPort >= 0) {
-        handler->connect(connectToHost.c_str(), connectToPort);
-    }
-
-    looper->start(true /* runOnCallingThread */);
-
-    return 0;
-}

media/libstagefright/wifi-display/rtp/RTPAssembler.cpp

diff --git a/media/libstagefright/wifi-display/rtp/RTPAssembler.cpp b/media/libstagefright/wifi-display/rtp/RTPAssembler.cpp
deleted file mode 100644
index 7a96081..0000000
--- a/media/libstagefright/wifi-display/rtp/RTPAssembler.cpp
+++ /dev/null
@@ -1,328 +0,0 @@
-/*
- * Copyright 2013, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "RTPAssembler"
-#include <utils/Log.h>
-
-#include "RTPAssembler.h"
-
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/foundation/hexdump.h>
-#include <media/stagefright/MediaErrors.h>
-
-namespace android {
-
-RTPReceiver::Assembler::Assembler(const sp<AMessage> &notify)
-    : mNotify(notify) {
-}
-
-void RTPReceiver::Assembler::postAccessUnit(
-        const sp<ABuffer> &accessUnit, bool followsDiscontinuity) {
-    sp<AMessage> notify = mNotify->dup();
-    notify->setInt32("what", RTPReceiver::kWhatAccessUnit);
-    notify->setBuffer("accessUnit", accessUnit);
-    notify->setInt32("followsDiscontinuity", followsDiscontinuity);
-    notify->post();
-}
-////////////////////////////////////////////////////////////////////////////////
-
-RTPReceiver::TSAssembler::TSAssembler(const sp<AMessage> &notify)
-    : Assembler(notify),
-      mSawDiscontinuity(false) {
-}
-
-void RTPReceiver::TSAssembler::signalDiscontinuity() {
-    mSawDiscontinuity = true;
-}
-
-status_t RTPReceiver::TSAssembler::processPacket(const sp<ABuffer> &packet) {
-    int32_t rtpTime;
-    CHECK(packet->meta()->findInt32("rtp-time", &rtpTime));
-
-    packet->meta()->setInt64("timeUs", (rtpTime * 100ll) / 9);
-
-    postAccessUnit(packet, mSawDiscontinuity);
-
-    if (mSawDiscontinuity) {
-        mSawDiscontinuity = false;
-    }
-
-    return OK;
-}
-
-////////////////////////////////////////////////////////////////////////////////
-
-RTPReceiver::H264Assembler::H264Assembler(const sp<AMessage> &notify)
-    : Assembler(notify),
-      mState(0),
-      mIndicator(0),
-      mNALType(0),
-      mAccessUnitRTPTime(0) {
-}
-
-void RTPReceiver::H264Assembler::signalDiscontinuity() {
-    reset();
-}
-
-status_t RTPReceiver::H264Assembler::processPacket(const sp<ABuffer> &packet) {
-    status_t err = internalProcessPacket(packet);
-
-    if (err != OK) {
-        reset();
-    }
-
-    return err;
-}
-
-status_t RTPReceiver::H264Assembler::internalProcessPacket(
-        const sp<ABuffer> &packet) {
-    const uint8_t *data = packet->data();
-    size_t size = packet->size();
-
-    switch (mState) {
-        case 0:
-        {
-            if (size < 1 || (data[0] & 0x80)) {
-                ALOGV("Malformed H264 RTP packet (empty or F-bit set)");
-                return ERROR_MALFORMED;
-            }
-
-            unsigned nalType = data[0] & 0x1f;
-            if (nalType >= 1 && nalType <= 23) {
-                addSingleNALUnit(packet);
-                ALOGV("added single NAL packet");
-            } else if (nalType == 28) {
-                // FU-A
-                unsigned indicator = data[0];
-                CHECK((indicator & 0x1f) == 28);
-
-                if (size < 2) {
-                    ALOGV("Malformed H264 FU-A packet (single byte)");
-                    return ERROR_MALFORMED;
-                }
-
-                if (!(data[1] & 0x80)) {
-                    ALOGV("Malformed H264 FU-A packet (no start bit)");
-                    return ERROR_MALFORMED;
-                }
-
-                mIndicator = data[0];
-                mNALType = data[1] & 0x1f;
-                uint32_t nri = (data[0] >> 5) & 3;
-
-                clearAccumulator();
-
-                uint8_t byte = mNALType | (nri << 5);
-                appendToAccumulator(&byte, 1);
-                appendToAccumulator(data + 2, size - 2);
-
-                int32_t rtpTime;
-                CHECK(packet->meta()->findInt32("rtp-time", &rtpTime));
-                mAccumulator->meta()->setInt32("rtp-time", rtpTime);
-
-                if (data[1] & 0x40) {
-                    // Huh? End bit also set on the first buffer.
-                    addSingleNALUnit(mAccumulator);
-                    clearAccumulator();
-
-                    ALOGV("added FU-A");
-                    break;
-                }
-
-                mState = 1;
-            } else if (nalType == 24) {
-                // STAP-A
-
-                status_t err = addSingleTimeAggregationPacket(packet);
-                if (err != OK) {
-                    return err;
-                }
-            } else {
-                ALOGV("Malformed H264 packet (unknown type %d)", nalType);
-                return ERROR_UNSUPPORTED;
-            }
-            break;
-        }
-
-        case 1:
-        {
-            if (size < 2
-                    || data[0] != mIndicator
-                    || (data[1] & 0x1f) != mNALType
-                    || (data[1] & 0x80)) {
-                ALOGV("Malformed H264 FU-A packet (indicator, "
-                      "type or start bit mismatch)");
-
-                return ERROR_MALFORMED;
-            }
-
-            appendToAccumulator(data + 2, size - 2);
-
-            if (data[1] & 0x40) {
-                addSingleNALUnit(mAccumulator);
-
-                clearAccumulator();
-                mState = 0;
-
-                ALOGV("added FU-A");
-            }
-            break;
-        }
-
-        default:
-            TRESPASS();
-    }
-
-    int32_t marker;
-    CHECK(packet->meta()->findInt32("M", &marker));
-
-    if (marker) {
-        flushAccessUnit();
-    }
-
-    return OK;
-}
-
-void RTPReceiver::H264Assembler::reset() {
-    mNALUnits.clear();
-
-    clearAccumulator();
-    mState = 0;
-}
-
-void RTPReceiver::H264Assembler::clearAccumulator() {
-    if (mAccumulator != NULL) {
-        // XXX Too expensive.
-        mAccumulator.clear();
-    }
-}
-
-void RTPReceiver::H264Assembler::appendToAccumulator(
-        const void *data, size_t size) {
-    if (mAccumulator == NULL) {
-        mAccumulator = new ABuffer(size);
-        memcpy(mAccumulator->data(), data, size);
-        return;
-    }
-
-    if (mAccumulator->size() + size > mAccumulator->capacity()) {
-        sp<ABuffer> buf = new ABuffer(mAccumulator->size() + size);
-        memcpy(buf->data(), mAccumulator->data(), mAccumulator->size());
-        buf->setRange(0, mAccumulator->size());
-
-        int32_t rtpTime;
-        if (mAccumulator->meta()->findInt32("rtp-time", &rtpTime)) {
-            buf->meta()->setInt32("rtp-time", rtpTime);
-        }
-
-        mAccumulator = buf;
-    }
-
-    memcpy(mAccumulator->data() + mAccumulator->size(), data, size);
-    mAccumulator->setRange(0, mAccumulator->size() + size);
-}
-
-void RTPReceiver::H264Assembler::addSingleNALUnit(const sp<ABuffer> &packet) {
-    if (mNALUnits.empty()) {
-        int32_t rtpTime;
-        CHECK(packet->meta()->findInt32("rtp-time", &rtpTime));
-
-        mAccessUnitRTPTime = rtpTime;
-    }
-
-    mNALUnits.push_back(packet);
-}
-
-void RTPReceiver::H264Assembler::flushAccessUnit() {
-    if (mNALUnits.empty()) {
-        return;
-    }
-
-    size_t totalSize = 0;
-    for (List<sp<ABuffer> >::iterator it = mNALUnits.begin();
-            it != mNALUnits.end(); ++it) {
-        totalSize += 4 + (*it)->size();
-    }
-
-    sp<ABuffer> accessUnit = new ABuffer(totalSize);
-    size_t offset = 0;
-    for (List<sp<ABuffer> >::iterator it = mNALUnits.begin();
-            it != mNALUnits.end(); ++it) {
-        const sp<ABuffer> nalUnit = *it;
-
-        memcpy(accessUnit->data() + offset, "\x00\x00\x00\x01", 4);
-
-        memcpy(accessUnit->data() + offset + 4,
-               nalUnit->data(),
-               nalUnit->size());
-
-        offset += 4 + nalUnit->size();
-    }
-
-    mNALUnits.clear();
-
-    accessUnit->meta()->setInt64("timeUs", mAccessUnitRTPTime * 100ll / 9ll);
-    postAccessUnit(accessUnit, false /* followsDiscontinuity */);
-}
-
-status_t RTPReceiver::H264Assembler::addSingleTimeAggregationPacket(
-        const sp<ABuffer> &packet) {
-    const uint8_t *data = packet->data();
-    size_t size = packet->size();
-
-    if (size < 3) {
-        ALOGV("Malformed H264 STAP-A packet (too small)");
-        return ERROR_MALFORMED;
-    }
-
-    int32_t rtpTime;
-    CHECK(packet->meta()->findInt32("rtp-time", &rtpTime));
-
-    ++data;
-    --size;
-    while (size >= 2) {
-        size_t nalSize = (data[0] << 8) | data[1];
-
-        if (size < nalSize + 2) {
-            ALOGV("Malformed H264 STAP-A packet (incomplete NAL unit)");
-            return ERROR_MALFORMED;
-        }
-
-        sp<ABuffer> unit = new ABuffer(nalSize);
-        memcpy(unit->data(), &data[2], nalSize);
-
-        unit->meta()->setInt32("rtp-time", rtpTime);
-
-        addSingleNALUnit(unit);
-
-        data += 2 + nalSize;
-        size -= 2 + nalSize;
-    }
-
-    if (size != 0) {
-        ALOGV("Unexpected padding at end of STAP-A packet.");
-    }
-
-    ALOGV("added STAP-A");
-
-    return OK;
-}
-
-}  // namespace android
-

media/libstagefright/wifi-display/rtp/RTPAssembler.h

diff --git a/media/libstagefright/wifi-display/rtp/RTPAssembler.h b/media/libstagefright/wifi-display/rtp/RTPAssembler.h
deleted file mode 100644
index e456d32..0000000
--- a/media/libstagefright/wifi-display/rtp/RTPAssembler.h
+++ /dev/null
@@ -1,92 +0,0 @@
-/*
- * Copyright 2013, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef RTP_ASSEMBLER_H_
-
-#define RTP_ASSEMBLER_H_
-
-#include "RTPReceiver.h"
-
-namespace android {
-
-// A helper class to reassemble the payload of RTP packets into access
-// units depending on the packetization scheme.
-struct RTPReceiver::Assembler : public RefBase {
-    Assembler(const sp<AMessage> &notify);
-
-    virtual void signalDiscontinuity() = 0;
-    virtual status_t processPacket(const sp<ABuffer> &packet) = 0;
-
-protected:
-    virtual ~Assembler() {}
-
-    void postAccessUnit(
-            const sp<ABuffer> &accessUnit, bool followsDiscontinuity);
-
-private:
-    sp<AMessage> mNotify;
-
-    DISALLOW_EVIL_CONSTRUCTORS(Assembler);
-};
-
-struct RTPReceiver::TSAssembler : public RTPReceiver::Assembler {
-    TSAssembler(const sp<AMessage> &notify);
-
-    virtual void signalDiscontinuity();
-    virtual status_t processPacket(const sp<ABuffer> &packet);
-
-private:
-    bool mSawDiscontinuity;
-
-    DISALLOW_EVIL_CONSTRUCTORS(TSAssembler);
-};
-
-struct RTPReceiver::H264Assembler : public RTPReceiver::Assembler {
-    H264Assembler(const sp<AMessage> &notify);
-
-    virtual void signalDiscontinuity();
-    virtual status_t processPacket(const sp<ABuffer> &packet);
-
-private:
-    int32_t mState;
-
-    uint8_t mIndicator;
-    uint8_t mNALType;
-
-    sp<ABuffer> mAccumulator;
-
-    List<sp<ABuffer> > mNALUnits;
-    int32_t mAccessUnitRTPTime;
-
-    status_t internalProcessPacket(const sp<ABuffer> &packet);
-
-    void addSingleNALUnit(const sp<ABuffer> &packet);
-    status_t addSingleTimeAggregationPacket(const sp<ABuffer> &packet);
-
-    void flushAccessUnit();
-
-    void clearAccumulator();
-    void appendToAccumulator(const void *data, size_t size);
-
-    void reset();
-
-    DISALLOW_EVIL_CONSTRUCTORS(H264Assembler);
-};
-
-}  // namespace android
-
-#endif  // RTP_ASSEMBLER_H_
-

media/libstagefright/wifi-display/rtp/RTPReceiver.cpp

diff --git a/media/libstagefright/wifi-display/rtp/RTPReceiver.cpp b/media/libstagefright/wifi-display/rtp/RTPReceiver.cpp
deleted file mode 100644
index 3b3bd63..0000000
--- a/media/libstagefright/wifi-display/rtp/RTPReceiver.cpp
+++ /dev/null
@@ -1,1152 +0,0 @@
-/*
- * Copyright 2013, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "RTPReceiver"
-#include <utils/Log.h>
-
-#include "RTPAssembler.h"
-#include "RTPReceiver.h"
-
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/foundation/ANetworkSession.h>
-#include <media/stagefright/foundation/hexdump.h>
-#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/Utils.h>
-
-#define TRACK_PACKET_LOSS       0
-
-namespace android {
-
-////////////////////////////////////////////////////////////////////////////////
-
-struct RTPReceiver::Source : public AHandler {
-    Source(RTPReceiver *receiver, uint32_t ssrc);
-
-    void onPacketReceived(uint16_t seq, const sp<ABuffer> &buffer);
-
-    void addReportBlock(uint32_t ssrc, const sp<ABuffer> &buf);
-
-protected:
-    virtual ~Source();
-
-    virtual void onMessageReceived(const sp<AMessage> &msg);
-
-private:
-    enum {
-        kWhatRetransmit,
-        kWhatDeclareLost,
-    };
-
-    static const uint32_t kMinSequential = 2;
-    static const uint32_t kMaxDropout = 3000;
-    static const uint32_t kMaxMisorder = 100;
-    static const uint32_t kRTPSeqMod = 1u << 16;
-    static const int64_t kReportIntervalUs = 10000000ll;
-
-    RTPReceiver *mReceiver;
-    uint32_t mSSRC;
-    bool mFirst;
-    uint16_t mMaxSeq;
-    uint32_t mCycles;
-    uint32_t mBaseSeq;
-    uint32_t mReceived;
-    uint32_t mExpectedPrior;
-    uint32_t mReceivedPrior;
-
-    int64_t mFirstArrivalTimeUs;
-    int64_t mFirstRTPTimeUs;
-
-    // Ordered by extended seq number.
-    List<sp<ABuffer> > mPackets;
-
-    enum StatusBits {
-        STATUS_DECLARED_LOST            = 1,
-        STATUS_REQUESTED_RETRANSMISSION = 2,
-        STATUS_ARRIVED_LATE             = 4,
-    };
-#if TRACK_PACKET_LOSS
-    KeyedVector<int32_t, uint32_t> mLostPackets;
-#endif
-
-    void modifyPacketStatus(int32_t extSeqNo, uint32_t mask);
-
-    int32_t mAwaitingExtSeqNo;
-    bool mRequestedRetransmission;
-
-    int32_t mActivePacketType;
-    sp<Assembler> mActiveAssembler;
-
-    int64_t mNextReportTimeUs;
-
-    int32_t mNumDeclaredLost;
-    int32_t mNumDeclaredLostPrior;
-
-    int32_t mRetransmitGeneration;
-    int32_t mDeclareLostGeneration;
-    bool mDeclareLostTimerPending;
-
-    void queuePacket(const sp<ABuffer> &packet);
-    void dequeueMore();
-
-    sp<ABuffer> getNextPacket();
-    void resync();
-
-    void postRetransmitTimer(int64_t delayUs);
-    void postDeclareLostTimer(int64_t delayUs);
-    void cancelTimers();
-
-    DISALLOW_EVIL_CONSTRUCTORS(Source);
-};
-
-////////////////////////////////////////////////////////////////////////////////
-
-RTPReceiver::Source::Source(RTPReceiver *receiver, uint32_t ssrc)
-    : mReceiver(receiver),
-      mSSRC(ssrc),
-      mFirst(true),
-      mMaxSeq(0),
-      mCycles(0),
-      mBaseSeq(0),
-      mReceived(0),
-      mExpectedPrior(0),
-      mReceivedPrior(0),
-      mFirstArrivalTimeUs(-1ll),
-      mFirstRTPTimeUs(-1ll),
-      mAwaitingExtSeqNo(-1),
-      mRequestedRetransmission(false),
-      mActivePacketType(-1),
-      mNextReportTimeUs(-1ll),
-      mNumDeclaredLost(0),
-      mNumDeclaredLostPrior(0),
-      mRetransmitGeneration(0),
-      mDeclareLostGeneration(0),
-      mDeclareLostTimerPending(false) {
-}
-
-RTPReceiver::Source::~Source() {
-}
-
-void RTPReceiver::Source::onMessageReceived(const sp<AMessage> &msg) {
-    switch (msg->what()) {
-        case kWhatRetransmit:
-        {
-            int32_t generation;
-            CHECK(msg->findInt32("generation", &generation));
-
-            if (generation != mRetransmitGeneration) {
-                break;
-            }
-
-            mRequestedRetransmission = true;
-            mReceiver->requestRetransmission(mSSRC, mAwaitingExtSeqNo);
-
-            modifyPacketStatus(
-                    mAwaitingExtSeqNo, STATUS_REQUESTED_RETRANSMISSION);
-            break;
-        }
-
-        case kWhatDeclareLost:
-        {
-            int32_t generation;
-            CHECK(msg->findInt32("generation", &generation));
-
-            if (generation != mDeclareLostGeneration) {
-                break;
-            }
-
-            cancelTimers();
-
-            ALOGV("Lost packet extSeqNo %d %s",
-                  mAwaitingExtSeqNo,
-                  mRequestedRetransmission ? "*" : "");
-
-            mRequestedRetransmission = false;
-            if (mActiveAssembler != NULL) {
-                mActiveAssembler->signalDiscontinuity();
-            }
-
-            modifyPacketStatus(mAwaitingExtSeqNo, STATUS_DECLARED_LOST);
-
-            // resync();
-            ++mAwaitingExtSeqNo;
-            ++mNumDeclaredLost;
-
-            mReceiver->notifyPacketLost();
-
-            dequeueMore();
-            break;
-        }
-
-        default:
-            TRESPASS();
-    }
-}
-
-void RTPReceiver::Source::onPacketReceived(
-        uint16_t seq, const sp<ABuffer> &buffer) {
-    if (mFirst) {
-        buffer->setInt32Data(mCycles | seq);
-        queuePacket(buffer);
-
-        mFirst = false;
-        mBaseSeq = seq;
-        mMaxSeq = seq;
-        ++mReceived;
-        return;
-    }
-
-    uint16_t udelta = seq - mMaxSeq;
-
-    if (udelta < kMaxDropout) {
-        // In order, with permissible gap.
-
-        if (seq < mMaxSeq) {
-            // Sequence number wrapped - count another 64K cycle
-            mCycles += kRTPSeqMod;
-        }
-
-        mMaxSeq = seq;
-
-        ++mReceived;
-    } else if (udelta <= kRTPSeqMod - kMaxMisorder) {
-        // The sequence number made a very large jump
-        return;
-    } else {
-        // Duplicate or reordered packet.
-    }
-
-    buffer->setInt32Data(mCycles | seq);
-    queuePacket(buffer);
-}
-
-void RTPReceiver::Source::queuePacket(const sp<ABuffer> &packet) {
-    int32_t newExtendedSeqNo = packet->int32Data();
-
-    if (mFirstArrivalTimeUs < 0ll) {
-        mFirstArrivalTimeUs = ALooper::GetNowUs();
-
-        uint32_t rtpTime;
-        CHECK(packet->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
-
-        mFirstRTPTimeUs = (rtpTime * 100ll) / 9ll;
-    }
-
-    if (mAwaitingExtSeqNo >= 0 && newExtendedSeqNo < mAwaitingExtSeqNo) {
-        // We're no longer interested in these. They're old.
-        ALOGV("dropping stale extSeqNo %d", newExtendedSeqNo);
-
-        modifyPacketStatus(newExtendedSeqNo, STATUS_ARRIVED_LATE);
-        return;
-    }
-
-    if (mPackets.empty()) {
-        mPackets.push_back(packet);
-        dequeueMore();
-        return;
-    }
-
-    List<sp<ABuffer> >::iterator firstIt = mPackets.begin();
-    List<sp<ABuffer> >::iterator it = --mPackets.end();
-    for (;;) {
-        int32_t extendedSeqNo = (*it)->int32Data();
-
-        if (extendedSeqNo == newExtendedSeqNo) {
-            // Duplicate packet.
-            return;
-        }
-
-        if (extendedSeqNo < newExtendedSeqNo) {
-            // Insert new packet after the one at "it".
-            mPackets.insert(++it, packet);
-            break;
-        }
-
-        if (it == firstIt) {
-            // Insert new packet before the first existing one.
-            mPackets.insert(it, packet);
-            break;
-        }
-
-        --it;
-    }
-
-    dequeueMore();
-}
-
-void RTPReceiver::Source::dequeueMore() {
-    int64_t nowUs = ALooper::GetNowUs();
-    if (mNextReportTimeUs < 0ll || nowUs >= mNextReportTimeUs) {
-        if (mNextReportTimeUs >= 0ll) {
-            uint32_t expected = (mMaxSeq | mCycles) - mBaseSeq + 1;
-
-            uint32_t expectedInterval = expected - mExpectedPrior;
-            mExpectedPrior = expected;
-
-            uint32_t receivedInterval = mReceived - mReceivedPrior;
-            mReceivedPrior = mReceived;
-
-            int64_t lostInterval =
-                (int64_t)expectedInterval - (int64_t)receivedInterval;
-
-            int32_t declaredLostInterval =
-                mNumDeclaredLost - mNumDeclaredLostPrior;
-
-            mNumDeclaredLostPrior = mNumDeclaredLost;
-
-            if (declaredLostInterval > 0) {
-                ALOGI("lost %lld packets (%.2f %%), declared %d lost\n",
-                      lostInterval,
-                      100.0f * lostInterval / expectedInterval,
-                      declaredLostInterval);
-            }
-        }
-
-        mNextReportTimeUs = nowUs + kReportIntervalUs;
-
-#if TRACK_PACKET_LOSS
-        for (size_t i = 0; i < mLostPackets.size(); ++i) {
-            int32_t key = mLostPackets.keyAt(i);
-            uint32_t value = mLostPackets.valueAt(i);
-
-            AString status;
-            if (value & STATUS_REQUESTED_RETRANSMISSION) {
-                status.append("retrans ");
-            }
-            if (value & STATUS_ARRIVED_LATE) {
-                status.append("arrived-late ");
-            }
-            ALOGI("Packet %d declared lost %s", key, status.c_str());
-        }
-#endif
-    }
-
-    sp<ABuffer> packet;
-    while ((packet = getNextPacket()) != NULL) {
-        if (mDeclareLostTimerPending) {
-            cancelTimers();
-        }
-
-        CHECK_GE(mAwaitingExtSeqNo, 0);
-#if TRACK_PACKET_LOSS
-        mLostPackets.removeItem(mAwaitingExtSeqNo);
-#endif
-
-        int32_t packetType;
-        CHECK(packet->meta()->findInt32("PT", &packetType));
-
-        if (packetType != mActivePacketType) {
-            mActiveAssembler = mReceiver->makeAssembler(packetType);
-            mActivePacketType = packetType;
-        }
-
-        if (mActiveAssembler != NULL) {
-            status_t err = mActiveAssembler->processPacket(packet);
-            if (err != OK) {
-                ALOGV("assembler returned error %d", err);
-            }
-        }
-
-        ++mAwaitingExtSeqNo;
-    }
-
-    if (mDeclareLostTimerPending) {
-        return;
-    }
-
-    if (mPackets.empty()) {
-        return;
-    }
-
-    CHECK_GE(mAwaitingExtSeqNo, 0);
-
-    const sp<ABuffer> &firstPacket = *mPackets.begin();
-
-    uint32_t rtpTime;
-    CHECK(firstPacket->meta()->findInt32(
-                "rtp-time", (int32_t *)&rtpTime));
-
-
-    int64_t rtpUs = (rtpTime * 100ll) / 9ll;
-
-    int64_t maxArrivalTimeUs =
-        mFirstArrivalTimeUs + rtpUs - mFirstRTPTimeUs;
-
-    nowUs = ALooper::GetNowUs();
-
-    CHECK_LT(mAwaitingExtSeqNo, firstPacket->int32Data());
-
-    ALOGV("waiting for %d, comparing against %d, %lld us left",
-          mAwaitingExtSeqNo,
-          firstPacket->int32Data(),
-          maxArrivalTimeUs - nowUs);
-
-    postDeclareLostTimer(maxArrivalTimeUs + kPacketLostAfterUs);
-
-    if (kRequestRetransmissionAfterUs > 0ll) {
-        postRetransmitTimer(
-                maxArrivalTimeUs + kRequestRetransmissionAfterUs);
-    }
-}
-
-sp<ABuffer> RTPReceiver::Source::getNextPacket() {
-    if (mPackets.empty()) {
-        return NULL;
-    }
-
-    int32_t extSeqNo = (*mPackets.begin())->int32Data();
-
-    if (mAwaitingExtSeqNo < 0) {
-        mAwaitingExtSeqNo = extSeqNo;
-    } else if (extSeqNo != mAwaitingExtSeqNo) {
-        return NULL;
-    }
-
-    sp<ABuffer> packet = *mPackets.begin();
-    mPackets.erase(mPackets.begin());
-
-    return packet;
-}
-
-void RTPReceiver::Source::resync() {
-    mAwaitingExtSeqNo = -1;
-}
-
-void RTPReceiver::Source::addReportBlock(
-        uint32_t ssrc, const sp<ABuffer> &buf) {
-    uint32_t extMaxSeq = mMaxSeq | mCycles;
-    uint32_t expected = extMaxSeq - mBaseSeq + 1;
-
-    int64_t lost = (int64_t)expected - (int64_t)mReceived;
-    if (lost > 0x7fffff) {
-        lost = 0x7fffff;
-    } else if (lost < -0x800000) {
-        lost = -0x800000;
-    }
-
-    uint32_t expectedInterval = expected - mExpectedPrior;
-    mExpectedPrior = expected;
-
-    uint32_t receivedInterval = mReceived - mReceivedPrior;
-    mReceivedPrior = mReceived;
-
-    int64_t lostInterval = expectedInterval - receivedInterval;
-
-    uint8_t fractionLost;
-    if (expectedInterval == 0 || lostInterval <=0) {
-        fractionLost = 0;
-    } else {
-        fractionLost = (lostInterval << 8) / expectedInterval;
-    }
-
-    uint8_t *ptr = buf->data() + buf->size();
-
-    ptr[0] = ssrc >> 24;
-    ptr[1] = (ssrc >> 16) & 0xff;
-    ptr[2] = (ssrc >> 8) & 0xff;
-    ptr[3] = ssrc & 0xff;
-
-    ptr[4] = fractionLost;
-
-    ptr[5] = (lost >> 16) & 0xff;
-    ptr[6] = (lost >> 8) & 0xff;
-    ptr[7] = lost & 0xff;
-
-    ptr[8] = extMaxSeq >> 24;
-    ptr[9] = (extMaxSeq >> 16) & 0xff;
-    ptr[10] = (extMaxSeq >> 8) & 0xff;
-    ptr[11] = extMaxSeq & 0xff;
-
-    // XXX TODO:
-
-    ptr[12] = 0x00;  // interarrival jitter
-    ptr[13] = 0x00;
-    ptr[14] = 0x00;
-    ptr[15] = 0x00;
-
-    ptr[16] = 0x00;  // last SR
-    ptr[17] = 0x00;
-    ptr[18] = 0x00;
-    ptr[19] = 0x00;
-
-    ptr[20] = 0x00;  // delay since last SR
-    ptr[21] = 0x00;
-    ptr[22] = 0x00;
-    ptr[23] = 0x00;
-
-    buf->setRange(buf->offset(), buf->size() + 24);
-}
-
-////////////////////////////////////////////////////////////////////////////////
-
-RTPReceiver::RTPReceiver(
-        const sp<ANetworkSession> &netSession,
-        const sp<AMessage> &notify,
-        uint32_t flags)
-    : mNetSession(netSession),
-      mNotify(notify),
-      mFlags(flags),
-      mRTPMode(TRANSPORT_UNDEFINED),
-      mRTCPMode(TRANSPORT_UNDEFINED),
-      mRTPSessionID(0),
-      mRTCPSessionID(0),
-      mRTPConnected(false),
-      mRTCPConnected(false),
-      mRTPClientSessionID(0),
-      mRTCPClientSessionID(0) {
-}
-
-RTPReceiver::~RTPReceiver() {
-    if (mRTCPClientSessionID != 0) {
-        mNetSession->destroySession(mRTCPClientSessionID);
-        mRTCPClientSessionID = 0;
-    }
-
-    if (mRTPClientSessionID != 0) {
-        mNetSession->destroySession(mRTPClientSessionID);
-        mRTPClientSessionID = 0;
-    }
-
-    if (mRTCPSessionID != 0) {
-        mNetSession->destroySession(mRTCPSessionID);
-        mRTCPSessionID = 0;
-    }
-
-    if (mRTPSessionID != 0) {
-        mNetSession->destroySession(mRTPSessionID);
-        mRTPSessionID = 0;
-    }
-}
-
-status_t RTPReceiver::initAsync(
-        TransportMode rtpMode,
-        TransportMode rtcpMode,
-        int32_t *outLocalRTPPort) {
-    if (mRTPMode != TRANSPORT_UNDEFINED
-            || rtpMode == TRANSPORT_UNDEFINED
-            || rtpMode == TRANSPORT_NONE
-            || rtcpMode == TRANSPORT_UNDEFINED) {
-        return INVALID_OPERATION;
-    }
-
-    CHECK_NE(rtpMode, TRANSPORT_TCP_INTERLEAVED);
-    CHECK_NE(rtcpMode, TRANSPORT_TCP_INTERLEAVED);
-
-    sp<AMessage> rtpNotify = new AMessage(kWhatRTPNotify, id());
-
-    sp<AMessage> rtcpNotify;
-    if (rtcpMode != TRANSPORT_NONE) {
-        rtcpNotify = new AMessage(kWhatRTCPNotify, id());
-    }
-
-    CHECK_EQ(mRTPSessionID, 0);
-    CHECK_EQ(mRTCPSessionID, 0);
-
-    int32_t localRTPPort;
-
-    struct in_addr ifaceAddr;
-    ifaceAddr.s_addr = INADDR_ANY;
-
-    for (;;) {
-        localRTPPort = PickRandomRTPPort();
-
-        status_t err;
-        if (rtpMode == TRANSPORT_UDP) {
-            err = mNetSession->createUDPSession(
-                    localRTPPort,
-                    rtpNotify,
-                    &mRTPSessionID);
-        } else {
-            CHECK_EQ(rtpMode, TRANSPORT_TCP);
-            err = mNetSession->createTCPDatagramSession(
-                    ifaceAddr,
-                    localRTPPort,
-                    rtpNotify,
-                    &mRTPSessionID);
-        }
-
-        if (err != OK) {
-            continue;
-        }
-
-        if (rtcpMode == TRANSPORT_NONE) {
-            break;
-        } else if (rtcpMode == TRANSPORT_UDP) {
-            err = mNetSession->createUDPSession(
-                    localRTPPort + 1,
-                    rtcpNotify,
-                    &mRTCPSessionID);
-        } else {
-            CHECK_EQ(rtpMode, TRANSPORT_TCP);
-            err = mNetSession->createTCPDatagramSession(
-                    ifaceAddr,
-                    localRTPPort + 1,
-                    rtcpNotify,
-                    &mRTCPSessionID);
-        }
-
-        if (err == OK) {
-            break;
-        }
-
-        mNetSession->destroySession(mRTPSessionID);
-        mRTPSessionID = 0;
-    }
-
-    mRTPMode = rtpMode;
-    mRTCPMode = rtcpMode;
-    *outLocalRTPPort = localRTPPort;
-
-    return OK;
-}
-
-status_t RTPReceiver::connect(
-        const char *remoteHost, int32_t remoteRTPPort, int32_t remoteRTCPPort) {
-    status_t err;
-
-    if (mRTPMode == TRANSPORT_UDP) {
-        CHECK(!mRTPConnected);
-
-        err = mNetSession->connectUDPSession(
-                mRTPSessionID, remoteHost, remoteRTPPort);
-
-        if (err != OK) {
-            notifyInitDone(err);
-            return err;
-        }
-
-        ALOGI("connectUDPSession RTP successful.");
-
-        mRTPConnected = true;
-    }
-
-    if (mRTCPMode == TRANSPORT_UDP) {
-        CHECK(!mRTCPConnected);
-
-        err = mNetSession->connectUDPSession(
-                mRTCPSessionID, remoteHost, remoteRTCPPort);
-
-        if (err != OK) {
-            notifyInitDone(err);
-            return err;
-        }
-
-        scheduleSendRR();
-
-        ALOGI("connectUDPSession RTCP successful.");
-
-        mRTCPConnected = true;
-    }
-
-    if (mRTPConnected
-            && (mRTCPConnected || mRTCPMode == TRANSPORT_NONE)) {
-        notifyInitDone(OK);
-    }
-
-    return OK;
-}
-
-status_t RTPReceiver::informSender(const sp<AMessage> &params) {
-    if (!mRTCPConnected) {
-        return INVALID_OPERATION;
-    }
-
-    int64_t avgLatencyUs;
-    CHECK(params->findInt64("avgLatencyUs", &avgLatencyUs));
-
-    int64_t maxLatencyUs;
-    CHECK(params->findInt64("maxLatencyUs", &maxLatencyUs));
-
-    sp<ABuffer> buf = new ABuffer(28);
-
-    uint8_t *ptr = buf->data();
-    ptr[0] = 0x80 | 0;
-    ptr[1] = 204;  // APP
-    ptr[2] = 0;
-
-    CHECK((buf->size() % 4) == 0u);
-    ptr[3] = (buf->size() / 4) - 1;
-
-    ptr[4] = kSourceID >> 24;  // SSRC
-    ptr[5] = (kSourceID >> 16) & 0xff;
-    ptr[6] = (kSourceID >> 8) & 0xff;
-    ptr[7] = kSourceID & 0xff;
-    ptr[8] = 'l';
-    ptr[9] = 'a';
-    ptr[10] = 't';
-    ptr[11] = 'e';
-
-    ptr[12] = avgLatencyUs >> 56;
-    ptr[13] = (avgLatencyUs >> 48) & 0xff;
-    ptr[14] = (avgLatencyUs >> 40) & 0xff;
-    ptr[15] = (avgLatencyUs >> 32) & 0xff;
-    ptr[16] = (avgLatencyUs >> 24) & 0xff;
-    ptr[17] = (avgLatencyUs >> 16) & 0xff;
-    ptr[18] = (avgLatencyUs >> 8) & 0xff;
-    ptr[19] = avgLatencyUs & 0xff;
-
-    ptr[20] = maxLatencyUs >> 56;
-    ptr[21] = (maxLatencyUs >> 48) & 0xff;
-    ptr[22] = (maxLatencyUs >> 40) & 0xff;
-    ptr[23] = (maxLatencyUs >> 32) & 0xff;
-    ptr[24] = (maxLatencyUs >> 24) & 0xff;
-    ptr[25] = (maxLatencyUs >> 16) & 0xff;
-    ptr[26] = (maxLatencyUs >> 8) & 0xff;
-    ptr[27] = maxLatencyUs & 0xff;
-
-    mNetSession->sendRequest(mRTCPSessionID, buf->data(), buf->size());
-
-    return OK;
-}
-
-void RTPReceiver::onMessageReceived(const sp<AMessage> &msg) {
-    switch (msg->what()) {
-        case kWhatRTPNotify:
-        case kWhatRTCPNotify:
-            onNetNotify(msg->what() == kWhatRTPNotify, msg);
-            break;
-
-        case kWhatSendRR:
-        {
-            onSendRR();
-            break;
-        }
-
-        default:
-            TRESPASS();
-    }
-}
-
-void RTPReceiver::onNetNotify(bool isRTP, const sp<AMessage> &msg) {
-    int32_t reason;
-    CHECK(msg->findInt32("reason", &reason));
-
-    switch (reason) {
-        case ANetworkSession::kWhatError:
-        {
-            int32_t sessionID;
-            CHECK(msg->findInt32("sessionID", &sessionID));
-
-            int32_t err;
-            CHECK(msg->findInt32("err", &err));
-
-            int32_t errorOccuredDuringSend;
-            CHECK(msg->findInt32("send", &errorOccuredDuringSend));
-
-            AString detail;
-            CHECK(msg->findString("detail", &detail));
-
-            ALOGE("An error occurred during %s in session %d "
-                  "(%d, '%s' (%s)).",
-                  errorOccuredDuringSend ? "send" : "receive",
-                  sessionID,
-                  err,
-                  detail.c_str(),
-                  strerror(-err));
-
-            mNetSession->destroySession(sessionID);
-
-            if (sessionID == mRTPSessionID) {
-                mRTPSessionID = 0;
-            } else if (sessionID == mRTCPSessionID) {
-                mRTCPSessionID = 0;
-            } else if (sessionID == mRTPClientSessionID) {
-                mRTPClientSessionID = 0;
-            } else if (sessionID == mRTCPClientSessionID) {
-                mRTCPClientSessionID = 0;
-            }
-
-            if (!mRTPConnected
-                    || (mRTCPMode != TRANSPORT_NONE && !mRTCPConnected)) {
-                notifyInitDone(err);
-                break;
-            }
-
-            notifyError(err);
-            break;
-        }
-
-        case ANetworkSession::kWhatDatagram:
-        {
-            sp<ABuffer> data;
-            CHECK(msg->findBuffer("data", &data));
-
-            if (isRTP) {
-                if (mFlags & FLAG_AUTO_CONNECT) {
-                    AString fromAddr;
-                    CHECK(msg->findString("fromAddr", &fromAddr));
-
-                    int32_t fromPort;
-                    CHECK(msg->findInt32("fromPort", &fromPort));
-
-                    CHECK_EQ((status_t)OK,
-                             connect(
-                                 fromAddr.c_str(), fromPort, fromPort + 1));
-
-                    mFlags &= ~FLAG_AUTO_CONNECT;
-                }
-
-                onRTPData(data);
-            } else {
-                onRTCPData(data);
-            }
-            break;
-        }
-
-        case ANetworkSession::kWhatClientConnected:
-        {
-            int32_t sessionID;
-            CHECK(msg->findInt32("sessionID", &sessionID));
-
-            if (isRTP) {
-                CHECK_EQ(mRTPMode, TRANSPORT_TCP);
-
-                if (mRTPClientSessionID != 0) {
-                    // We only allow a single client connection.
-                    mNetSession->destroySession(sessionID);
-                    sessionID = 0;
-                    break;
-                }
-
-                mRTPClientSessionID = sessionID;
-                mRTPConnected = true;
-            } else {
-                CHECK_EQ(mRTCPMode, TRANSPORT_TCP);
-
-                if (mRTCPClientSessionID != 0) {
-                    // We only allow a single client connection.
-                    mNetSession->destroySession(sessionID);
-                    sessionID = 0;
-                    break;
-                }
-
-                mRTCPClientSessionID = sessionID;
-                mRTCPConnected = true;
-            }
-
-            if (mRTPConnected
-                    && (mRTCPConnected || mRTCPMode == TRANSPORT_NONE)) {
-                notifyInitDone(OK);
-            }
-            break;
-        }
-    }
-}
-
-void RTPReceiver::notifyInitDone(status_t err) {
-    sp<AMessage> notify = mNotify->dup();
-    notify->setInt32("what", kWhatInitDone);
-    notify->setInt32("err", err);
-    notify->post();
-}
-
-void RTPReceiver::notifyError(status_t err) {
-    sp<AMessage> notify = mNotify->dup();
-    notify->setInt32("what", kWhatError);
-    notify->setInt32("err", err);
-    notify->post();
-}
-
-void RTPReceiver::notifyPacketLost() {
-    sp<AMessage> notify = mNotify->dup();
-    notify->setInt32("what", kWhatPacketLost);
-    notify->post();
-}
-
-status_t RTPReceiver::onRTPData(const sp<ABuffer> &buffer) {
-    size_t size = buffer->size();
-    if (size < 12) {
-        // Too short to be a valid RTP header.
-        return ERROR_MALFORMED;
-    }
-
-    const uint8_t *data = buffer->data();
-
-    if ((data[0] >> 6) != 2) {
-        // Unsupported version.
-        return ERROR_UNSUPPORTED;
-    }
-
-    if (data[0] & 0x20) {
-        // Padding present.
-
-        size_t paddingLength = data[size - 1];
-
-        if (paddingLength + 12 > size) {
-            // If we removed this much padding we'd end up with something
-            // that's too short to be a valid RTP header.
-            return ERROR_MALFORMED;
-        }
-
-        size -= paddingLength;
-    }
-
-    int numCSRCs = data[0] & 0x0f;
-
-    size_t payloadOffset = 12 + 4 * numCSRCs;
-
-    if (size < payloadOffset) {
-        // Not enough data to fit the basic header and all the CSRC entries.
-        return ERROR_MALFORMED;
-    }
-
-    if (data[0] & 0x10) {
-        // Header eXtension present.
-
-        if (size < payloadOffset + 4) {
-            // Not enough data to fit the basic header, all CSRC entries
-            // and the first 4 bytes of the extension header.
-
-            return ERROR_MALFORMED;
-        }
-
-        const uint8_t *extensionData = &data[payloadOffset];
-
-        size_t extensionLength =
-            4 * (extensionData[2] << 8 | extensionData[3]);
-
-        if (size < payloadOffset + 4 + extensionLength) {
-            return ERROR_MALFORMED;
-        }
-
-        payloadOffset += 4 + extensionLength;
-    }
-
-    uint32_t srcId = U32_AT(&data[8]);
-    uint32_t rtpTime = U32_AT(&data[4]);
-    uint16_t seqNo = U16_AT(&data[2]);
-
-    sp<AMessage> meta = buffer->meta();
-    meta->setInt32("ssrc", srcId);
-    meta->setInt32("rtp-time", rtpTime);
-    meta->setInt32("PT", data[1] & 0x7f);
-    meta->setInt32("M", data[1] >> 7);
-
-    buffer->setRange(payloadOffset, size - payloadOffset);
-
-    ssize_t index = mSources.indexOfKey(srcId);
-    sp<Source> source;
-    if (index < 0) {
-        source = new Source(this, srcId);
-        looper()->registerHandler(source);
-
-        mSources.add(srcId, source);
-    } else {
-        source = mSources.valueAt(index);
-    }
-
-    source->onPacketReceived(seqNo, buffer);
-
-    return OK;
-}
-
-status_t RTPReceiver::onRTCPData(const sp<ABuffer> &data) {
-    ALOGI("onRTCPData");
-    return OK;
-}
-
-void RTPReceiver::addSDES(const sp<ABuffer> &buffer) {
-    uint8_t *data = buffer->data() + buffer->size();
-    data[0] = 0x80 | 1;
-    data[1] = 202;  // SDES
-    data[4] = kSourceID >> 24;  // SSRC
-    data[5] = (kSourceID >> 16) & 0xff;
-    data[6] = (kSourceID >> 8) & 0xff;
-    data[7] = kSourceID & 0xff;
-
-    size_t offset = 8;
-
-    data[offset++] = 1;  // CNAME
-
-    AString cname = "stagefright@somewhere";
-    data[offset++] = cname.size();
-
-    memcpy(&data[offset], cname.c_str(), cname.size());
-    offset += cname.size();
-
-    data[offset++] = 6;  // TOOL
-
-    AString tool = "stagefright/1.0";
-    data[offset++] = tool.size();
-
-    memcpy(&data[offset], tool.c_str(), tool.size());
-    offset += tool.size();
-
-    data[offset++] = 0;
-
-    if ((offset % 4) > 0) {
-        size_t count = 4 - (offset % 4);
-        switch (count) {
-            case 3:
-                data[offset++] = 0;
-            case 2:
-                data[offset++] = 0;
-            case 1:
-                data[offset++] = 0;
-        }
-    }
-
-    size_t numWords = (offset / 4) - 1;
-    data[2] = numWords >> 8;
-    data[3] = numWords & 0xff;
-
-    buffer->setRange(buffer->offset(), buffer->size() + offset);
-}
-
-void RTPReceiver::scheduleSendRR() {
-    (new AMessage(kWhatSendRR, id()))->post(5000000ll);
-}
-
-void RTPReceiver::onSendRR() {
-    sp<ABuffer> buf = new ABuffer(kMaxUDPPacketSize);
-    buf->setRange(0, 0);
-
-    uint8_t *ptr = buf->data();
-    ptr[0] = 0x80 | 0;
-    ptr[1] = 201;  // RR
-    ptr[2] = 0;
-    ptr[3] = 1;
-    ptr[4] = kSourceID >> 24;  // SSRC
-    ptr[5] = (kSourceID >> 16) & 0xff;
-    ptr[6] = (kSourceID >> 8) & 0xff;
-    ptr[7] = kSourceID & 0xff;
-
-    buf->setRange(0, 8);
-
-    size_t numReportBlocks = 0;
-    for (size_t i = 0; i < mSources.size(); ++i) {
-        uint32_t ssrc = mSources.keyAt(i);
-        sp<Source> source = mSources.valueAt(i);
-
-        if (numReportBlocks > 31 || buf->size() + 24 > buf->capacity()) {
-            // Cannot fit another report block.
-            break;
-        }
-
-        source->addReportBlock(ssrc, buf);
-        ++numReportBlocks;
-    }
-
-    ptr[0] |= numReportBlocks;  // 5 bit
-
-    size_t sizeInWordsMinus1 = 1 + 6 * numReportBlocks;
-    ptr[2] = sizeInWordsMinus1 >> 8;
-    ptr[3] = sizeInWordsMinus1 & 0xff;
-
-    buf->setRange(0, (sizeInWordsMinus1 + 1) * 4);
-
-    addSDES(buf);
-
-    mNetSession->sendRequest(mRTCPSessionID, buf->data(), buf->size());
-
-    scheduleSendRR();
-}
-
-status_t RTPReceiver::registerPacketType(
-        uint8_t packetType, PacketizationMode mode) {
-    mPacketTypes.add(packetType, mode);
-
-    return OK;
-}
-
-sp<RTPReceiver::Assembler> RTPReceiver::makeAssembler(uint8_t packetType) {
-    ssize_t index = mPacketTypes.indexOfKey(packetType);
-    if (index < 0) {
-        return NULL;
-    }
-
-    PacketizationMode mode = mPacketTypes.valueAt(index);
-
-    switch (mode) {
-        case PACKETIZATION_NONE:
-        case PACKETIZATION_TRANSPORT_STREAM:
-            return new TSAssembler(mNotify);
-
-        case PACKETIZATION_H264:
-            return new H264Assembler(mNotify);
-
-        default:
-            return NULL;
-    }
-}
-
-void RTPReceiver::requestRetransmission(uint32_t senderSSRC, int32_t extSeqNo) {
-    int32_t blp = 0;
-
-    sp<ABuffer> buf = new ABuffer(16);
-    buf->setRange(0, 0);
-
-    uint8_t *ptr = buf->data();
-    ptr[0] = 0x80 | 1;  // generic NACK
-    ptr[1] = 205;  // TSFB
-    ptr[2] = 0;
-    ptr[3] = 3;
-    ptr[8] = (senderSSRC >> 24) & 0xff;
-    ptr[9] = (senderSSRC >> 16) & 0xff;
-    ptr[10] = (senderSSRC >> 8) & 0xff;
-    ptr[11] = (senderSSRC & 0xff);
-    ptr[8] = (kSourceID >> 24) & 0xff;
-    ptr[9] = (kSourceID >> 16) & 0xff;
-    ptr[10] = (kSourceID >> 8) & 0xff;
-    ptr[11] = (kSourceID & 0xff);
-    ptr[12] = (extSeqNo >> 8) & 0xff;
-    ptr[13] = (extSeqNo & 0xff);
-    ptr[14] = (blp >> 8) & 0xff;
-    ptr[15] = (blp & 0xff);
-
-    buf->setRange(0, 16);
-
-     mNetSession->sendRequest(mRTCPSessionID, buf->data(), buf->size());
-}
-
-void RTPReceiver::Source::modifyPacketStatus(int32_t extSeqNo, uint32_t mask) {
-#if TRACK_PACKET_LOSS
-    ssize_t index = mLostPackets.indexOfKey(extSeqNo);
-    if (index < 0) {
-        mLostPackets.add(extSeqNo, mask);
-    } else {
-        mLostPackets.editValueAt(index) |= mask;
-    }
-#endif
-}
-
-void RTPReceiver::Source::postRetransmitTimer(int64_t timeUs) {
-    int64_t delayUs = timeUs - ALooper::GetNowUs();
-    sp<AMessage> msg = new AMessage(kWhatRetransmit, id());
-    msg->setInt32("generation", mRetransmitGeneration);
-    msg->post(delayUs);
-}
-
-void RTPReceiver::Source::postDeclareLostTimer(int64_t timeUs) {
-    CHECK(!mDeclareLostTimerPending);
-    mDeclareLostTimerPending = true;
-
-    int64_t delayUs = timeUs - ALooper::GetNowUs();
-    sp<AMessage> msg = new AMessage(kWhatDeclareLost, id());
-    msg->setInt32("generation", mDeclareLostGeneration);
-    msg->post(delayUs);
-}
-
-void RTPReceiver::Source::cancelTimers() {
-    ++mRetransmitGeneration;
-    ++mDeclareLostGeneration;
-    mDeclareLostTimerPending = false;
-}
-
-}  // namespace android
-

media/libstagefright/wifi-display/rtp/RTPReceiver.h

diff --git a/media/libstagefright/wifi-display/rtp/RTPReceiver.h b/media/libstagefright/wifi-display/rtp/RTPReceiver.h
deleted file mode 100644
index 240ab2e..0000000
--- a/media/libstagefright/wifi-display/rtp/RTPReceiver.h
+++ /dev/null
@@ -1,125 +0,0 @@
-/*
- * Copyright 2013, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef RTP_RECEIVER_H_
-
-#define RTP_RECEIVER_H_
-
-#include "RTPBase.h"
-
-#include <media/stagefright/foundation/AHandler.h>
-
-namespace android {
-
-struct ABuffer;
-struct ANetworkSession;
-
-// An object of this class facilitates receiving of media data on an RTP
-// channel. The channel is established over a UDP or TCP connection depending
-// on which "TransportMode" was chosen. In addition different RTP packetization
-// schemes are supported such as "Transport Stream Packets over RTP",
-// or "AVC/H.264 encapsulation as specified in RFC 3984 (non-interleaved mode)"
-struct RTPReceiver : public RTPBase, public AHandler {
-    enum {
-        kWhatInitDone,
-        kWhatError,
-        kWhatAccessUnit,
-        kWhatPacketLost,
-    };
-
-    enum Flags {
-        FLAG_AUTO_CONNECT = 1,
-    };
-    RTPReceiver(
-            const sp<ANetworkSession> &netSession,
-            const sp<AMessage> &notify,
-            uint32_t flags = 0);
-
-    status_t registerPacketType(
-            uint8_t packetType, PacketizationMode mode);
-
-    status_t initAsync(
-            TransportMode rtpMode,
-            TransportMode rtcpMode,
-            int32_t *outLocalRTPPort);
-
-    status_t connect(
-            const char *remoteHost,
-            int32_t remoteRTPPort,
-            int32_t remoteRTCPPort);
-
-    status_t informSender(const sp<AMessage> &params);
-
-protected:
-    virtual ~RTPReceiver();
-    virtual void onMessageReceived(const sp<AMessage> &msg);
-
-private:
-    enum {
-        kWhatRTPNotify,
-        kWhatRTCPNotify,
-        kWhatSendRR,
-    };
-
-    enum {
-        kSourceID                       = 0xdeadbeef,
-        kPacketLostAfterUs              = 100000,
-        kRequestRetransmissionAfterUs   = -1,
-    };
-
-    struct Assembler;
-    struct H264Assembler;
-    struct Source;
-    struct TSAssembler;
-
-    sp<ANetworkSession> mNetSession;
-    sp<AMessage> mNotify;
-    uint32_t mFlags;
-    TransportMode mRTPMode;
-    TransportMode mRTCPMode;
-    int32_t mRTPSessionID;
-    int32_t mRTCPSessionID;
-    bool mRTPConnected;
-    bool mRTCPConnected;
-
-    int32_t mRTPClientSessionID;  // in TRANSPORT_TCP mode.
-    int32_t mRTCPClientSessionID;  // in TRANSPORT_TCP mode.
-
-    KeyedVector<uint8_t, PacketizationMode> mPacketTypes;
-    KeyedVector<uint32_t, sp<Source> > mSources;
-
-    void onNetNotify(bool isRTP, const sp<AMessage> &msg);
-    status_t onRTPData(const sp<ABuffer> &data);
-    status_t onRTCPData(const sp<ABuffer> &data);
-    void onSendRR();
-
-    void scheduleSendRR();
-    void addSDES(const sp<ABuffer> &buffer);
-
-    void notifyInitDone(status_t err);
-    void notifyError(status_t err);
-    void notifyPacketLost();
-
-    sp<Assembler> makeAssembler(uint8_t packetType);
-
-    void requestRetransmission(uint32_t senderSSRC, int32_t extSeqNo);
-
-    DISALLOW_EVIL_CONSTRUCTORS(RTPReceiver);
-};
-
-}  // namespace android
-
-#endif  // RTP_RECEIVER_H_

media/libstagefright/wifi-display/rtp/RTPSender.cpp

diff --git a/media/libstagefright/wifi-display/rtp/RTPSender.cpp b/media/libstagefright/wifi-display/rtp/RTPSender.cpp
index 1887b8b..e88a3bd 100644
--- a/media/libstagefright/wifi-display/rtp/RTPSender.cpp
+++ b/media/libstagefright/wifi-display/rtp/RTPSender.cpp
@@ -685,9 +685,8 @@
     return OK;
 }
 
-status_t RTPSender::parseReceiverReport(const uint8_t *data, size_t size) {
-    // hexdump(data, size);
-
+status_t RTPSender::parseReceiverReport(
+        const uint8_t *data, size_t /* size */) {
     float fractionLost = data[12] / 256.0f;
 
     ALOGI("lost %.2f %% of packets during report interval.",

media/libstagefright/wifi-display/rtptest.cpp

diff --git a/media/libstagefright/wifi-display/rtptest.cpp b/media/libstagefright/wifi-display/rtptest.cpp
deleted file mode 100644
index b902f29..0000000
--- a/media/libstagefright/wifi-display/rtptest.cpp
+++ /dev/null
@@ -1,565 +0,0 @@
-/*
- * Copyright 2013, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NEBUG 0
-#define LOG_TAG "rtptest"
-#include <utils/Log.h>
-
-#include "rtp/RTPSender.h"
-#include "rtp/RTPReceiver.h"
-#include "TimeSyncer.h"
-
-#include <binder/ProcessState.h>
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AHandler.h>
-#include <media/stagefright/foundation/ALooper.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/foundation/ANetworkSession.h>
-#include <media/stagefright/foundation/hexdump.h>
-#include <media/stagefright/DataSource.h>
-#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/NuMediaExtractor.h>
-#include <media/stagefright/Utils.h>
-
-#define MEDIA_FILENAME "/sdcard/Frame Counter HD 30FPS_1080p.mp4"
-
-namespace android {
-
-struct PacketSource : public RefBase {
-    PacketSource() {}
-
-    virtual sp<ABuffer> getNextAccessUnit() = 0;
-
-protected:
-    virtual ~PacketSource() {}
-
-private:
-    DISALLOW_EVIL_CONSTRUCTORS(PacketSource);
-};
-
-struct MediaPacketSource : public PacketSource {
-    MediaPacketSource()
-        : mMaxSampleSize(1024 * 1024) {
-        mExtractor = new NuMediaExtractor;
-        CHECK_EQ((status_t)OK,
-                 mExtractor->setDataSource(MEDIA_FILENAME));
-
-        bool haveVideo = false;
-        for (size_t i = 0; i < mExtractor->countTracks(); ++i) {
-            sp<AMessage> format;
-            CHECK_EQ((status_t)OK, mExtractor->getTrackFormat(i, &format));
-
-            AString mime;
-            CHECK(format->findString("mime", &mime));
-
-            if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_AVC, mime.c_str())) {
-                mExtractor->selectTrack(i);
-                haveVideo = true;
-                break;
-            }
-        }
-
-        CHECK(haveVideo);
-    }
-
-    virtual sp<ABuffer> getNextAccessUnit() {
-        int64_t timeUs;
-        status_t err = mExtractor->getSampleTime(&timeUs);
-
-        if (err != OK) {
-            return NULL;
-        }
-
-        sp<ABuffer> accessUnit = new ABuffer(mMaxSampleSize);
-        CHECK_EQ((status_t)OK, mExtractor->readSampleData(accessUnit));
-
-        accessUnit->meta()->setInt64("timeUs", timeUs);
-
-        CHECK_EQ((status_t)OK, mExtractor->advance());
-
-        return accessUnit;
-    }
-
-protected:
-    virtual ~MediaPacketSource() {
-    }
-
-private:
-    sp<NuMediaExtractor> mExtractor;
-    size_t mMaxSampleSize;
-
-    DISALLOW_EVIL_CONSTRUCTORS(MediaPacketSource);
-};
-
-struct SimplePacketSource : public PacketSource {
-    SimplePacketSource()
-        : mCounter(0) {
-    }
-
-    virtual sp<ABuffer> getNextAccessUnit() {
-        sp<ABuffer> buffer = new ABuffer(4);
-        uint8_t *dst = buffer->data();
-        dst[0] = mCounter >> 24;
-        dst[1] = (mCounter >> 16) & 0xff;
-        dst[2] = (mCounter >> 8) & 0xff;
-        dst[3] = mCounter & 0xff;
-
-        buffer->meta()->setInt64("timeUs", mCounter * 1000000ll / kFrameRate);
-
-        ++mCounter;
-
-        return buffer;
-    }
-
-protected:
-    virtual ~SimplePacketSource() {
-    }
-
-private:
-    enum {
-        kFrameRate = 30
-    };
-
-    uint32_t mCounter;
-
-    DISALLOW_EVIL_CONSTRUCTORS(SimplePacketSource);
-};
-
-struct TestHandler : public AHandler {
-    TestHandler(const sp<ANetworkSession> &netSession);
-
-    void listen();
-    void connect(const char *host, int32_t port);
-
-protected:
-    virtual ~TestHandler();
-    virtual void onMessageReceived(const sp<AMessage> &msg);
-
-private:
-    enum {
-        kWhatListen,
-        kWhatConnect,
-        kWhatReceiverNotify,
-        kWhatSenderNotify,
-        kWhatSendMore,
-        kWhatStop,
-        kWhatTimeSyncerNotify,
-    };
-
-#if 1
-    static const RTPBase::TransportMode kRTPMode = RTPBase::TRANSPORT_UDP;
-    static const RTPBase::TransportMode kRTCPMode = RTPBase::TRANSPORT_UDP;
-#else
-    static const RTPBase::TransportMode kRTPMode = RTPBase::TRANSPORT_TCP;
-    static const RTPBase::TransportMode kRTCPMode = RTPBase::TRANSPORT_NONE;
-#endif
-
-#if 1
-    static const RTPBase::PacketizationMode kPacketizationMode
-        = RTPBase::PACKETIZATION_H264;
-#else
-    static const RTPBase::PacketizationMode kPacketizationMode
-        = RTPBase::PACKETIZATION_NONE;
-#endif
-
-    sp<ANetworkSession> mNetSession;
-    sp<PacketSource> mSource;
-    sp<RTPSender> mSender;
-    sp<RTPReceiver> mReceiver;
-
-    sp<TimeSyncer> mTimeSyncer;
-    bool mTimeSyncerStarted;
-
-    int64_t mFirstTimeRealUs;
-    int64_t mFirstTimeMediaUs;
-
-    int64_t mTimeOffsetUs;
-    bool mTimeOffsetValid;
-
-    status_t readMore();
-
-    DISALLOW_EVIL_CONSTRUCTORS(TestHandler);
-};
-
-TestHandler::TestHandler(const sp<ANetworkSession> &netSession)
-    : mNetSession(netSession),
-      mTimeSyncerStarted(false),
-      mFirstTimeRealUs(-1ll),
-      mFirstTimeMediaUs(-1ll),
-      mTimeOffsetUs(-1ll),
-      mTimeOffsetValid(false) {
-}
-
-TestHandler::~TestHandler() {
-}
-
-void TestHandler::listen() {
-    sp<AMessage> msg = new AMessage(kWhatListen, id());
-    msg->post();
-}
-
-void TestHandler::connect(const char *host, int32_t port) {
-    sp<AMessage> msg = new AMessage(kWhatConnect, id());
-    msg->setString("host", host);
-    msg->setInt32("port", port);
-    msg->post();
-}
-
-static void dumpDelay(int64_t delayMs) {
-    static const int64_t kMinDelayMs = 0;
-    static const int64_t kMaxDelayMs = 300;
-
-    const char *kPattern = "########################################";
-    size_t kPatternSize = strlen(kPattern);
-
-    int n = (kPatternSize * (delayMs - kMinDelayMs))
-                / (kMaxDelayMs - kMinDelayMs);
-
-    if (n < 0) {
-        n = 0;
-    } else if ((size_t)n > kPatternSize) {
-        n = kPatternSize;
-    }
-
-    ALOGI("(%4lld ms) %s\n",
-          delayMs,
-          kPattern + kPatternSize - n);
-}
-
-void TestHandler::onMessageReceived(const sp<AMessage> &msg) {
-    switch (msg->what()) {
-        case kWhatListen:
-        {
-            sp<AMessage> notify = new AMessage(kWhatTimeSyncerNotify, id());
-            mTimeSyncer = new TimeSyncer(mNetSession, notify);
-            looper()->registerHandler(mTimeSyncer);
-
-            notify = new AMessage(kWhatReceiverNotify, id());
-            mReceiver = new RTPReceiver(
-                    mNetSession, notify, RTPReceiver::FLAG_AUTO_CONNECT);
-            looper()->registerHandler(mReceiver);
-
-            CHECK_EQ((status_t)OK,
-                     mReceiver->registerPacketType(33, kPacketizationMode));
-
-            int32_t receiverRTPPort;
-            CHECK_EQ((status_t)OK,
-                     mReceiver->initAsync(
-                         kRTPMode,
-                         kRTCPMode,
-                         &receiverRTPPort));
-
-            printf("picked receiverRTPPort %d\n", receiverRTPPort);
-
-#if 0
-            CHECK_EQ((status_t)OK,
-                     mReceiver->connect(
-                         "127.0.0.1", senderRTPPort, senderRTPPort + 1));
-#endif
-            break;
-        }
-
-        case kWhatConnect:
-        {
-            AString host;
-            CHECK(msg->findString("host", &host));
-
-            sp<AMessage> notify = new AMessage(kWhatTimeSyncerNotify, id());
-            mTimeSyncer = new TimeSyncer(mNetSession, notify);
-            looper()->registerHandler(mTimeSyncer);
-            mTimeSyncer->startServer(8123);
-
-            int32_t receiverRTPPort;
-            CHECK(msg->findInt32("port", &receiverRTPPort));
-
-#if 1
-            mSource = new MediaPacketSource;
-#else
-            mSource = new SimplePacketSource;
-#endif
-
-            notify = new AMessage(kWhatSenderNotify, id());
-            mSender = new RTPSender(mNetSession, notify);
-
-            looper()->registerHandler(mSender);
-
-            int32_t senderRTPPort;
-            CHECK_EQ((status_t)OK,
-                     mSender->initAsync(
-                         host.c_str(),
-                         receiverRTPPort,
-                         kRTPMode,
-                         kRTCPMode == RTPBase::TRANSPORT_NONE
-                            ? -1 : receiverRTPPort + 1,
-                         kRTCPMode,
-                         &senderRTPPort));
-
-            printf("picked senderRTPPort %d\n", senderRTPPort);
-            break;
-        }
-
-        case kWhatSenderNotify:
-        {
-            ALOGI("kWhatSenderNotify");
-
-            int32_t what;
-            CHECK(msg->findInt32("what", &what));
-
-            switch (what) {
-                case RTPSender::kWhatInitDone:
-                {
-                    int32_t err;
-                    CHECK(msg->findInt32("err", &err));
-
-                    ALOGI("RTPSender::initAsync completed w/ err %d", err);
-
-                    if (err == OK) {
-                        err = readMore();
-
-                        if (err != OK) {
-                            (new AMessage(kWhatStop, id()))->post();
-                        }
-                    }
-                    break;
-                }
-
-                case RTPSender::kWhatError:
-                    break;
-            }
-            break;
-        }
-
-        case kWhatReceiverNotify:
-        {
-            ALOGV("kWhatReceiverNotify");
-
-            int32_t what;
-            CHECK(msg->findInt32("what", &what));
-
-            switch (what) {
-                case RTPReceiver::kWhatInitDone:
-                {
-                    int32_t err;
-                    CHECK(msg->findInt32("err", &err));
-
-                    ALOGI("RTPReceiver::initAsync completed w/ err %d", err);
-                    break;
-                }
-
-                case RTPReceiver::kWhatError:
-                    break;
-
-                case RTPReceiver::kWhatAccessUnit:
-                {
-#if 0
-                    if (!mTimeSyncerStarted) {
-                        mTimeSyncer->startClient("172.18.41.216", 8123);
-                        mTimeSyncerStarted = true;
-                    }
-
-                    sp<ABuffer> accessUnit;
-                    CHECK(msg->findBuffer("accessUnit", &accessUnit));
-
-                    int64_t timeUs;
-                    CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
-
-                    if (mTimeOffsetValid) {
-                        timeUs -= mTimeOffsetUs;
-                        int64_t nowUs = ALooper::GetNowUs();
-                        int64_t delayMs = (nowUs - timeUs) / 1000ll;
-
-                        dumpDelay(delayMs);
-                    }
-#endif
-                    break;
-                }
-
-                case RTPReceiver::kWhatPacketLost:
-                    ALOGV("kWhatPacketLost");
-                    break;
-
-                default:
-                    TRESPASS();
-            }
-            break;
-        }
-
-        case kWhatSendMore:
-        {
-            sp<ABuffer> accessUnit;
-            CHECK(msg->findBuffer("accessUnit", &accessUnit));
-
-            CHECK_EQ((status_t)OK,
-                     mSender->queueBuffer(
-                         accessUnit,
-                         33,
-                         kPacketizationMode));
-
-            status_t err = readMore();
-
-            if (err != OK) {
-                (new AMessage(kWhatStop, id()))->post();
-            }
-            break;
-        }
-
-        case kWhatStop:
-        {
-            if (mReceiver != NULL) {
-                looper()->unregisterHandler(mReceiver->id());
-                mReceiver.clear();
-            }
-
-            if (mSender != NULL) {
-                looper()->unregisterHandler(mSender->id());
-                mSender.clear();
-            }
-
-            mSource.clear();
-
-            looper()->stop();
-            break;
-        }
-
-        case kWhatTimeSyncerNotify:
-        {
-            CHECK(msg->findInt64("offset", &mTimeOffsetUs));
-            mTimeOffsetValid = true;
-            break;
-        }
-
-        default:
-            TRESPASS();
-    }
-}
-
-status_t TestHandler::readMore() {
-    sp<ABuffer> accessUnit = mSource->getNextAccessUnit();
-
-    if (accessUnit == NULL) {
-        return ERROR_END_OF_STREAM;
-    }
-
-    int64_t timeUs;
-    CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
-
-    int64_t nowUs = ALooper::GetNowUs();
-    int64_t whenUs;
-
-    if (mFirstTimeRealUs < 0ll) {
-        mFirstTimeRealUs = whenUs = nowUs;
-        mFirstTimeMediaUs = timeUs;
-    } else {
-        whenUs = mFirstTimeRealUs + timeUs - mFirstTimeMediaUs;
-    }
-
-    accessUnit->meta()->setInt64("timeUs", whenUs);
-
-    sp<AMessage> msg = new AMessage(kWhatSendMore, id());
-    msg->setBuffer("accessUnit", accessUnit);
-    msg->post(whenUs - nowUs);
-
-    return OK;
-}
-
-}  // namespace android
-
-static void usage(const char *me) {
-    fprintf(stderr,
-            "usage: %s -c host:port\tconnect to remote host\n"
-            "               -l       \tlisten\n",
-            me);
-}
-
-int main(int argc, char **argv) {
-    using namespace android;
-
-    // srand(time(NULL));
-
-    ProcessState::self()->startThreadPool();
-
-    DataSource::RegisterDefaultSniffers();
-
-    bool listen = false;
-    int32_t connectToPort = -1;
-    AString connectToHost;
-
-    int res;
-    while ((res = getopt(argc, argv, "hc:l")) >= 0) {
-        switch (res) {
-            case 'c':
-            {
-                const char *colonPos = strrchr(optarg, ':');
-
-                if (colonPos == NULL) {
-                    usage(argv[0]);
-                    exit(1);
-                }
-
-                connectToHost.setTo(optarg, colonPos - optarg);
-
-                char *end;
-                connectToPort = strtol(colonPos + 1, &end, 10);
-
-                if (*end != '\0' || end == colonPos + 1
-                        || connectToPort < 1 || connectToPort > 65535) {
-                    fprintf(stderr, "Illegal port specified.\n");
-                    exit(1);
-                }
-                break;
-            }
-
-            case 'l':
-            {
-                listen = true;
-                break;
-            }
-
-            case '?':
-            case 'h':
-                usage(argv[0]);
-                exit(1);
-        }
-    }
-
-    if (!listen && connectToPort < 0) {
-        fprintf(stderr,
-                "You need to select either client or server mode.\n");
-        exit(1);
-    }
-
-    sp<ANetworkSession> netSession = new ANetworkSession;
-    netSession->start();
-
-    sp<ALooper> looper = new ALooper;
-
-    sp<TestHandler> handler = new TestHandler(netSession);
-    looper->registerHandler(handler);
-
-    if (listen) {
-        handler->listen();
-    }
-
-    if (connectToPort >= 0) {
-        handler->connect(connectToHost.c_str(), connectToPort);
-    }
-
-    looper->start(true /* runOnCallingThread */);
-
-    return 0;
-}
-

media/libstagefright/wifi-display/sink/DirectRenderer.cpp

diff --git a/media/libstagefright/wifi-display/sink/DirectRenderer.cpp b/media/libstagefright/wifi-display/sink/DirectRenderer.cpp
deleted file mode 100644
index cdb2267..0000000
--- a/media/libstagefright/wifi-display/sink/DirectRenderer.cpp
+++ /dev/null
@@ -1,653 +0,0 @@
-/*
- * Copyright 2012, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "DirectRenderer"
-#include <utils/Log.h>
-
-#include "DirectRenderer.h"
-
-#include <gui/SurfaceComposerClient.h>
-#include <gui/Surface.h>
-#include <media/AudioTrack.h>
-#include <media/ICrypto.h>
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/foundation/hexdump.h>
-#include <media/stagefright/MediaCodec.h>
-#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/MediaErrors.h>
-
-namespace android {
-
-/*
-   Drives the decoding process using a MediaCodec instance. Input buffers
-   queued by calls to "queueInputBuffer" are fed to the decoder as soon
-   as the decoder is ready for them, the client is notified about output
-   buffers as the decoder spits them out.
-*/
-struct DirectRenderer::DecoderContext : public AHandler {
-    enum {
-        kWhatOutputBufferReady,
-    };
-    DecoderContext(const sp<AMessage> &notify);
-
-    status_t init(
-            const sp<AMessage> &format,
-            const sp<IGraphicBufferProducer> &surfaceTex);
-
-    void queueInputBuffer(const sp<ABuffer> &accessUnit);
-
-    status_t renderOutputBufferAndRelease(size_t index);
-    status_t releaseOutputBuffer(size_t index);
-
-protected:
-    virtual ~DecoderContext();
-
-    virtual void onMessageReceived(const sp<AMessage> &msg);
-
-private:
-    enum {
-        kWhatDecoderNotify,
-    };
-
-    sp<AMessage> mNotify;
-    sp<ALooper> mDecoderLooper;
-    sp<MediaCodec> mDecoder;
-    Vector<sp<ABuffer> > mDecoderInputBuffers;
-    Vector<sp<ABuffer> > mDecoderOutputBuffers;
-    List<size_t> mDecoderInputBuffersAvailable;
-    bool mDecoderNotificationPending;
-
-    List<sp<ABuffer> > mAccessUnits;
-
-    void onDecoderNotify();
-    void scheduleDecoderNotification();
-    void queueDecoderInputBuffers();
-
-    void queueOutputBuffer(
-            size_t index, int64_t timeUs, const sp<ABuffer> &buffer);
-
-    DISALLOW_EVIL_CONSTRUCTORS(DecoderContext);
-};
-
-////////////////////////////////////////////////////////////////////////////////
-
-/*
-   A "push" audio renderer. The primary function of this renderer is to use
-   an AudioTrack in push mode and making sure not to block the event loop
-   be ensuring that calls to AudioTrack::write never block. This is done by
-   estimating an upper bound of data that can be written to the AudioTrack
-   buffer without delay.
-*/
-struct DirectRenderer::AudioRenderer : public AHandler {
-    AudioRenderer(const sp<DecoderContext> &decoderContext);
-
-    void queueInputBuffer(
-            size_t index, int64_t timeUs, const sp<ABuffer> &buffer);
-
-protected:
-    virtual ~AudioRenderer();
-    virtual void onMessageReceived(const sp<AMessage> &msg);
-
-private:
-    enum {
-        kWhatPushAudio,
-    };
-
-    struct BufferInfo {
-        size_t mIndex;
-        int64_t mTimeUs;
-        sp<ABuffer> mBuffer;
-    };
-
-    sp<DecoderContext> mDecoderContext;
-    sp<AudioTrack> mAudioTrack;
-
-    List<BufferInfo> mInputBuffers;
-    bool mPushPending;
-
-    size_t mNumFramesWritten;
-
-    void schedulePushIfNecessary();
-    void onPushAudio();
-
-    ssize_t writeNonBlocking(const uint8_t *data, size_t size);
-
-    DISALLOW_EVIL_CONSTRUCTORS(AudioRenderer);
-};
-
-////////////////////////////////////////////////////////////////////////////////
-
-DirectRenderer::DecoderContext::DecoderContext(const sp<AMessage> &notify)
-    : mNotify(notify),
-      mDecoderNotificationPending(false) {
-}
-
-DirectRenderer::DecoderContext::~DecoderContext() {
-    if (mDecoder != NULL) {
-        mDecoder->release();
-        mDecoder.clear();
-
-        mDecoderLooper->stop();
-        mDecoderLooper.clear();
-    }
-}
-
-status_t DirectRenderer::DecoderContext::init(
-        const sp<AMessage> &format,
-        const sp<IGraphicBufferProducer> &surfaceTex) {
-    CHECK(mDecoder == NULL);
-
-    AString mime;
-    CHECK(format->findString("mime", &mime));
-
-    mDecoderLooper = new ALooper;
-    mDecoderLooper->setName("video codec looper");
-
-    mDecoderLooper->start(
-            false /* runOnCallingThread */,
-            false /* canCallJava */,
-            PRIORITY_DEFAULT);
-
-    mDecoder = MediaCodec::CreateByType(
-            mDecoderLooper, mime.c_str(), false /* encoder */);
-
-    CHECK(mDecoder != NULL);
-
-    status_t err = mDecoder->configure(
-            format,
-            surfaceTex == NULL
-                ? NULL : new Surface(surfaceTex),
-            NULL /* crypto */,
-            0 /* flags */);
-    CHECK_EQ(err, (status_t)OK);
-
-    err = mDecoder->start();
-    CHECK_EQ(err, (status_t)OK);
-
-    err = mDecoder->getInputBuffers(
-            &mDecoderInputBuffers);
-    CHECK_EQ(err, (status_t)OK);
-
-    err = mDecoder->getOutputBuffers(
-            &mDecoderOutputBuffers);
-    CHECK_EQ(err, (status_t)OK);
-
-    scheduleDecoderNotification();
-
-    return OK;
-}
-
-void DirectRenderer::DecoderContext::queueInputBuffer(
-        const sp<ABuffer> &accessUnit) {
-    CHECK(mDecoder != NULL);
-
-    mAccessUnits.push_back(accessUnit);
-    queueDecoderInputBuffers();
-}
-
-status_t DirectRenderer::DecoderContext::renderOutputBufferAndRelease(
-        size_t index) {
-    return mDecoder->renderOutputBufferAndRelease(index);
-}
-
-status_t DirectRenderer::DecoderContext::releaseOutputBuffer(size_t index) {
-    return mDecoder->releaseOutputBuffer(index);
-}
-
-void DirectRenderer::DecoderContext::queueDecoderInputBuffers() {
-    if (mDecoder == NULL) {
-        return;
-    }
-
-    bool submittedMore = false;
-
-    while (!mAccessUnits.empty()
-            && !mDecoderInputBuffersAvailable.empty()) {
-        size_t index = *mDecoderInputBuffersAvailable.begin();
-
-        mDecoderInputBuffersAvailable.erase(
-                mDecoderInputBuffersAvailable.begin());
-
-        sp<ABuffer> srcBuffer = *mAccessUnits.begin();
-        mAccessUnits.erase(mAccessUnits.begin());
-
-        const sp<ABuffer> &dstBuffer =
-            mDecoderInputBuffers.itemAt(index);
-
-        memcpy(dstBuffer->data(), srcBuffer->data(), srcBuffer->size());
-
-        int64_t timeUs;
-        CHECK(srcBuffer->meta()->findInt64("timeUs", &timeUs));
-
-        status_t err = mDecoder->queueInputBuffer(
-                index,
-                0 /* offset */,
-                srcBuffer->size(),
-                timeUs,
-                0 /* flags */);
-        CHECK_EQ(err, (status_t)OK);
-
-        submittedMore = true;
-    }
-
-    if (submittedMore) {
-        scheduleDecoderNotification();
-    }
-}
-
-void DirectRenderer::DecoderContext::onMessageReceived(
-        const sp<AMessage> &msg) {
-    switch (msg->what()) {
-        case kWhatDecoderNotify:
-        {
-            onDecoderNotify();
-            break;
-        }
-
-        default:
-            TRESPASS();
-    }
-}
-
-void DirectRenderer::DecoderContext::onDecoderNotify() {
-    mDecoderNotificationPending = false;
-
-    for (;;) {
-        size_t index;
-        status_t err = mDecoder->dequeueInputBuffer(&index);
-
-        if (err == OK) {
-            mDecoderInputBuffersAvailable.push_back(index);
-        } else if (err == -EAGAIN) {
-            break;
-        } else {
-            TRESPASS();
-        }
-    }
-
-    queueDecoderInputBuffers();
-
-    for (;;) {
-        size_t index;
-        size_t offset;
-        size_t size;
-        int64_t timeUs;
-        uint32_t flags;
-        status_t err = mDecoder->dequeueOutputBuffer(
-                &index,
-                &offset,
-                &size,
-                &timeUs,
-                &flags);
-
-        if (err == OK) {
-            queueOutputBuffer(
-                    index, timeUs, mDecoderOutputBuffers.itemAt(index));
-        } else if (err == INFO_OUTPUT_BUFFERS_CHANGED) {
-            err = mDecoder->getOutputBuffers(
-                    &mDecoderOutputBuffers);
-            CHECK_EQ(err, (status_t)OK);
-        } else if (err == INFO_FORMAT_CHANGED) {
-            // We don't care.
-        } else if (err == -EAGAIN) {
-            break;
-        } else {
-            TRESPASS();
-        }
-    }
-
-    scheduleDecoderNotification();
-}
-
-void DirectRenderer::DecoderContext::scheduleDecoderNotification() {
-    if (mDecoderNotificationPending) {
-        return;
-    }
-
-    sp<AMessage> notify =
-        new AMessage(kWhatDecoderNotify, id());
-
-    mDecoder->requestActivityNotification(notify);
-    mDecoderNotificationPending = true;
-}
-
-void DirectRenderer::DecoderContext::queueOutputBuffer(
-        size_t index, int64_t timeUs, const sp<ABuffer> &buffer) {
-    sp<AMessage> msg = mNotify->dup();
-    msg->setInt32("what", kWhatOutputBufferReady);
-    msg->setSize("index", index);
-    msg->setInt64("timeUs", timeUs);
-    msg->setBuffer("buffer", buffer);
-    msg->post();
-}
-
-////////////////////////////////////////////////////////////////////////////////
-
-DirectRenderer::AudioRenderer::AudioRenderer(
-        const sp<DecoderContext> &decoderContext)
-    : mDecoderContext(decoderContext),
-      mPushPending(false),
-      mNumFramesWritten(0) {
-    mAudioTrack = new AudioTrack(
-            AUDIO_STREAM_DEFAULT,
-            48000.0f,
-            AUDIO_FORMAT_PCM,
-            AUDIO_CHANNEL_OUT_STEREO,
-            (int)0 /* frameCount */);
-
-    CHECK_EQ((status_t)OK, mAudioTrack->initCheck());
-
-    mAudioTrack->start();
-}
-
-DirectRenderer::AudioRenderer::~AudioRenderer() {
-}
-
-void DirectRenderer::AudioRenderer::queueInputBuffer(
-        size_t index, int64_t timeUs, const sp<ABuffer> &buffer) {
-    BufferInfo info;
-    info.mIndex = index;
-    info.mTimeUs = timeUs;
-    info.mBuffer = buffer;
-
-    mInputBuffers.push_back(info);
-    schedulePushIfNecessary();
-}
-
-void DirectRenderer::AudioRenderer::onMessageReceived(
-        const sp<AMessage> &msg) {
-    switch (msg->what()) {
-        case kWhatPushAudio:
-        {
-            onPushAudio();
-            break;
-        }
-
-        default:
-            break;
-    }
-}
-
-void DirectRenderer::AudioRenderer::schedulePushIfNecessary() {
-    if (mPushPending || mInputBuffers.empty()) {
-        return;
-    }
-
-    mPushPending = true;
-
-    uint32_t numFramesPlayed;
-    CHECK_EQ(mAudioTrack->getPosition(&numFramesPlayed),
-             (status_t)OK);
-
-    uint32_t numFramesPendingPlayout = mNumFramesWritten - numFramesPlayed;
-
-    // This is how long the audio sink will have data to
-    // play back.
-    const float msecsPerFrame = 1000.0f / mAudioTrack->getSampleRate();
-
-    int64_t delayUs =
-        msecsPerFrame * numFramesPendingPlayout * 1000ll;
-
-    // Let's give it more data after about half that time
-    // has elapsed.
-    (new AMessage(kWhatPushAudio, id()))->post(delayUs / 2);
-}
-
-void DirectRenderer::AudioRenderer::onPushAudio() {
-    mPushPending = false;
-
-    while (!mInputBuffers.empty()) {
-        const BufferInfo &info = *mInputBuffers.begin();
-
-        ssize_t n = writeNonBlocking(
-                info.mBuffer->data(), info.mBuffer->size());
-
-        if (n < (ssize_t)info.mBuffer->size()) {
-            CHECK_GE(n, 0);
-
-            info.mBuffer->setRange(
-                    info.mBuffer->offset() + n, info.mBuffer->size() - n);
-            break;
-        }
-
-        mDecoderContext->releaseOutputBuffer(info.mIndex);
-
-        mInputBuffers.erase(mInputBuffers.begin());
-    }
-
-    schedulePushIfNecessary();
-}
-
-ssize_t DirectRenderer::AudioRenderer::writeNonBlocking(
-        const uint8_t *data, size_t size) {
-    uint32_t numFramesPlayed;
-    status_t err = mAudioTrack->getPosition(&numFramesPlayed);
-    if (err != OK) {
-        return err;
-    }
-
-    ssize_t numFramesAvailableToWrite =
-        mAudioTrack->frameCount() - (mNumFramesWritten - numFramesPlayed);
-
-    size_t numBytesAvailableToWrite =
-        numFramesAvailableToWrite * mAudioTrack->frameSize();
-
-    if (size > numBytesAvailableToWrite) {
-        size = numBytesAvailableToWrite;
-    }
-
-    CHECK_EQ(mAudioTrack->write(data, size), (ssize_t)size);
-
-    size_t numFramesWritten = size / mAudioTrack->frameSize();
-    mNumFramesWritten += numFramesWritten;
-
-    return size;
-}
-
-////////////////////////////////////////////////////////////////////////////////
-
-DirectRenderer::DirectRenderer(
-        const sp<IGraphicBufferProducer> &bufferProducer)
-    : mSurfaceTex(bufferProducer),
-      mVideoRenderPending(false),
-      mNumFramesLate(0),
-      mNumFrames(0) {
-}
-
-DirectRenderer::~DirectRenderer() {
-}
-
-void DirectRenderer::onMessageReceived(const sp<AMessage> &msg) {
-    switch (msg->what()) {
-        case kWhatDecoderNotify:
-        {
-            onDecoderNotify(msg);
-            break;
-        }
-
-        case kWhatRenderVideo:
-        {
-            onRenderVideo();
-            break;
-        }
-
-        case kWhatQueueAccessUnit:
-            onQueueAccessUnit(msg);
-            break;
-
-        case kWhatSetFormat:
-            onSetFormat(msg);
-            break;
-
-        default:
-            TRESPASS();
-    }
-}
-
-void DirectRenderer::setFormat(size_t trackIndex, const sp<AMessage> &format) {
-    sp<AMessage> msg = new AMessage(kWhatSetFormat, id());
-    msg->setSize("trackIndex", trackIndex);
-    msg->setMessage("format", format);
-    msg->post();
-}
-
-void DirectRenderer::onSetFormat(const sp<AMessage> &msg) {
-    size_t trackIndex;
-    CHECK(msg->findSize("trackIndex", &trackIndex));
-
-    sp<AMessage> format;
-    CHECK(msg->findMessage("format", &format));
-
-    internalSetFormat(trackIndex, format);
-}
-
-void DirectRenderer::internalSetFormat(
-        size_t trackIndex, const sp<AMessage> &format) {
-    CHECK_LT(trackIndex, 2u);
-
-    CHECK(mDecoderContext[trackIndex] == NULL);
-
-    sp<AMessage> notify = new AMessage(kWhatDecoderNotify, id());
-    notify->setSize("trackIndex", trackIndex);
-
-    mDecoderContext[trackIndex] = new DecoderContext(notify);
-    looper()->registerHandler(mDecoderContext[trackIndex]);
-
-    CHECK_EQ((status_t)OK,
-             mDecoderContext[trackIndex]->init(
-                 format, trackIndex == 0 ? mSurfaceTex : NULL));
-
-    if (trackIndex == 1) {
-        // Audio
-        mAudioRenderer = new AudioRenderer(mDecoderContext[1]);
-        looper()->registerHandler(mAudioRenderer);
-    }
-}
-
-void DirectRenderer::queueAccessUnit(
-        size_t trackIndex, const sp<ABuffer> &accessUnit) {
-    sp<AMessage> msg = new AMessage(kWhatQueueAccessUnit, id());
-    msg->setSize("trackIndex", trackIndex);
-    msg->setBuffer("accessUnit", accessUnit);
-    msg->post();
-}
-
-void DirectRenderer::onQueueAccessUnit(const sp<AMessage> &msg) {
-    size_t trackIndex;
-    CHECK(msg->findSize("trackIndex", &trackIndex));
-
-    sp<ABuffer> accessUnit;
-    CHECK(msg->findBuffer("accessUnit", &accessUnit));
-
-    CHECK_LT(trackIndex, 2u);
-    CHECK(mDecoderContext[trackIndex] != NULL);
-
-    mDecoderContext[trackIndex]->queueInputBuffer(accessUnit);
-}
-
-void DirectRenderer::onDecoderNotify(const sp<AMessage> &msg) {
-    size_t trackIndex;
-    CHECK(msg->findSize("trackIndex", &trackIndex));
-
-    int32_t what;
-    CHECK(msg->findInt32("what", &what));
-
-    switch (what) {
-        case DecoderContext::kWhatOutputBufferReady:
-        {
-            size_t index;
-            CHECK(msg->findSize("index", &index));
-
-            int64_t timeUs;
-            CHECK(msg->findInt64("timeUs", &timeUs));
-
-            sp<ABuffer> buffer;
-            CHECK(msg->findBuffer("buffer", &buffer));
-
-            queueOutputBuffer(trackIndex, index, timeUs, buffer);
-            break;
-        }
-
-        default:
-            TRESPASS();
-    }
-}
-
-void DirectRenderer::queueOutputBuffer(
-        size_t trackIndex,
-        size_t index, int64_t timeUs, const sp<ABuffer> &buffer) {
-    if (trackIndex == 1) {
-        // Audio
-        mAudioRenderer->queueInputBuffer(index, timeUs, buffer);
-        return;
-    }
-
-    OutputInfo info;
-    info.mIndex = index;
-    info.mTimeUs = timeUs;
-    info.mBuffer = buffer;
-    mVideoOutputBuffers.push_back(info);
-
-    scheduleVideoRenderIfNecessary();
-}
-
-void DirectRenderer::scheduleVideoRenderIfNecessary() {
-    if (mVideoRenderPending || mVideoOutputBuffers.empty()) {
-        return;
-    }
-
-    mVideoRenderPending = true;
-
-    int64_t timeUs = (*mVideoOutputBuffers.begin()).mTimeUs;
-    int64_t nowUs = ALooper::GetNowUs();
-
-    int64_t delayUs = timeUs - nowUs;
-
-    (new AMessage(kWhatRenderVideo, id()))->post(delayUs);
-}
-
-void DirectRenderer::onRenderVideo() {
-    mVideoRenderPending = false;
-
-    int64_t nowUs = ALooper::GetNowUs();
-
-    while (!mVideoOutputBuffers.empty()) {
-        const OutputInfo &info = *mVideoOutputBuffers.begin();
-
-        if (info.mTimeUs > nowUs) {
-            break;
-        }
-
-        if (info.mTimeUs + 15000ll < nowUs) {
-            ++mNumFramesLate;
-        }
-        ++mNumFrames;
-
-        status_t err =
-            mDecoderContext[0]->renderOutputBufferAndRelease(info.mIndex);
-        CHECK_EQ(err, (status_t)OK);
-
-        mVideoOutputBuffers.erase(mVideoOutputBuffers.begin());
-    }
-
-    scheduleVideoRenderIfNecessary();
-}
-
-}  // namespace android
-

media/libstagefright/wifi-display/sink/DirectRenderer.h

diff --git a/media/libstagefright/wifi-display/sink/DirectRenderer.h b/media/libstagefright/wifi-display/sink/DirectRenderer.h
deleted file mode 100644
index 07c2170..0000000
--- a/media/libstagefright/wifi-display/sink/DirectRenderer.h
+++ /dev/null
@@ -1,87 +0,0 @@
-/*
- * Copyright 2012, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef DIRECT_RENDERER_H_
-
-#define DIRECT_RENDERER_H_
-
-#include <media/stagefright/foundation/AHandler.h>
-
-namespace android {
-
-struct ABuffer;
-struct IGraphicBufferProducer;
-
-// Renders audio and video data queued by calls to "queueAccessUnit".
-struct DirectRenderer : public AHandler {
-    DirectRenderer(const sp<IGraphicBufferProducer> &bufferProducer);
-
-    void setFormat(size_t trackIndex, const sp<AMessage> &format);
-    void queueAccessUnit(size_t trackIndex, const sp<ABuffer> &accessUnit);
-
-protected:
-    virtual void onMessageReceived(const sp<AMessage> &msg);
-    virtual ~DirectRenderer();
-
-private:
-    struct DecoderContext;
-    struct AudioRenderer;
-
-    enum {
-        kWhatDecoderNotify,
-        kWhatRenderVideo,
-        kWhatQueueAccessUnit,
-        kWhatSetFormat,
-    };
-
-    struct OutputInfo {
-        size_t mIndex;
-        int64_t mTimeUs;
-        sp<ABuffer> mBuffer;
-    };
-
-    sp<IGraphicBufferProducer> mSurfaceTex;
-
-    sp<DecoderContext> mDecoderContext[2];
-    List<OutputInfo> mVideoOutputBuffers;
-
-    bool mVideoRenderPending;
-
-    sp<AudioRenderer> mAudioRenderer;
-
-    int32_t mNumFramesLate;
-    int32_t mNumFrames;
-
-    void onDecoderNotify(const sp<AMessage> &msg);
-
-    void queueOutputBuffer(
-            size_t trackIndex,
-            size_t index, int64_t timeUs, const sp<ABuffer> &buffer);
-
-    void scheduleVideoRenderIfNecessary();
-    void onRenderVideo();
-
-    void onSetFormat(const sp<AMessage> &msg);
-    void onQueueAccessUnit(const sp<AMessage> &msg);
-
-    void internalSetFormat(size_t trackIndex, const sp<AMessage> &format);
-
-    DISALLOW_EVIL_CONSTRUCTORS(DirectRenderer);
-};
-
-}  // namespace android
-
-#endif  // DIRECT_RENDERER_H_

media/libstagefright/wifi-display/sink/WifiDisplaySink.cpp

diff --git a/media/libstagefright/wifi-display/sink/WifiDisplaySink.cpp b/media/libstagefright/wifi-display/sink/WifiDisplaySink.cpp
deleted file mode 100644
index bc88f1e..0000000
--- a/media/libstagefright/wifi-display/sink/WifiDisplaySink.cpp
+++ /dev/null
@@ -1,917 +0,0 @@
-/*
- * Copyright 2012, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "WifiDisplaySink"
-#include <utils/Log.h>
-
-#include "WifiDisplaySink.h"
-
-#include "DirectRenderer.h"
-#include "MediaReceiver.h"
-#include "TimeSyncer.h"
-
-#include <cutils/properties.h>
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/foundation/ParsedMessage.h>
-#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/Utils.h>
-
-namespace android {
-
-// static
-const AString WifiDisplaySink::sUserAgent = MakeUserAgent();
-
-WifiDisplaySink::WifiDisplaySink(
-        uint32_t flags,
-        const sp<ANetworkSession> &netSession,
-        const sp<IGraphicBufferProducer> &bufferProducer,
-        const sp<AMessage> &notify)
-    : mState(UNDEFINED),
-      mFlags(flags),
-      mNetSession(netSession),
-      mSurfaceTex(bufferProducer),
-      mNotify(notify),
-      mUsingTCPTransport(false),
-      mUsingTCPInterleaving(false),
-      mSessionID(0),
-      mNextCSeq(1),
-      mIDRFrameRequestPending(false),
-      mTimeOffsetUs(0ll),
-      mTimeOffsetValid(false),
-      mSetupDeferred(false),
-      mLatencyCount(0),
-      mLatencySumUs(0ll),
-      mLatencyMaxUs(0ll),
-      mMaxDelayMs(-1ll) {
-    // We support any and all resolutions, but prefer 720p30
-    mSinkSupportedVideoFormats.setNativeResolution(
-            VideoFormats::RESOLUTION_CEA, 5);  // 1280 x 720 p30
-
-    mSinkSupportedVideoFormats.enableAll();
-}
-
-WifiDisplaySink::~WifiDisplaySink() {
-}
-
-void WifiDisplaySink::start(const char *sourceHost, int32_t sourcePort) {
-    sp<AMessage> msg = new AMessage(kWhatStart, id());
-    msg->setString("sourceHost", sourceHost);
-    msg->setInt32("sourcePort", sourcePort);
-    msg->post();
-}
-
-void WifiDisplaySink::start(const char *uri) {
-    sp<AMessage> msg = new AMessage(kWhatStart, id());
-    msg->setString("setupURI", uri);
-    msg->post();
-}
-
-// static
-bool WifiDisplaySink::ParseURL(
-        const char *url, AString *host, int32_t *port, AString *path,
-        AString *user, AString *pass) {
-    host->clear();
-    *port = 0;
-    path->clear();
-    user->clear();
-    pass->clear();
-
-    if (strncasecmp("rtsp://", url, 7)) {
-        return false;
-    }
-
-    const char *slashPos = strchr(&url[7], '/');
-
-    if (slashPos == NULL) {
-        host->setTo(&url[7]);
-        path->setTo("/");
-    } else {
-        host->setTo(&url[7], slashPos - &url[7]);
-        path->setTo(slashPos);
-    }
-
-    ssize_t atPos = host->find("@");
-
-    if (atPos >= 0) {
-        // Split of user:pass@ from hostname.
-
-        AString userPass(*host, 0, atPos);
-        host->erase(0, atPos + 1);
-
-        ssize_t colonPos = userPass.find(":");
-
-        if (colonPos < 0) {
-            *user = userPass;
-        } else {
-            user->setTo(userPass, 0, colonPos);
-            pass->setTo(userPass, colonPos + 1, userPass.size() - colonPos - 1);
-        }
-    }
-
-    const char *colonPos = strchr(host->c_str(), ':');
-
-    if (colonPos != NULL) {
-        char *end;
-        unsigned long x = strtoul(colonPos + 1, &end, 10);
-
-        if (end == colonPos + 1 || *end != '\0' || x >= 65536) {
-            return false;
-        }
-
-        *port = x;
-
-        size_t colonOffset = colonPos - host->c_str();
-        size_t trailing = host->size() - colonOffset;
-        host->erase(colonOffset, trailing);
-    } else {
-        *port = 554;
-    }
-
-    return true;
-}
-
-void WifiDisplaySink::onMessageReceived(const sp<AMessage> &msg) {
-    switch (msg->what()) {
-        case kWhatStart:
-        {
-            sleep(2);  // XXX
-
-            int32_t sourcePort;
-            CHECK(msg->findString("sourceHost", &mRTSPHost));
-            CHECK(msg->findInt32("sourcePort", &sourcePort));
-
-            sp<AMessage> notify = new AMessage(kWhatRTSPNotify, id());
-
-            status_t err = mNetSession->createRTSPClient(
-                    mRTSPHost.c_str(), sourcePort, notify, &mSessionID);
-            CHECK_EQ(err, (status_t)OK);
-
-            mState = CONNECTING;
-            break;
-        }
-
-        case kWhatRTSPNotify:
-        {
-            int32_t reason;
-            CHECK(msg->findInt32("reason", &reason));
-
-            switch (reason) {
-                case ANetworkSession::kWhatError:
-                {
-                    int32_t sessionID;
-                    CHECK(msg->findInt32("sessionID", &sessionID));
-
-                    int32_t err;
-                    CHECK(msg->findInt32("err", &err));
-
-                    AString detail;
-                    CHECK(msg->findString("detail", &detail));
-
-                    ALOGE("An error occurred in session %d (%d, '%s/%s').",
-                          sessionID,
-                          err,
-                          detail.c_str(),
-                          strerror(-err));
-
-                    if (sessionID == mSessionID) {
-                        ALOGI("Lost control connection.");
-
-                        // The control connection is dead now.
-                        mNetSession->destroySession(mSessionID);
-                        mSessionID = 0;
-
-                        if (mNotify == NULL) {
-                            looper()->stop();
-                        } else {
-                            sp<AMessage> notify = mNotify->dup();
-                            notify->setInt32("what", kWhatDisconnected);
-                            notify->post();
-                        }
-                    }
-                    break;
-                }
-
-                case ANetworkSession::kWhatConnected:
-                {
-                    ALOGI("We're now connected.");
-                    mState = CONNECTED;
-
-                    if (mFlags & FLAG_SPECIAL_MODE) {
-                        sp<AMessage> notify = new AMessage(
-                                kWhatTimeSyncerNotify, id());
-
-                        mTimeSyncer = new TimeSyncer(mNetSession, notify);
-                        looper()->registerHandler(mTimeSyncer);
-
-                        mTimeSyncer->startClient(mRTSPHost.c_str(), 8123);
-                    }
-                    break;
-                }
-
-                case ANetworkSession::kWhatData:
-                {
-                    onReceiveClientData(msg);
-                    break;
-                }
-
-                default:
-                    TRESPASS();
-            }
-            break;
-        }
-
-        case kWhatStop:
-        {
-            looper()->stop();
-            break;
-        }
-
-        case kWhatMediaReceiverNotify:
-        {
-            onMediaReceiverNotify(msg);
-            break;
-        }
-
-        case kWhatTimeSyncerNotify:
-        {
-            int32_t what;
-            CHECK(msg->findInt32("what", &what));
-
-            if (what == TimeSyncer::kWhatTimeOffset) {
-                CHECK(msg->findInt64("offset", &mTimeOffsetUs));
-                mTimeOffsetValid = true;
-
-                if (mSetupDeferred) {
-                    CHECK_EQ((status_t)OK,
-                             sendSetup(
-                                mSessionID,
-                                "rtsp://x.x.x.x:x/wfd1.0/streamid=0"));
-
-                    mSetupDeferred = false;
-                }
-            }
-            break;
-        }
-
-        case kWhatReportLateness:
-        {
-            if (mLatencyCount > 0) {
-                int64_t avgLatencyUs = mLatencySumUs / mLatencyCount;
-
-                ALOGV("avg. latency = %lld ms (max %lld ms)",
-                      avgLatencyUs / 1000ll,
-                      mLatencyMaxUs / 1000ll);
-
-                sp<AMessage> params = new AMessage;
-                params->setInt64("avgLatencyUs", avgLatencyUs);
-                params->setInt64("maxLatencyUs", mLatencyMaxUs);
-                mMediaReceiver->informSender(0 /* trackIndex */, params);
-            }
-
-            mLatencyCount = 0;
-            mLatencySumUs = 0ll;
-            mLatencyMaxUs = 0ll;
-
-            msg->post(kReportLatenessEveryUs);
-            break;
-        }
-
-        default:
-            TRESPASS();
-    }
-}
-
-void WifiDisplaySink::dumpDelay(size_t trackIndex, int64_t timeUs) {
-    int64_t delayMs = (ALooper::GetNowUs() - timeUs) / 1000ll;
-
-    if (delayMs > mMaxDelayMs) {
-        mMaxDelayMs = delayMs;
-    }
-
-    static const int64_t kMinDelayMs = 0;
-    static const int64_t kMaxDelayMs = 300;
-
-    const char *kPattern = "########################################";
-    size_t kPatternSize = strlen(kPattern);
-
-    int n = (kPatternSize * (delayMs - kMinDelayMs))
-                / (kMaxDelayMs - kMinDelayMs);
-
-    if (n < 0) {
-        n = 0;
-    } else if ((size_t)n > kPatternSize) {
-        n = kPatternSize;
-    }
-
-    ALOGI("[%lld]: (%4lld ms / %4lld ms) %s",
-          timeUs / 1000,
-          delayMs,
-          mMaxDelayMs,
-          kPattern + kPatternSize - n);
-}
-
-void WifiDisplaySink::onMediaReceiverNotify(const sp<AMessage> &msg) {
-    int32_t what;
-    CHECK(msg->findInt32("what", &what));
-
-    switch (what) {
-        case MediaReceiver::kWhatInitDone:
-        {
-            status_t err;
-            CHECK(msg->findInt32("err", &err));
-
-            ALOGI("MediaReceiver initialization completed w/ err %d", err);
-            break;
-        }
-
-        case MediaReceiver::kWhatError:
-        {
-            status_t err;
-            CHECK(msg->findInt32("err", &err));
-
-            ALOGE("MediaReceiver signaled error %d", err);
-            break;
-        }
-
-        case MediaReceiver::kWhatAccessUnit:
-        {
-            if (mRenderer == NULL) {
-                mRenderer = new DirectRenderer(mSurfaceTex);
-                looper()->registerHandler(mRenderer);
-            }
-
-            sp<ABuffer> accessUnit;
-            CHECK(msg->findBuffer("accessUnit", &accessUnit));
-
-            int64_t timeUs;
-            CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
-
-            if (!mTimeOffsetValid && !(mFlags & FLAG_SPECIAL_MODE)) {
-                mTimeOffsetUs = timeUs - ALooper::GetNowUs();
-                mTimeOffsetValid = true;
-            }
-
-            CHECK(mTimeOffsetValid);
-
-            // We are the timesync _client_,
-            // client time = server time - time offset.
-            timeUs -= mTimeOffsetUs;
-
-            size_t trackIndex;
-            CHECK(msg->findSize("trackIndex", &trackIndex));
-
-            int64_t nowUs = ALooper::GetNowUs();
-            int64_t delayUs = nowUs - timeUs;
-
-            mLatencySumUs += delayUs;
-            if (mLatencyCount == 0 || delayUs > mLatencyMaxUs) {
-                mLatencyMaxUs = delayUs;
-            }
-            ++mLatencyCount;
-
-            // dumpDelay(trackIndex, timeUs);
-
-            timeUs += 220000ll;  // Assume 220 ms of latency
-            accessUnit->meta()->setInt64("timeUs", timeUs);
-
-            sp<AMessage> format;
-            if (msg->findMessage("format", &format)) {
-                mRenderer->setFormat(trackIndex, format);
-            }
-
-            mRenderer->queueAccessUnit(trackIndex, accessUnit);
-            break;
-        }
-
-        case MediaReceiver::kWhatPacketLost:
-        {
-#if 0
-            if (!mIDRFrameRequestPending) {
-                ALOGI("requesting IDR frame");
-
-                sendIDRFrameRequest(mSessionID);
-            }
-#endif
-            break;
-        }
-
-        default:
-            TRESPASS();
-    }
-}
-
-void WifiDisplaySink::registerResponseHandler(
-        int32_t sessionID, int32_t cseq, HandleRTSPResponseFunc func) {
-    ResponseID id;
-    id.mSessionID = sessionID;
-    id.mCSeq = cseq;
-    mResponseHandlers.add(id, func);
-}
-
-status_t WifiDisplaySink::sendM2(int32_t sessionID) {
-    AString request = "OPTIONS * RTSP/1.0\r\n";
-    AppendCommonResponse(&request, mNextCSeq);
-
-    request.append(
-            "Require: org.wfa.wfd1.0\r\n"
-            "\r\n");
-
-    status_t err =
-        mNetSession->sendRequest(sessionID, request.c_str(), request.size());
-
-    if (err != OK) {
-        return err;
-    }
-
-    registerResponseHandler(
-            sessionID, mNextCSeq, &WifiDisplaySink::onReceiveM2Response);
-
-    ++mNextCSeq;
-
-    return OK;
-}
-
-status_t WifiDisplaySink::onReceiveM2Response(
-        int32_t sessionID, const sp<ParsedMessage> &msg) {
-    int32_t statusCode;
-    if (!msg->getStatusCode(&statusCode)) {
-        return ERROR_MALFORMED;
-    }
-
-    if (statusCode != 200) {
-        return ERROR_UNSUPPORTED;
-    }
-
-    return OK;
-}
-
-status_t WifiDisplaySink::onReceiveSetupResponse(
-        int32_t sessionID, const sp<ParsedMessage> &msg) {
-    int32_t statusCode;
-    if (!msg->getStatusCode(&statusCode)) {
-        return ERROR_MALFORMED;
-    }
-
-    if (statusCode != 200) {
-        return ERROR_UNSUPPORTED;
-    }
-
-    if (!msg->findString("session", &mPlaybackSessionID)) {
-        return ERROR_MALFORMED;
-    }
-
-    if (!ParsedMessage::GetInt32Attribute(
-                mPlaybackSessionID.c_str(),
-                "timeout",
-                &mPlaybackSessionTimeoutSecs)) {
-        mPlaybackSessionTimeoutSecs = -1;
-    }
-
-    ssize_t colonPos = mPlaybackSessionID.find(";");
-    if (colonPos >= 0) {
-        // Strip any options from the returned session id.
-        mPlaybackSessionID.erase(
-                colonPos, mPlaybackSessionID.size() - colonPos);
-    }
-
-    status_t err = configureTransport(msg);
-
-    if (err != OK) {
-        return err;
-    }
-
-    mState = PAUSED;
-
-    return sendPlay(
-            sessionID,
-            "rtsp://x.x.x.x:x/wfd1.0/streamid=0");
-}
-
-status_t WifiDisplaySink::configureTransport(const sp<ParsedMessage> &msg) {
-    if (mUsingTCPTransport && !(mFlags & FLAG_SPECIAL_MODE)) {
-        // In "special" mode we still use a UDP RTCP back-channel that
-        // needs connecting.
-        return OK;
-    }
-
-    AString transport;
-    if (!msg->findString("transport", &transport)) {
-        ALOGE("Missing 'transport' field in SETUP response.");
-        return ERROR_MALFORMED;
-    }
-
-    AString sourceHost;
-    if (!ParsedMessage::GetAttribute(
-                transport.c_str(), "source", &sourceHost)) {
-        sourceHost = mRTSPHost;
-    }
-
-    AString serverPortStr;
-    if (!ParsedMessage::GetAttribute(
-                transport.c_str(), "server_port", &serverPortStr)) {
-        ALOGE("Missing 'server_port' in Transport field.");
-        return ERROR_MALFORMED;
-    }
-
-    int rtpPort, rtcpPort;
-    if (sscanf(serverPortStr.c_str(), "%d-%d", &rtpPort, &rtcpPort) != 2
-            || rtpPort <= 0 || rtpPort > 65535
-            || rtcpPort <=0 || rtcpPort > 65535
-            || rtcpPort != rtpPort + 1) {
-        ALOGE("Invalid server_port description '%s'.",
-                serverPortStr.c_str());
-
-        return ERROR_MALFORMED;
-    }
-
-    if (rtpPort & 1) {
-        ALOGW("Server picked an odd numbered RTP port.");
-    }
-
-    return mMediaReceiver->connectTrack(
-            0 /* trackIndex */, sourceHost.c_str(), rtpPort, rtcpPort);
-}
-
-status_t WifiDisplaySink::onReceivePlayResponse(
-        int32_t sessionID, const sp<ParsedMessage> &msg) {
-    int32_t statusCode;
-    if (!msg->getStatusCode(&statusCode)) {
-        return ERROR_MALFORMED;
-    }
-
-    if (statusCode != 200) {
-        return ERROR_UNSUPPORTED;
-    }
-
-    mState = PLAYING;
-
-    (new AMessage(kWhatReportLateness, id()))->post(kReportLatenessEveryUs);
-
-    return OK;
-}
-
-status_t WifiDisplaySink::onReceiveIDRFrameRequestResponse(
-        int32_t sessionID, const sp<ParsedMessage> &msg) {
-    CHECK(mIDRFrameRequestPending);
-    mIDRFrameRequestPending = false;
-
-    return OK;
-}
-
-void WifiDisplaySink::onReceiveClientData(const sp<AMessage> &msg) {
-    int32_t sessionID;
-    CHECK(msg->findInt32("sessionID", &sessionID));
-
-    sp<RefBase> obj;
-    CHECK(msg->findObject("data", &obj));
-
-    sp<ParsedMessage> data =
-        static_cast<ParsedMessage *>(obj.get());
-
-    ALOGV("session %d received '%s'",
-          sessionID, data->debugString().c_str());
-
-    AString method;
-    AString uri;
-    data->getRequestField(0, &method);
-
-    int32_t cseq;
-    if (!data->findInt32("cseq", &cseq)) {
-        sendErrorResponse(sessionID, "400 Bad Request", -1 /* cseq */);
-        return;
-    }
-
-    if (method.startsWith("RTSP/")) {
-        // This is a response.
-
-        ResponseID id;
-        id.mSessionID = sessionID;
-        id.mCSeq = cseq;
-
-        ssize_t index = mResponseHandlers.indexOfKey(id);
-
-        if (index < 0) {
-            ALOGW("Received unsolicited server response, cseq %d", cseq);
-            return;
-        }
-
-        HandleRTSPResponseFunc func = mResponseHandlers.valueAt(index);
-        mResponseHandlers.removeItemsAt(index);
-
-        status_t err = (this->*func)(sessionID, data);
-        CHECK_EQ(err, (status_t)OK);
-    } else {
-        AString version;
-        data->getRequestField(2, &version);
-        if (!(version == AString("RTSP/1.0"))) {
-            sendErrorResponse(sessionID, "505 RTSP Version not supported", cseq);
-            return;
-        }
-
-        if (method == "OPTIONS") {
-            onOptionsRequest(sessionID, cseq, data);
-        } else if (method == "GET_PARAMETER") {
-            onGetParameterRequest(sessionID, cseq, data);
-        } else if (method == "SET_PARAMETER") {
-            onSetParameterRequest(sessionID, cseq, data);
-        } else {
-            sendErrorResponse(sessionID, "405 Method Not Allowed", cseq);
-        }
-    }
-}
-
-void WifiDisplaySink::onOptionsRequest(
-        int32_t sessionID,
-        int32_t cseq,
-        const sp<ParsedMessage> &data) {
-    AString response = "RTSP/1.0 200 OK\r\n";
-    AppendCommonResponse(&response, cseq);
-    response.append("Public: org.wfa.wfd1.0, GET_PARAMETER, SET_PARAMETER\r\n");
-    response.append("\r\n");
-
-    status_t err = mNetSession->sendRequest(sessionID, response.c_str());
-    CHECK_EQ(err, (status_t)OK);
-
-    err = sendM2(sessionID);
-    CHECK_EQ(err, (status_t)OK);
-}
-
-void WifiDisplaySink::onGetParameterRequest(
-        int32_t sessionID,
-        int32_t cseq,
-        const sp<ParsedMessage> &data) {
-    AString body;
-
-    if (mState == CONNECTED) {
-        mUsingTCPTransport = false;
-        mUsingTCPInterleaving = false;
-
-        char val[PROPERTY_VALUE_MAX];
-        if (property_get("media.wfd-sink.tcp-mode", val, NULL)) {
-            if (!strcasecmp("true", val) || !strcmp("1", val)) {
-                ALOGI("Using TCP unicast transport.");
-                mUsingTCPTransport = true;
-                mUsingTCPInterleaving = false;
-            } else if (!strcasecmp("interleaved", val)) {
-                ALOGI("Using TCP interleaved transport.");
-                mUsingTCPTransport = true;
-                mUsingTCPInterleaving = true;
-            }
-        } else if (mFlags & FLAG_SPECIAL_MODE) {
-            mUsingTCPTransport = true;
-        }
-
-        body = "wfd_video_formats: ";
-        body.append(mSinkSupportedVideoFormats.getFormatSpec());
-
-        body.append(
-                "\r\nwfd_audio_codecs: AAC 0000000F 00\r\n"
-                "wfd_client_rtp_ports: RTP/AVP/");
-
-        if (mUsingTCPTransport) {
-            body.append("TCP;");
-            if (mUsingTCPInterleaving) {
-                body.append("interleaved");
-            } else {
-                body.append("unicast 19000 0");
-            }
-        } else {
-            body.append("UDP;unicast 19000 0");
-        }
-
-        body.append(" mode=play\r\n");
-    }
-
-    AString response = "RTSP/1.0 200 OK\r\n";
-    AppendCommonResponse(&response, cseq);
-    response.append("Content-Type: text/parameters\r\n");
-    response.append(StringPrintf("Content-Length: %d\r\n", body.size()));
-    response.append("\r\n");
-    response.append(body);
-
-    status_t err = mNetSession->sendRequest(sessionID, response.c_str());
-    CHECK_EQ(err, (status_t)OK);
-}
-
-status_t WifiDisplaySink::sendSetup(int32_t sessionID, const char *uri) {
-    sp<AMessage> notify = new AMessage(kWhatMediaReceiverNotify, id());
-
-    mMediaReceiverLooper = new ALooper;
-    mMediaReceiverLooper->setName("media_receiver");
-
-    mMediaReceiverLooper->start(
-            false /* runOnCallingThread */,
-            false /* canCallJava */,
-            PRIORITY_AUDIO);
-
-    mMediaReceiver = new MediaReceiver(mNetSession, notify);
-    mMediaReceiverLooper->registerHandler(mMediaReceiver);
-
-    RTPReceiver::TransportMode rtpMode = RTPReceiver::TRANSPORT_UDP;
-    if (mUsingTCPTransport) {
-        if (mUsingTCPInterleaving) {
-            rtpMode = RTPReceiver::TRANSPORT_TCP_INTERLEAVED;
-        } else {
-            rtpMode = RTPReceiver::TRANSPORT_TCP;
-        }
-    }
-
-    int32_t localRTPPort;
-    status_t err = mMediaReceiver->addTrack(
-            rtpMode, RTPReceiver::TRANSPORT_UDP /* rtcpMode */, &localRTPPort);
-
-    if (err == OK) {
-        err = mMediaReceiver->initAsync(MediaReceiver::MODE_TRANSPORT_STREAM);
-    }
-
-    if (err != OK) {
-        mMediaReceiverLooper->unregisterHandler(mMediaReceiver->id());
-        mMediaReceiver.clear();
-
-        mMediaReceiverLooper->stop();
-        mMediaReceiverLooper.clear();
-
-        return err;
-    }
-
-    AString request = StringPrintf("SETUP %s RTSP/1.0\r\n", uri);
-
-    AppendCommonResponse(&request, mNextCSeq);
-
-    if (rtpMode == RTPReceiver::TRANSPORT_TCP_INTERLEAVED) {
-        request.append("Transport: RTP/AVP/TCP;interleaved=0-1\r\n");
-    } else if (rtpMode == RTPReceiver::TRANSPORT_TCP) {
-        if (mFlags & FLAG_SPECIAL_MODE) {
-            // This isn't quite true, since the RTP connection is through TCP
-            // and the RTCP connection through UDP...
-            request.append(
-                    StringPrintf(
-                        "Transport: RTP/AVP/TCP;unicast;client_port=%d-%d\r\n",
-                        localRTPPort, localRTPPort + 1));
-        } else {
-            request.append(
-                    StringPrintf(
-                        "Transport: RTP/AVP/TCP;unicast;client_port=%d\r\n",
-                        localRTPPort));
-        }
-    } else {
-        request.append(
-                StringPrintf(
-                    "Transport: RTP/AVP/UDP;unicast;client_port=%d-%d\r\n",
-                    localRTPPort,
-                    localRTPPort + 1));
-    }
-
-    request.append("\r\n");
-
-    ALOGV("request = '%s'", request.c_str());
-
-    err = mNetSession->sendRequest(sessionID, request.c_str(), request.size());
-
-    if (err != OK) {
-        return err;
-    }
-
-    registerResponseHandler(
-            sessionID, mNextCSeq, &WifiDisplaySink::onReceiveSetupResponse);
-
-    ++mNextCSeq;
-
-    return OK;
-}
-
-status_t WifiDisplaySink::sendPlay(int32_t sessionID, const char *uri) {
-    AString request = StringPrintf("PLAY %s RTSP/1.0\r\n", uri);
-
-    AppendCommonResponse(&request, mNextCSeq);
-
-    request.append(StringPrintf("Session: %s\r\n", mPlaybackSessionID.c_str()));
-    request.append("\r\n");
-
-    status_t err =
-        mNetSession->sendRequest(sessionID, request.c_str(), request.size());
-
-    if (err != OK) {
-        return err;
-    }
-
-    registerResponseHandler(
-            sessionID, mNextCSeq, &WifiDisplaySink::onReceivePlayResponse);
-
-    ++mNextCSeq;
-
-    return OK;
-}
-
-status_t WifiDisplaySink::sendIDRFrameRequest(int32_t sessionID) {
-    CHECK(!mIDRFrameRequestPending);
-
-    AString request = "SET_PARAMETER rtsp://localhost/wfd1.0 RTSP/1.0\r\n";
-
-    AppendCommonResponse(&request, mNextCSeq);
-
-    AString content = "wfd_idr_request\r\n";
-
-    request.append(StringPrintf("Session: %s\r\n", mPlaybackSessionID.c_str()));
-    request.append(StringPrintf("Content-Length: %d\r\n", content.size()));
-    request.append("\r\n");
-    request.append(content);
-
-    status_t err =
-        mNetSession->sendRequest(sessionID, request.c_str(), request.size());
-
-    if (err != OK) {
-        return err;
-    }
-
-    registerResponseHandler(
-            sessionID,
-            mNextCSeq,
-            &WifiDisplaySink::onReceiveIDRFrameRequestResponse);
-
-    ++mNextCSeq;
-
-    mIDRFrameRequestPending = true;
-
-    return OK;
-}
-
-void WifiDisplaySink::onSetParameterRequest(
-        int32_t sessionID,
-        int32_t cseq,
-        const sp<ParsedMessage> &data) {
-    const char *content = data->getContent();
-
-    if (strstr(content, "wfd_trigger_method: SETUP\r\n") != NULL) {
-        if ((mFlags & FLAG_SPECIAL_MODE) && !mTimeOffsetValid) {
-            mSetupDeferred = true;
-        } else {
-            status_t err =
-                sendSetup(
-                        sessionID,
-                        "rtsp://x.x.x.x:x/wfd1.0/streamid=0");
-
-            CHECK_EQ(err, (status_t)OK);
-        }
-    }
-
-    AString response = "RTSP/1.0 200 OK\r\n";
-    AppendCommonResponse(&response, cseq);
-    response.append("\r\n");
-
-    status_t err = mNetSession->sendRequest(sessionID, response.c_str());
-    CHECK_EQ(err, (status_t)OK);
-}
-
-void WifiDisplaySink::sendErrorResponse(
-        int32_t sessionID,
-        const char *errorDetail,
-        int32_t cseq) {
-    AString response;
-    response.append("RTSP/1.0 ");
-    response.append(errorDetail);
-    response.append("\r\n");
-
-    AppendCommonResponse(&response, cseq);
-
-    response.append("\r\n");
-
-    status_t err = mNetSession->sendRequest(sessionID, response.c_str());
-    CHECK_EQ(err, (status_t)OK);
-}
-
-// static
-void WifiDisplaySink::AppendCommonResponse(AString *response, int32_t cseq) {
-    time_t now = time(NULL);
-    struct tm *now2 = gmtime(&now);
-    char buf[128];
-    strftime(buf, sizeof(buf), "%a, %d %b %Y %H:%M:%S %z", now2);
-
-    response->append("Date: ");
-    response->append(buf);
-    response->append("\r\n");
-
-    response->append(StringPrintf("User-Agent: %s\r\n", sUserAgent.c_str()));
-
-    if (cseq >= 0) {
-        response->append(StringPrintf("CSeq: %d\r\n", cseq));
-    }
-}
-
-}  // namespace android

media/libstagefright/wifi-display/sink/WifiDisplaySink.h

diff --git a/media/libstagefright/wifi-display/sink/WifiDisplaySink.h b/media/libstagefright/wifi-display/sink/WifiDisplaySink.h
deleted file mode 100644
index dc1fc32..0000000
--- a/media/libstagefright/wifi-display/sink/WifiDisplaySink.h
+++ /dev/null
@@ -1,195 +0,0 @@
-/*
- * Copyright 2012, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef WIFI_DISPLAY_SINK_H_
-
-#define WIFI_DISPLAY_SINK_H_
-
-#include "VideoFormats.h"
-
-#include <gui/Surface.h>
-#include <media/stagefright/foundation/AHandler.h>
-#include <media/stagefright/foundation/ANetworkSession.h>
-
-namespace android {
-
-struct AMessage;
-struct DirectRenderer;
-struct MediaReceiver;
-struct ParsedMessage;
-struct TimeSyncer;
-
-// Represents the RTSP client acting as a wifi display sink.
-// Connects to a wifi display source and renders the incoming
-// transport stream using a MediaPlayer instance.
-struct WifiDisplaySink : public AHandler {
-    enum {
-        kWhatDisconnected,
-    };
-
-    enum Flags {
-        FLAG_SPECIAL_MODE = 1,
-    };
-
-    // If no notification message is specified (notify == NULL)
-    // the sink will stop its looper() once the session ends,
-    // otherwise it will post an appropriate notification but leave
-    // the looper() running.
-    WifiDisplaySink(
-            uint32_t flags,
-            const sp<ANetworkSession> &netSession,
-            const sp<IGraphicBufferProducer> &bufferProducer = NULL,
-            const sp<AMessage> &notify = NULL);
-
-    void start(const char *sourceHost, int32_t sourcePort);
-    void start(const char *uri);
-
-protected:
-    virtual ~WifiDisplaySink();
-    virtual void onMessageReceived(const sp<AMessage> &msg);
-
-private:
-    enum State {
-        UNDEFINED,
-        CONNECTING,
-        CONNECTED,
-        PAUSED,
-        PLAYING,
-    };
-
-    enum {
-        kWhatStart,
-        kWhatRTSPNotify,
-        kWhatStop,
-        kWhatMediaReceiverNotify,
-        kWhatTimeSyncerNotify,
-        kWhatReportLateness,
-    };
-
-    struct ResponseID {
-        int32_t mSessionID;
-        int32_t mCSeq;
-
-        bool operator<(const ResponseID &other) const {
-            return mSessionID < other.mSessionID
-                || (mSessionID == other.mSessionID
-                        && mCSeq < other.mCSeq);
-        }
-    };
-
-    typedef status_t (WifiDisplaySink::*HandleRTSPResponseFunc)(
-            int32_t sessionID, const sp<ParsedMessage> &msg);
-
-    static const int64_t kReportLatenessEveryUs = 1000000ll;
-
-    static const AString sUserAgent;
-
-    State mState;
-    uint32_t mFlags;
-    VideoFormats mSinkSupportedVideoFormats;
-    sp<ANetworkSession> mNetSession;
-    sp<IGraphicBufferProducer> mSurfaceTex;
-    sp<AMessage> mNotify;
-    sp<TimeSyncer> mTimeSyncer;
-    bool mUsingTCPTransport;
-    bool mUsingTCPInterleaving;
-    AString mRTSPHost;
-    int32_t mSessionID;
-
-    int32_t mNextCSeq;
-
-    KeyedVector<ResponseID, HandleRTSPResponseFunc> mResponseHandlers;
-
-    sp<ALooper> mMediaReceiverLooper;
-    sp<MediaReceiver> mMediaReceiver;
-    sp<DirectRenderer> mRenderer;
-
-    AString mPlaybackSessionID;
-    int32_t mPlaybackSessionTimeoutSecs;
-
-    bool mIDRFrameRequestPending;
-
-    int64_t mTimeOffsetUs;
-    bool mTimeOffsetValid;
-
-    bool mSetupDeferred;
-
-    size_t mLatencyCount;
-    int64_t mLatencySumUs;
-    int64_t mLatencyMaxUs;
-
-    int64_t mMaxDelayMs;
-
-    status_t sendM2(int32_t sessionID);
-    status_t sendSetup(int32_t sessionID, const char *uri);
-    status_t sendPlay(int32_t sessionID, const char *uri);
-    status_t sendIDRFrameRequest(int32_t sessionID);
-
-    status_t onReceiveM2Response(
-            int32_t sessionID, const sp<ParsedMessage> &msg);
-
-    status_t onReceiveSetupResponse(
-            int32_t sessionID, const sp<ParsedMessage> &msg);
-
-    status_t configureTransport(const sp<ParsedMessage> &msg);
-
-    status_t onReceivePlayResponse(
-            int32_t sessionID, const sp<ParsedMessage> &msg);
-
-    status_t onReceiveIDRFrameRequestResponse(
-            int32_t sessionID, const sp<ParsedMessage> &msg);
-
-    void registerResponseHandler(
-            int32_t sessionID, int32_t cseq, HandleRTSPResponseFunc func);
-
-    void onReceiveClientData(const sp<AMessage> &msg);
-
-    void onOptionsRequest(
-            int32_t sessionID,
-            int32_t cseq,
-            const sp<ParsedMessage> &data);
-
-    void onGetParameterRequest(
-            int32_t sessionID,
-            int32_t cseq,
-            const sp<ParsedMessage> &data);
-
-    void onSetParameterRequest(
-            int32_t sessionID,
-            int32_t cseq,
-            const sp<ParsedMessage> &data);
-
-    void onMediaReceiverNotify(const sp<AMessage> &msg);
-
-    void sendErrorResponse(
-            int32_t sessionID,
-            const char *errorDetail,
-            int32_t cseq);
-
-    static void AppendCommonResponse(AString *response, int32_t cseq);
-
-    bool ParseURL(
-            const char *url, AString *host, int32_t *port, AString *path,
-            AString *user, AString *pass);
-
-    void dumpDelay(size_t trackIndex, int64_t timeUs);
-
-    DISALLOW_EVIL_CONSTRUCTORS(WifiDisplaySink);
-};
-
-}  // namespace android
-
-#endif  // WIFI_DISPLAY_SINK_H_

media/libstagefright/wifi-display/source/PlaybackSession.cpp

diff --git a/media/libstagefright/wifi-display/source/PlaybackSession.cpp b/media/libstagefright/wifi-display/source/PlaybackSession.cpp
index 286ea13..1a5acba 100644
--- a/media/libstagefright/wifi-display/source/PlaybackSession.cpp
+++ b/media/libstagefright/wifi-display/source/PlaybackSession.cpp
@@ -29,6 +29,7 @@
 #include <binder/IServiceManager.h>
 #include <cutils/properties.h>
 #include <media/IHDCP.h>
+#include <media/IMediaHTTPService.h>
 #include <media/stagefright/foundation/ABitReader.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -749,7 +750,8 @@
 
     mExtractor = new NuMediaExtractor;
 
-    status_t err = mExtractor->setDataSource(mMediaPath.c_str());
+    status_t err = mExtractor->setDataSource(
+            NULL /* httpService */, mMediaPath.c_str());
 
     if (err != OK) {
         return err;

media/libstagefright/wifi-display/source/RepeaterSource.cpp

diff --git a/media/libstagefright/wifi-display/source/RepeaterSource.cpp b/media/libstagefright/wifi-display/source/RepeaterSource.cpp
index cc8dee3..59d7e6e 100644
--- a/media/libstagefright/wifi-display/source/RepeaterSource.cpp
+++ b/media/libstagefright/wifi-display/source/RepeaterSource.cpp
@@ -79,6 +79,8 @@
 
     ALOGV("stopping");
 
+    status_t err = mSource->stop();
+
     if (mLooper != NULL) {
         mLooper->stop();
         mLooper.clear();
@@ -92,7 +94,6 @@
         mBuffer = NULL;
     }
 
-    status_t err = mSource->stop();
 
     ALOGV("stopped");
 

media/libstagefright/wifi-display/source/TSPacketizer.cpp

diff --git a/media/libstagefright/wifi-display/source/TSPacketizer.cpp b/media/libstagefright/wifi-display/source/TSPacketizer.cpp
index c674700..50d317a 100644
--- a/media/libstagefright/wifi-display/source/TSPacketizer.cpp
+++ b/media/libstagefright/wifi-display/source/TSPacketizer.cpp
@@ -216,7 +216,7 @@
     uint8_t *ptr = dup->data();
 
     *ptr++ = 0xff;
-    *ptr++ = 0xf1;  // b11110001, ID=0, layer=0, protection_absent=1
+    *ptr++ = 0xf9;  // b11111001, ID=1(MPEG-2), layer=0, protection_absent=1
 
     *ptr++ =
         profile << 6
@@ -565,7 +565,7 @@
             }
         }
 
-        // size_t numPaddingBytes = sizeAvailableForPayload - numBytesOfPayload;
+        size_t numPaddingBytes = sizeAvailableForPayload - numBytesOfPayload;
         ALOGV("packet 1 contains %zd padding bytes and %zd bytes of payload",
               numPaddingBytes, numBytesOfPayload);
 

media/libstagefright/wifi-display/source/WifiDisplaySource.cpp

diff --git a/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp b/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp
index d72349d..da405e2 100644
--- a/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp
+++ b/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp
@@ -22,7 +22,6 @@
 #include "PlaybackSession.h"
 #include "Parameters.h"
 #include "rtp/RTPSender.h"
-#include "TimeSyncer.h"
 
 #include <binder/IServiceManager.h>
 #include <gui/IGraphicBufferProducer.h>
@@ -173,15 +172,7 @@
                 }
             }
 
-            if (err == OK) {
-                sp<AMessage> notify = new AMessage(kWhatTimeSyncerNotify, id());
-                mTimeSyncer = new TimeSyncer(mNetSession, notify);
-                looper()->registerHandler(mTimeSyncer);
-
-                mTimeSyncer->startServer(8123);
-
-                mState = AWAITING_CLIENT_CONNECTION;
-            }
+            mState = AWAITING_CLIENT_CONNECTION;
 
             sp<AMessage> response = new AMessage;
             response->setInt32("err", err);
@@ -556,11 +547,6 @@
             break;
         }
 
-        case kWhatTimeSyncerNotify:
-        {
-            break;
-        }
-
         default:
             TRESPASS();
     }
@@ -760,7 +746,7 @@
 }
 
 status_t WifiDisplaySource::onReceiveM1Response(
-        int32_t sessionID, const sp<ParsedMessage> &msg) {
+        int32_t /* sessionID */, const sp<ParsedMessage> &msg) {
     int32_t statusCode;
     if (!msg->getStatusCode(&statusCode)) {
         return ERROR_MALFORMED;
@@ -1005,7 +991,7 @@
 }
 
 status_t WifiDisplaySource::onReceiveM5Response(
-        int32_t sessionID, const sp<ParsedMessage> &msg) {
+        int32_t /* sessionID */, const sp<ParsedMessage> &msg) {
     int32_t statusCode;
     if (!msg->getStatusCode(&statusCode)) {
         return ERROR_MALFORMED;
@@ -1019,7 +1005,7 @@
 }
 
 status_t WifiDisplaySource::onReceiveM16Response(
-        int32_t sessionID, const sp<ParsedMessage> &msg) {
+        int32_t sessionID, const sp<ParsedMessage> & /* msg */) {
     // If only the response was required to include a "Session:" header...
 
     CHECK_EQ(sessionID, mClientSessionID);
@@ -1694,7 +1680,7 @@
 }
 
 void WifiDisplaySource::HDCPObserver::notify(
-        int msg, int ext1, int ext2, const Parcel *obj) {
+        int msg, int ext1, int ext2, const Parcel * /* obj */) {
     sp<AMessage> notify = mNotify->dup();
     notify->setInt32("msg", msg);
     notify->setInt32("ext1", ext1);

media/libstagefright/wifi-display/source/WifiDisplaySource.h

diff --git a/media/libstagefright/wifi-display/source/WifiDisplaySource.h b/media/libstagefright/wifi-display/source/WifiDisplaySource.h
index 4f11712..750265f 100644
--- a/media/libstagefright/wifi-display/source/WifiDisplaySource.h
+++ b/media/libstagefright/wifi-display/source/WifiDisplaySource.h
@@ -30,7 +30,6 @@
 struct IHDCP;
 struct IRemoteDisplayClient;
 struct ParsedMessage;
-struct TimeSyncer;
 
 // Represents the RTSP server acting as a wifi display source.
 // Manages incoming connections, sets up Playback sessions as necessary.
@@ -83,7 +82,6 @@
         kWhatHDCPNotify,
         kWhatFinishStop2,
         kWhatTeardownTriggerTimedOut,
-        kWhatTimeSyncerNotify,
     };
 
     struct ResponseID {
@@ -120,7 +118,6 @@
     sp<ANetworkSession> mNetSession;
     sp<IRemoteDisplayClient> mClient;
     AString mMediaPath;
-    sp<TimeSyncer> mTimeSyncer;
     struct in_addr mInterfaceAddr;
     int32_t mSessionID;
 

media/libstagefright/wifi-display/udptest.cpp

diff --git a/media/libstagefright/wifi-display/udptest.cpp b/media/libstagefright/wifi-display/udptest.cpp
deleted file mode 100644
index 61eb9f9..0000000
--- a/media/libstagefright/wifi-display/udptest.cpp
+++ /dev/null
@@ -1,116 +0,0 @@
-/*
- * Copyright 2012, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NEBUG 0
-#define LOG_TAG "udptest"
-#include <utils/Log.h>
-
-#include "TimeSyncer.h"
-
-#include <binder/ProcessState.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/foundation/ANetworkSession.h>
-
-namespace android {
-
-}  // namespace android
-
-static void usage(const char *me) {
-    fprintf(stderr,
-            "usage: %s -c host[:port]\tconnect to test server\n"
-            "           -l            \tcreate a test server\n",
-            me);
-}
-
-int main(int argc, char **argv) {
-    using namespace android;
-
-    ProcessState::self()->startThreadPool();
-
-    int32_t localPort = -1;
-    int32_t connectToPort = -1;
-    AString connectToHost;
-
-    int res;
-    while ((res = getopt(argc, argv, "hc:l:")) >= 0) {
-        switch (res) {
-            case 'c':
-            {
-                const char *colonPos = strrchr(optarg, ':');
-
-                if (colonPos == NULL) {
-                    connectToHost = optarg;
-                    connectToPort = 49152;
-                } else {
-                    connectToHost.setTo(optarg, colonPos - optarg);
-
-                    char *end;
-                    connectToPort = strtol(colonPos + 1, &end, 10);
-
-                    if (*end != '\0' || end == colonPos + 1
-                            || connectToPort < 1 || connectToPort > 65535) {
-                        fprintf(stderr, "Illegal port specified.\n");
-                        exit(1);
-                    }
-                }
-                break;
-            }
-
-            case 'l':
-            {
-                char *end;
-                localPort = strtol(optarg, &end, 10);
-
-                if (*end != '\0' || end == optarg
-                        || localPort < 1 || localPort > 65535) {
-                    fprintf(stderr, "Illegal port specified.\n");
-                    exit(1);
-                }
-                break;
-            }
-
-            case '?':
-            case 'h':
-                usage(argv[0]);
-                exit(1);
-        }
-    }
-
-    if (localPort < 0 && connectToPort < 0) {
-        fprintf(stderr,
-                "You need to select either client or server mode.\n");
-        exit(1);
-    }
-
-    sp<ANetworkSession> netSession = new ANetworkSession;
-    netSession->start();
-
-    sp<ALooper> looper = new ALooper;
-
-    sp<TimeSyncer> handler = new TimeSyncer(netSession, NULL /* notify */);
-    looper->registerHandler(handler);
-
-    if (localPort >= 0) {
-        handler->startServer(localPort);
-    } else {
-        handler->startClient(connectToHost.c_str(), connectToPort);
-    }
-
-    looper->start(true /* runOnCallingThread */);
-
-    return 0;
-}
-

media/libstagefright/wifi-display/wfd.cpp

diff --git a/media/libstagefright/wifi-display/wfd.cpp b/media/libstagefright/wifi-display/wfd.cpp
deleted file mode 100644
index 52e4e26..0000000
--- a/media/libstagefright/wifi-display/wfd.cpp
+++ /dev/null
@@ -1,363 +0,0 @@
-/*
- * Copyright 2012, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "wfd"
-#include <utils/Log.h>
-
-#include "sink/WifiDisplaySink.h"
-#include "source/WifiDisplaySource.h"
-
-#include <binder/ProcessState.h>
-#include <binder/IServiceManager.h>
-#include <gui/ISurfaceComposer.h>
-#include <gui/SurfaceComposerClient.h>
-#include <media/AudioSystem.h>
-#include <media/IMediaPlayerService.h>
-#include <media/IRemoteDisplay.h>
-#include <media/IRemoteDisplayClient.h>
-#include <media/stagefright/DataSource.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <ui/DisplayInfo.h>
-
-namespace android {
-
-static void usage(const char *me) {
-    fprintf(stderr,
-            "usage:\n"
-            "           %s -c host[:port]\tconnect to wifi source\n"
-            "               -u uri        \tconnect to an rtsp uri\n"
-            "               -l ip[:port] \tlisten on the specified port "
-            "               -f(ilename)  \tstream media "
-            "(create a sink)\n"
-            "               -s(pecial)   \trun in 'special' mode\n",
-            me);
-}
-
-struct RemoteDisplayClient : public BnRemoteDisplayClient {
-    RemoteDisplayClient();
-
-    virtual void onDisplayConnected(
-            const sp<IGraphicBufferProducer> &bufferProducer,
-            uint32_t width,
-            uint32_t height,
-            uint32_t flags,
-            uint32_t session);
-
-    virtual void onDisplayDisconnected();
-    virtual void onDisplayError(int32_t error);
-
-    void waitUntilDone();
-
-protected:
-    virtual ~RemoteDisplayClient();
-
-private:
-    Mutex mLock;
-    Condition mCondition;
-
-    bool mDone;
-
-    sp<SurfaceComposerClient> mComposerClient;
-    sp<IGraphicBufferProducer> mSurfaceTexture;
-    sp<IBinder> mDisplayBinder;
-
-    DISALLOW_EVIL_CONSTRUCTORS(RemoteDisplayClient);
-};
-
-RemoteDisplayClient::RemoteDisplayClient()
-    : mDone(false) {
-    mComposerClient = new SurfaceComposerClient;
-    CHECK_EQ(mComposerClient->initCheck(), (status_t)OK);
-}
-
-RemoteDisplayClient::~RemoteDisplayClient() {
-}
-
-void RemoteDisplayClient::onDisplayConnected(
-        const sp<IGraphicBufferProducer> &bufferProducer,
-        uint32_t width,
-        uint32_t height,
-        uint32_t flags,
-        uint32_t session) {
-    ALOGI("onDisplayConnected width=%u, height=%u, flags = 0x%08x, session = %d",
-          width, height, flags, session);
-
-    if (bufferProducer != NULL) {
-        mSurfaceTexture = bufferProducer;
-        mDisplayBinder = mComposerClient->createDisplay(
-                String8("foo"), false /* secure */);
-
-        SurfaceComposerClient::openGlobalTransaction();
-        mComposerClient->setDisplaySurface(mDisplayBinder, mSurfaceTexture);
-
-        Rect layerStackRect(1280, 720);  // XXX fix this.
-        Rect displayRect(1280, 720);
-
-        mComposerClient->setDisplayProjection(
-                mDisplayBinder, 0 /* 0 degree rotation */,
-                layerStackRect,
-                displayRect);
-
-        SurfaceComposerClient::closeGlobalTransaction();
-    }
-}
-
-void RemoteDisplayClient::onDisplayDisconnected() {
-    ALOGI("onDisplayDisconnected");
-
-    Mutex::Autolock autoLock(mLock);
-    mDone = true;
-    mCondition.broadcast();
-}
-
-void RemoteDisplayClient::onDisplayError(int32_t error) {
-    ALOGI("onDisplayError error=%d", error);
-
-    Mutex::Autolock autoLock(mLock);
-    mDone = true;
-    mCondition.broadcast();
-}
-
-void RemoteDisplayClient::waitUntilDone() {
-    Mutex::Autolock autoLock(mLock);
-    while (!mDone) {
-        mCondition.wait(mLock);
-    }
-}
-
-static void createSource(const AString &addr, int32_t port) {
-    sp<IServiceManager> sm = defaultServiceManager();
-    sp<IBinder> binder = sm->getService(String16("media.player"));
-    sp<IMediaPlayerService> service =
-        interface_cast<IMediaPlayerService>(binder);
-
-    CHECK(service.get() != NULL);
-
-    String8 iface;
-    iface.append(addr.c_str());
-    iface.append(StringPrintf(":%d", port).c_str());
-
-    sp<RemoteDisplayClient> client = new RemoteDisplayClient;
-    sp<IRemoteDisplay> display =
-        service->listenForRemoteDisplay(client, iface);
-
-    client->waitUntilDone();
-
-    display->dispose();
-    display.clear();
-}
-
-static void createFileSource(
-        const AString &addr, int32_t port, const char *path) {
-    sp<ANetworkSession> session = new ANetworkSession;
-    session->start();
-
-    sp<ALooper> looper = new ALooper;
-    looper->start();
-
-    sp<RemoteDisplayClient> client = new RemoteDisplayClient;
-    sp<WifiDisplaySource> source = new WifiDisplaySource(session, client, path);
-    looper->registerHandler(source);
-
-    AString iface = StringPrintf("%s:%d", addr.c_str(), port);
-    CHECK_EQ((status_t)OK, source->start(iface.c_str()));
-
-    client->waitUntilDone();
-
-    source->stop();
-}
-
-}  // namespace android
-
-int main(int argc, char **argv) {
-    using namespace android;
-
-    ProcessState::self()->startThreadPool();
-
-    DataSource::RegisterDefaultSniffers();
-
-    AString connectToHost;
-    int32_t connectToPort = -1;
-    AString uri;
-
-    AString listenOnAddr;
-    int32_t listenOnPort = -1;
-
-    AString path;
-
-    bool specialMode = false;
-
-    int res;
-    while ((res = getopt(argc, argv, "hc:l:u:f:s")) >= 0) {
-        switch (res) {
-            case 'c':
-            {
-                const char *colonPos = strrchr(optarg, ':');
-
-                if (colonPos == NULL) {
-                    connectToHost = optarg;
-                    connectToPort = WifiDisplaySource::kWifiDisplayDefaultPort;
-                } else {
-                    connectToHost.setTo(optarg, colonPos - optarg);
-
-                    char *end;
-                    connectToPort = strtol(colonPos + 1, &end, 10);
-
-                    if (*end != '\0' || end == colonPos + 1
-                            || connectToPort < 1 || connectToPort > 65535) {
-                        fprintf(stderr, "Illegal port specified.\n");
-                        exit(1);
-                    }
-                }
-                break;
-            }
-
-            case 'u':
-            {
-                uri = optarg;
-                break;
-            }
-
-            case 'f':
-            {
-                path = optarg;
-                break;
-            }
-
-            case 'l':
-            {
-                const char *colonPos = strrchr(optarg, ':');
-
-                if (colonPos == NULL) {
-                    listenOnAddr = optarg;
-                    listenOnPort = WifiDisplaySource::kWifiDisplayDefaultPort;
-                } else {
-                    listenOnAddr.setTo(optarg, colonPos - optarg);
-
-                    char *end;
-                    listenOnPort = strtol(colonPos + 1, &end, 10);
-
-                    if (*end != '\0' || end == colonPos + 1
-                            || listenOnPort < 1 || listenOnPort > 65535) {
-                        fprintf(stderr, "Illegal port specified.\n");
-                        exit(1);
-                    }
-                }
-                break;
-            }
-
-            case 's':
-            {
-                specialMode = true;
-                break;
-            }
-
-            case '?':
-            case 'h':
-            default:
-                usage(argv[0]);
-                exit(1);
-        }
-    }
-
-    if (connectToPort >= 0 && listenOnPort >= 0) {
-        fprintf(stderr,
-                "You can connect to a source or create one, "
-                "but not both at the same time.\n");
-        exit(1);
-    }
-
-    if (listenOnPort >= 0) {
-        if (path.empty()) {
-            createSource(listenOnAddr, listenOnPort);
-        } else {
-            createFileSource(listenOnAddr, listenOnPort, path.c_str());
-        }
-
-        exit(0);
-    }
-
-    if (connectToPort < 0 && uri.empty()) {
-        fprintf(stderr,
-                "You need to select either source host or uri.\n");
-
-        exit(1);
-    }
-
-    if (connectToPort >= 0 && !uri.empty()) {
-        fprintf(stderr,
-                "You need to either connect to a wfd host or an rtsp url, "
-                "not both.\n");
-        exit(1);
-    }
-
-    sp<SurfaceComposerClient> composerClient = new SurfaceComposerClient;
-    CHECK_EQ(composerClient->initCheck(), (status_t)OK);
-
-    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 %d x %d\n", displayWidth, displayHeight);
-
-    sp<SurfaceControl> control =
-        composerClient->createSurface(
-                String8("A Surface"),
-                displayWidth,
-                displayHeight,
-                PIXEL_FORMAT_RGB_565,
-                0);
-
-    CHECK(control != NULL);
-    CHECK(control->isValid());
-
-    SurfaceComposerClient::openGlobalTransaction();
-    CHECK_EQ(control->setLayer(INT_MAX), (status_t)OK);
-    CHECK_EQ(control->show(), (status_t)OK);
-    SurfaceComposerClient::closeGlobalTransaction();
-
-    sp<Surface> surface = control->getSurface();
-    CHECK(surface != NULL);
-
-    sp<ANetworkSession> session = new ANetworkSession;
-    session->start();
-
-    sp<ALooper> looper = new ALooper;
-
-    sp<WifiDisplaySink> sink = new WifiDisplaySink(
-            specialMode ? WifiDisplaySink::FLAG_SPECIAL_MODE : 0 /* flags */,
-            session,
-            surface->getIGraphicBufferProducer());
-
-    looper->registerHandler(sink);
-
-    if (connectToPort >= 0) {
-        sink->start(connectToHost.c_str(), connectToPort);
-    } else {
-        sink->start(uri.c_str());
-    }
-
-    looper->start(true /* runOnCallingThread */);
-
-    composerClient->dispose();
-
-    return 0;
-}

media/libstagefright/yuv/Android.mk

diff --git a/media/libstagefright/yuv/Android.mk b/media/libstagefright/yuv/Android.mk
index b3f7b1b..bb86dfc 100644
--- a/media/libstagefright/yuv/Android.mk
+++ b/media/libstagefright/yuv/Android.mk
@@ -12,5 +12,7 @@
 LOCAL_MODULE:= libstagefright_yuv
 
 
+LOCAL_CFLAGS += -Werror
+
 
 include $(BUILD_SHARED_LIBRARY)

media/libstagefright/yuv/YUVImage.cpp

diff --git a/media/libstagefright/yuv/YUVImage.cpp b/media/libstagefright/yuv/YUVImage.cpp
index 7b9000b..bb3e2fd 100644
--- a/media/libstagefright/yuv/YUVImage.cpp
+++ b/media/libstagefright/yuv/YUVImage.cpp
@@ -226,8 +226,8 @@
             &ySrcOffsetIncrement, &uSrcOffsetIncrement, &vSrcOffsetIncrement);
 
     int32_t yDestOffsetIncrement;
-    int32_t uDestOffsetIncrement;
-    int32_t vDestOffsetIncrement;
+    int32_t uDestOffsetIncrement = 0;
+    int32_t vDestOffsetIncrement = 0;
     destImage.getOffsetIncrementsPerDataRow(
             &yDestOffsetIncrement, &uDestOffsetIncrement, &vDestOffsetIncrement);
 
@@ -309,7 +309,7 @@
 
     int32_t yDestOffsetIncrement;
     int32_t uDestOffsetIncrement;
-    int32_t vDestOffsetIncrement;
+    int32_t vDestOffsetIncrement = 0;
     destImage.getOffsetIncrementsPerDataRow(
             &yDestOffsetIncrement, &uDestOffsetIncrement, &vDestOffsetIncrement);
 
@@ -393,9 +393,9 @@
     fprintf(fp, "255\n");
     for (int32_t y = 0; y < mHeight; ++y) {
         for (int32_t x = 0; x < mWidth; ++x) {
-            uint8_t yValue;
-            uint8_t uValue;
-            uint8_t vValue;
+            uint8_t yValue = 0u;
+            uint8_t uValue = 0u;
+            uint8_t vValue = 0u;
             getPixelValue(x, y, &yValue, &uValue, & vValue);
 
             uint8_t rValue;

media/mediaserver/Android.mk

diff --git a/media/mediaserver/Android.mk b/media/mediaserver/Android.mk
index 1ac647a..f848054 100644
--- a/media/mediaserver/Android.mk
+++ b/media/mediaserver/Android.mk
@@ -15,6 +15,7 @@
 
 LOCAL_SHARED_LIBRARIES := \
 	libaudioflinger \
+	libcamera_metadata\
 	libcameraservice \
 	libmedialogservice \
 	libcutils \

media/mediaserver/main_mediaserver.cpp

diff --git a/media/mediaserver/main_mediaserver.cpp b/media/mediaserver/main_mediaserver.cpp
index d5207d5..a347951 100644
--- a/media/mediaserver/main_mediaserver.cpp
+++ b/media/mediaserver/main_mediaserver.cpp
@@ -37,7 +37,7 @@
 
 using namespace android;
 
-int main(int argc, char** argv)
+int main(int argc __unused, char** argv)
 {
     signal(SIGPIPE, SIG_IGN);
     char value[PROPERTY_VALUE_MAX];

media/mtp/MtpDataPacket.cpp

diff --git a/media/mtp/MtpDataPacket.cpp b/media/mtp/MtpDataPacket.cpp
index 930f0b0..c4f87a0 100644
--- a/media/mtp/MtpDataPacket.cpp
+++ b/media/mtp/MtpDataPacket.cpp
@@ -331,7 +331,7 @@
 
 void MtpDataPacket::putString(const uint16_t* string) {
     int count = 0;
-    for (int i = 0; i < 256; i++) {
+    for (int i = 0; i <= MTP_STRING_MAX_CHARACTER_NUMBER; i++) {
         if (string[i])
             count++;
         else

media/mtp/MtpProperty.cpp

diff --git a/media/mtp/MtpProperty.cpp b/media/mtp/MtpProperty.cpp
index 64dd45b..375ed9a 100644
--- a/media/mtp/MtpProperty.cpp
+++ b/media/mtp/MtpProperty.cpp
@@ -16,6 +16,7 @@
 
 #define LOG_TAG "MtpProperty"
 
+#include <inttypes.h>
 #include "MtpDataPacket.h"
 #include "MtpDebug.h"
 #include "MtpProperty.h"
@@ -385,10 +386,10 @@
             buffer.appendFormat("%d", value.u.u32);
             break;
         case MTP_TYPE_INT64:
-            buffer.appendFormat("%lld", value.u.i64);
+            buffer.appendFormat("%" PRId64, value.u.i64);
             break;
         case MTP_TYPE_UINT64:
-            buffer.appendFormat("%lld", value.u.u64);
+            buffer.appendFormat("%" PRIu64, value.u.u64);
             break;
         case MTP_TYPE_INT128:
             buffer.appendFormat("%08X%08X%08X%08X", value.u.i128[0], value.u.i128[1],

media/mtp/MtpStringBuffer.cpp

diff --git a/media/mtp/MtpStringBuffer.cpp b/media/mtp/MtpStringBuffer.cpp
index fe8cf04..f3420a4 100644
--- a/media/mtp/MtpStringBuffer.cpp
+++ b/media/mtp/MtpStringBuffer.cpp
@@ -56,42 +56,47 @@
 }
 
 void MtpStringBuffer::set(const char* src) {
-    int length = strlen(src);
-    if (length >= sizeof(mBuffer))
-        length = sizeof(mBuffer) - 1;
-    memcpy(mBuffer, src, length);
-
     // count the characters
     int count = 0;
     char ch;
-    while ((ch = *src++) != 0) {
+    char* dest = (char*)mBuffer;
+
+    while ((ch = *src++) != 0 && count < MTP_STRING_MAX_CHARACTER_NUMBER) {
         if ((ch & 0x80) == 0) {
             // single byte character
+            *dest++ = ch;
         } else if ((ch & 0xE0) == 0xC0) {
             // two byte character
-            if (! *src++) {
+            char ch1 = *src++;
+            if (! ch1) {
                 // last character was truncated, so ignore last byte
-                length--;
                 break;
             }
+
+            *dest++ = ch;
+            *dest++ = ch1;
         } else if ((ch & 0xF0) == 0xE0) {
             // 3 byte char
-            if (! *src++) {
+            char ch1 = *src++;
+            if (! ch1) {
                 // last character was truncated, so ignore last byte
-                length--;
                 break;
             }
-            if (! *src++) {
-                // last character was truncated, so ignore last two bytes
-                length -= 2;
+            char ch2 = *src++;
+            if (! ch2) {
+                // last character was truncated, so ignore last byte
                 break;
             }
+
+            *dest++ = ch;
+            *dest++ = ch1;
+            *dest++ = ch2;
         }
         count++;
     }
 
-    mByteCount = length + 1;
-    mBuffer[length] = 0;
+    *dest++ = 0;
+    mByteCount = dest - (char*)mBuffer;
     mCharCount = count;
 }
 
@@ -100,7 +105,7 @@
     uint16_t ch;
     uint8_t* dest = mBuffer;
 
-    while ((ch = *src++) != 0 && count < 255) {
+    while ((ch = *src++) != 0 && count < MTP_STRING_MAX_CHARACTER_NUMBER) {
         if (ch >= 0x0800) {
             *dest++ = (uint8_t)(0xE0 | (ch >> 12));
             *dest++ = (uint8_t)(0x80 | ((ch >> 6) & 0x3F));

media/mtp/MtpStringBuffer.h

diff --git a/media/mtp/MtpStringBuffer.h b/media/mtp/MtpStringBuffer.h
index cbc8307..e5150df 100644
--- a/media/mtp/MtpStringBuffer.h
+++ b/media/mtp/MtpStringBuffer.h
@@ -19,6 +19,9 @@
 
 #include <stdint.h>
 
+// Max Character number of a MTP String
+#define MTP_STRING_MAX_CHARACTER_NUMBER             255
+
 namespace android {
 
 class MtpDataPacket;
@@ -29,7 +32,7 @@
 private:
     // mBuffer contains string in UTF8 format
     // maximum 3 bytes/character, with 1 extra for zero termination
-    uint8_t         mBuffer[255 * 3 + 1];
+    uint8_t         mBuffer[MTP_STRING_MAX_CHARACTER_NUMBER * 3 + 1];
     int             mCharCount;
     int             mByteCount;
 

services/audioflinger/Android.mk

diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index 54377f1..4524d3c 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -23,7 +23,8 @@
     AudioPolicyService.cpp      \
     ServiceUtilities.cpp        \
     AudioResamplerCubic.cpp.arm \
-    AudioResamplerSinc.cpp.arm
+    AudioResamplerSinc.cpp.arm  \
+    AudioResamplerDyn.cpp.arm
 
 LOCAL_SRC_FILES += StateQueue.cpp
 
@@ -74,12 +75,20 @@
 include $(CLEAR_VARS)
 
 LOCAL_SRC_FILES:=               \
-	test-resample.cpp 			\
+    test-resample.cpp           \
     AudioResampler.cpp.arm      \
-	AudioResamplerCubic.cpp.arm \
-    AudioResamplerSinc.cpp.arm
+    AudioResamplerCubic.cpp.arm \
+    AudioResamplerSinc.cpp.arm  \
+    AudioResamplerDyn.cpp.arm
+
+LOCAL_C_INCLUDES := \
+    $(call include-path-for, audio-utils)
+
+LOCAL_STATIC_LIBRARIES := \
+    libsndfile
 
 LOCAL_SHARED_LIBRARIES := \
+    libaudioutils \
     libdl \
     libcutils \
     libutils \

services/audioflinger/AudioFlinger.cpp

diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 04f0564..92ee30e 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -104,6 +104,27 @@
 
 // ----------------------------------------------------------------------------
 
+const char *formatToString(audio_format_t format) {
+    switch(format) {
+    case AUDIO_FORMAT_PCM_SUB_8_BIT: return "pcm8";
+    case AUDIO_FORMAT_PCM_SUB_16_BIT: return "pcm16";
+    case AUDIO_FORMAT_PCM_SUB_32_BIT: return "pcm32";
+    case AUDIO_FORMAT_PCM_SUB_8_24_BIT: return "pcm8.24";
+    case AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED: return "pcm24";
+    case AUDIO_FORMAT_PCM_SUB_FLOAT: return "pcmfloat";
+    case AUDIO_FORMAT_MP3: return "mp3";
+    case AUDIO_FORMAT_AMR_NB: return "amr-nb";
+    case AUDIO_FORMAT_AMR_WB: return "amr-wb";
+    case AUDIO_FORMAT_AAC: return "aac";
+    case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1";
+    case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2";
+    case AUDIO_FORMAT_VORBIS: return "vorbis";
+    default:
+        break;
+    }
+    return "unknown";
+}
+
 static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
 {
     const hw_module_t *mod;
@@ -162,6 +183,7 @@
         (void) property_get("af.tee", value, "0");
         teeEnabled = atoi(value);
     }
+    // FIXME symbolic constants here
     if (teeEnabled & 1) {
         mTeeSinkInputEnabled = true;
     }
@@ -213,6 +235,18 @@
         audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
         delete mAudioHwDevs.valueAt(i);
     }
+
+    // Tell media.log service about any old writers that still need to be unregistered
+    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
+    if (binder != 0) {
+        sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
+        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
+            sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
+            mUnregisteredWriters.pop();
+            mediaLogService->unregisterWriter(iMemory);
+        }
+    }
+
 }
 
 static const char * const audio_interfaces[] = {
@@ -252,7 +286,7 @@
     return NULL;
 }
 
-void AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
+void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused)
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
@@ -274,17 +308,17 @@
     }
 
     result.append("Global session refs:\n");
-    result.append(" session pid count\n");
+    result.append("  session   pid count\n");
     for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
         AudioSessionRef *r = mAudioSessionRefs[i];
-        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt);
+        snprintf(buffer, SIZE, "  %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt);
         result.append(buffer);
     }
     write(fd, result.string(), result.size());
 }
 
 
-void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
+void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused)
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
@@ -299,7 +333,7 @@
     write(fd, result.string(), result.size());
 }
 
-void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
+void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused)
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
@@ -406,16 +440,44 @@
 
 sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
 {
+    // If there is no memory allocated for logs, return a dummy writer that does nothing
     if (mLogMemoryDealer == 0) {
         return new NBLog::Writer();
     }
-    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
-    sp<NBLog::Writer> writer = new NBLog::Writer(size, shared);
     sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
-    if (binder != 0) {
-        interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name);
+    // Similarly if we can't contact the media.log service, also return a dummy writer
+    if (binder == 0) {
+        return new NBLog::Writer();
     }
-    return writer;
+    sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
+    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
+    // If allocation fails, consult the vector of previously unregistered writers
+    // and garbage-collect one or more them until an allocation succeeds
+    if (shared == 0) {
+        Mutex::Autolock _l(mUnregisteredWritersLock);
+        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
+            {
+                // Pick the oldest stale writer to garbage-collect
+                sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory());
+                mUnregisteredWriters.removeAt(0);
+                mediaLogService->unregisterWriter(iMemory);
+                // Now the media.log remote reference to IMemory is gone.  When our last local
+                // reference to IMemory also drops to zero at end of this block,
+                // the IMemory destructor will deallocate the region from mLogMemoryDealer.
+            }
+            // Re-attempt the allocation
+            shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
+            if (shared != 0) {
+                goto success;
+            }
+        }
+        // Even after garbage-collecting all old writers, there is still not enough memory,
+        // so return a dummy writer
+        return new NBLog::Writer();
+    }
+success:
+    mediaLogService->registerWriter(shared, size, name);
+    return new NBLog::Writer(size, shared);
 }
 
 void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
@@ -427,13 +489,10 @@
     if (iMemory == 0) {
         return;
     }
-    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
-    if (binder != 0) {
-        interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory);
-        // Now the media.log remote reference to IMemory is gone.
-        // When our last local reference to IMemory also drops to zero,
-        // the IMemory destructor will deallocate the region from mMemoryDealer.
-    }
+    // Rather than removing the writer immediately, append it to a queue of old writers to
+    // be garbage-collected later.  This allows us to continue to view old logs for a while.
+    Mutex::Autolock _l(mUnregisteredWritersLock);
+    mUnregisteredWriters.push(writer);
 }
 
 // IAudioFlinger interface
@@ -444,13 +503,14 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        size_t frameCount,
+        size_t *frameCount,
         IAudioFlinger::track_flags_t *flags,
         const sp<IMemory>& sharedBuffer,
         audio_io_handle_t output,
         pid_t tid,
         int *sessionId,
         String8& name,
+        int clientUid,
         status_t *status)
 {
     sp<PlaybackThread::Track> track;
@@ -467,10 +527,31 @@
         goto Exit;
     }
 
+    // further sample rate checks are performed by createTrack_l() depending on the thread type
+    if (sampleRate == 0) {
+        ALOGE("createTrack() invalid sample rate %u", sampleRate);
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+
+    // further channel mask checks are performed by createTrack_l() depending on the thread type
+    if (!audio_is_output_channel(channelMask)) {
+        ALOGE("createTrack() invalid channel mask %#x", channelMask);
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+
     // client is responsible for conversion of 8-bit PCM to 16-bit PCM,
     // and we don't yet support 8.24 or 32-bit PCM
-    if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) {
-        ALOGE("createTrack() invalid format %d", format);
+    if (!audio_is_valid_format(format) ||
+            (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT)) {
+        ALOGE("createTrack() invalid format %#x", format);
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+
+    if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) {
+        ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()");
         lStatus = BAD_VALUE;
         goto Exit;
     }
@@ -486,10 +567,11 @@
         }
 
         pid_t pid = IPCThreadState::self()->getCallingPid();
+
         client = registerPid_l(pid);
 
         ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
-        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
+        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
             // check if an effect chain with the same session ID is present on another
             // output thread and move it here.
             for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
@@ -513,7 +595,8 @@
         ALOGV("createTrack() lSessionId: %d", lSessionId);
 
         track = thread->createTrack_l(client, streamType, sampleRate, format,
-                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus);
+                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
+        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
         // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
 
         // move effect chain to this output thread if an effect on same session was waiting
@@ -1013,7 +1096,7 @@
     return size;
 }
 
-unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
+uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
 {
     Mutex::Autolock _l(mLock);
 
@@ -1045,7 +1128,7 @@
     return ret;
 }
 
-status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames,
+status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
         audio_io_handle_t output) const
 {
     status_t status;
@@ -1213,7 +1296,7 @@
 {
 }
 
-void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
+void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
 {
     sp<NotificationClient> keep(this);
     mAudioFlinger->removeNotificationClient(mPid);
@@ -1231,7 +1314,7 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        size_t frameCount,
+        size_t *frameCount,
         IAudioFlinger::track_flags_t *flags,
         pid_t tid,
         int *sessionId,
@@ -1247,12 +1330,29 @@
 
     // check calling permissions
     if (!recordingAllowed()) {
+        ALOGE("openRecord() permission denied: recording not allowed");
         lStatus = PERMISSION_DENIED;
         goto Exit;
     }
 
+    // further sample rate checks are performed by createRecordTrack_l()
+    if (sampleRate == 0) {
+        ALOGE("openRecord() invalid sample rate %u", sampleRate);
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+
+    // FIXME when we support more formats, add audio_is_valid_format(format)
+    //       and any explicit restrictions if audio_is_linear_pcm(format)
     if (format != AUDIO_FORMAT_PCM_16_BIT) {
-        ALOGE("openRecord() invalid format %d", format);
+        ALOGE("openRecord() invalid format %#x", format);
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+
+    // further channel mask checks are performed by createRecordTrack_l()
+    if (!audio_is_input_channel(channelMask)) {
+        ALOGE("openRecord() invalid channel mask %#x", channelMask);
         lStatus = BAD_VALUE;
         goto Exit;
     }
@@ -1262,12 +1362,14 @@
         Mutex::Autolock _l(mLock);
         thread = checkRecordThread_l(input);
         if (thread == NULL) {
+            ALOGE("openRecord() checkRecordThread_l failed");
             lStatus = BAD_VALUE;
             goto Exit;
         }
 
         if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice())
                 && !captureAudioOutputAllowed()) {
+            ALOGE("openRecord() permission denied: capture not allowed");
             lStatus = PERMISSION_DENIED;
             goto Exit;
         }
@@ -1276,7 +1378,7 @@
         client = registerPid_l(pid);
 
         // If no audio session id is provided, create one here
-        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
+        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
             lSessionId = *sessionId;
         } else {
             lSessionId = nextUniqueId();
@@ -1286,8 +1388,12 @@
         }
         // create new record track.
         // The record track uses one track in mHardwareMixerThread by convention.
+        // TODO: the uid should be passed in as a parameter to openRecord
         recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
-                                                  frameCount, lSessionId, flags, tid, &lStatus);
+                                                  frameCount, lSessionId,
+                                                  IPCThreadState::self()->getCallingUid(),
+                                                  flags, tid, &lStatus);
+        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
     }
 
     if (lStatus != NO_ERROR) {
@@ -1693,7 +1799,7 @@
     audio_stream_in_t *inStream = NULL;
     status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
                                         &inStream);
-    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, "
+    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %#x, Channels %x, "
             "status %d",
             inStream,
             config.sample_rate,
@@ -1708,9 +1814,11 @@
         reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
         (config.sample_rate <= 2 * reqSamplingRate) &&
         (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) {
+        // FIXME describe the change proposed by HAL (save old values so we can log them here)
         ALOGV("openInput() reopening with proposed sampling rate and channel mask");
         inStream = NULL;
         status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
+        // FIXME log this new status; HAL should not propose any further changes
     }
 
     if (status == NO_ERROR && inStream != NULL) {
@@ -1728,13 +1836,13 @@
                                         popcount(inStream->common.get_channels(&inStream->common)));
         if (!mTeeSinkInputEnabled) {
             kind = TEE_SINK_NO;
-        } else if (format == Format_Invalid) {
+        } else if (!Format_isValid(format)) {
             kind = TEE_SINK_NO;
         } else if (mRecordTeeSink == 0) {
             kind = TEE_SINK_NEW;
         } else if (mRecordTeeSink->getStrongCount() != 1) {
             kind = TEE_SINK_NO;
-        } else if (format == mRecordTeeSink->format()) {
+        } else if (Format_isEqual(format, mRecordTeeSink->format())) {
             kind = TEE_SINK_OLD;
         } else {
             kind = TEE_SINK_NEW;
@@ -1771,8 +1879,6 @@
         // pre processing modules
         RecordThread *thread = new RecordThread(this,
                                   input,
-                                  reqSamplingRate,
-                                  reqChannelMask,
                                   id,
                                   primaryOutputDevice_l(),
                                   *pDevices
@@ -1834,10 +1940,10 @@
     return NO_ERROR;
 }
 
-status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
+status_t AudioFlinger::invalidateStream(audio_stream_type_t stream)
 {
     Mutex::Autolock _l(mLock);
-    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
+    ALOGV("invalidateStream() stream %d", stream);
 
     for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
         PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
@@ -1853,18 +1959,21 @@
     return nextUniqueId();
 }
 
-void AudioFlinger::acquireAudioSessionId(int audioSession)
+void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid)
 {
     Mutex::Autolock _l(mLock);
     pid_t caller = IPCThreadState::self()->getCallingPid();
-    ALOGV("acquiring %d from %d", audioSession, caller);
+    ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
+    if (pid != -1 && (caller == getpid_cached)) {
+        caller = pid;
+    }
 
     // Ignore requests received from processes not known as notification client. The request
     // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
     // called from a different pid leaving a stale session reference.  Also we don't know how
     // to clear this reference if the client process dies.
     if (mNotificationClients.indexOfKey(caller) < 0) {
-        ALOGV("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
+        ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
         return;
     }
 
@@ -1881,11 +1990,14 @@
     ALOGV(" added new entry for %d", audioSession);
 }
 
-void AudioFlinger::releaseAudioSessionId(int audioSession)
+void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid)
 {
     Mutex::Autolock _l(mLock);
     pid_t caller = IPCThreadState::self()->getCallingPid();
-    ALOGV("releasing %d from %d", audioSession, caller);
+    ALOGV("releasing %d from %d for %d", audioSession, caller, pid);
+    if (pid != -1 && (caller == getpid_cached)) {
+        caller = pid;
+    }
     size_t num = mAudioSessionRefs.size();
     for (size_t i = 0; i< num; i++) {
         AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
@@ -2014,7 +2126,7 @@
                                     int triggerSession,
                                     int listenerSession,
                                     sync_event_callback_t callBack,
-                                    void *cookie)
+                                    wp<RefBase> cookie)
 {
     Mutex::Autolock _l(mLock);
 
@@ -2331,6 +2443,7 @@
                                         strategy,
                                         sessionId,
                                         effect->id());
+            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
         }
         effect = chain->getEffectFromId_l(0);
     }
@@ -2345,6 +2458,7 @@
                                             strategy,
                                             sessionId,
                                             removed[i]->id());
+                AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
             }
         }
     }

services/audioflinger/AudioFlinger.h

diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 7f2f9ec..c2b516b 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -60,8 +60,8 @@
 
 namespace android {
 
-class audio_track_cblk_t;
-class effect_param_cblk_t;
+struct audio_track_cblk_t;
+struct effect_param_cblk_t;
 class AudioMixer;
 class AudioBuffer;
 class AudioResampler;
@@ -102,13 +102,14 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                size_t frameCount,
+                                size_t *pFrameCount,
                                 IAudioFlinger::track_flags_t *flags,
                                 const sp<IMemory>& sharedBuffer,
                                 audio_io_handle_t output,
                                 pid_t tid,
                                 int *sessionId,
                                 String8& name,
+                                int clientUid,
                                 status_t *status /*non-NULL*/);
 
     virtual sp<IAudioRecord> openRecord(
@@ -116,7 +117,7 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                size_t frameCount,
+                                size_t *pFrameCount,
                                 IAudioFlinger::track_flags_t *flags,
                                 pid_t tid,
                                 int *sessionId,
@@ -181,20 +182,20 @@
 
     virtual status_t closeInput(audio_io_handle_t input);
 
-    virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output);
+    virtual status_t invalidateStream(audio_stream_type_t stream);
 
     virtual status_t setVoiceVolume(float volume);
 
-    virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
+    virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
                                        audio_io_handle_t output) const;
 
-    virtual     unsigned int  getInputFramesLost(audio_io_handle_t ioHandle) const;
+    virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const;
 
     virtual int newAudioSessionId();
 
-    virtual void acquireAudioSessionId(int audioSession);
+    virtual void acquireAudioSessionId(int audioSession, pid_t pid);
 
-    virtual void releaseAudioSessionId(int audioSession);
+    virtual void releaseAudioSessionId(int audioSession, pid_t pid);
 
     virtual status_t queryNumberEffects(uint32_t *numEffects) const;
 
@@ -234,8 +235,12 @@
     sp<NBLog::Writer>   newWriter_l(size_t size, const char *name);
     void                unregisterWriter(const sp<NBLog::Writer>& writer);
 private:
-    static const size_t kLogMemorySize = 10 * 1024;
+    static const size_t kLogMemorySize = 40 * 1024;
     sp<MemoryDealer>    mLogMemoryDealer;   // == 0 when NBLog is disabled
+    // When a log writer is unregistered, it is done lazily so that media.log can continue to see it
+    // for as long as possible.  The memory is only freed when it is needed for another log writer.
+    Vector< sp<NBLog::Writer> > mUnregisteredWriters;
+    Mutex               mUnregisteredWritersLock;
 public:
 
     class SyncEvent;
@@ -248,7 +253,7 @@
                   int triggerSession,
                   int listenerSession,
                   sync_event_callback_t callBack,
-                  void *cookie)
+                  wp<RefBase> cookie)
         : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession),
           mCallback(callBack), mCookie(cookie)
         {}
@@ -261,14 +266,14 @@
         AudioSystem::sync_event_t type() const { return mType; }
         int triggerSession() const { return mTriggerSession; }
         int listenerSession() const { return mListenerSession; }
-        void *cookie() const { return mCookie; }
+        wp<RefBase> cookie() const { return mCookie; }
 
     private:
           const AudioSystem::sync_event_t mType;
           const int mTriggerSession;
           const int mListenerSession;
           sync_event_callback_t mCallback;
-          void * const mCookie;
+          const wp<RefBase> mCookie;
           mutable Mutex mLock;
     };
 
@@ -276,7 +281,7 @@
                                         int triggerSession,
                                         int listenerSession,
                                         sync_event_callback_t callBack,
-                                        void *cookie);
+                                        wp<RefBase> cookie);
 
 private:
     class AudioHwDevice;    // fwd declaration for findSuitableHwDev_l
@@ -412,6 +417,7 @@
                                                   int target);
         virtual status_t    setParameters(const String8& keyValuePairs);
         virtual status_t    getTimestamp(AudioTimestamp& timestamp);
+        virtual void        signal(); // signal playback thread for a change in control block
 
         virtual status_t onTransact(
             uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
@@ -497,7 +503,7 @@
     private:
         const char * const mModuleName;
         audio_hw_device_t * const mHwDevice;
-        Flags mFlags;
+        const Flags mFlags;
     };
 
     // AudioStreamOut and AudioStreamIn are immutable, so their fields are const.
@@ -507,7 +513,7 @@
     struct AudioStreamOut {
         AudioHwDevice* const audioHwDev;
         audio_stream_out_t* const stream;
-        audio_output_flags_t flags;
+        const audio_output_flags_t flags;
 
         audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); }
 
@@ -632,7 +638,7 @@
     // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes
     static const size_t kTeeSinkInputFramesDefault = 0x200000;
     static const size_t kTeeSinkOutputFramesDefault = 0x200000;
-    static const size_t kTeeSinkTrackFramesDefault = 0x1000;
+    static const size_t kTeeSinkTrackFramesDefault = 0x200000;
 #endif
 
     // This method reads from a variable without mLock, but the variable is updated under mLock.  So
@@ -649,6 +655,8 @@
 
 #undef INCLUDING_FROM_AUDIOFLINGER_H
 
+const char *formatToString(audio_format_t format);
+
 // ----------------------------------------------------------------------------
 
 }; // namespace android

services/audioflinger/AudioMixer.cpp

diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 91aedbb..3ac5da9 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -58,7 +58,7 @@
 status_t AudioMixer::DownmixerBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer,
         int64_t pts) {
     //ALOGV("DownmixerBufferProvider::getNextBuffer()");
-    if (this->mTrackBufferProvider != NULL) {
+    if (mTrackBufferProvider != NULL) {
         status_t res = mTrackBufferProvider->getNextBuffer(pBuffer, pts);
         if (res == OK) {
             mDownmixConfig.inputCfg.buffer.frameCount = pBuffer->frameCount;
@@ -81,7 +81,7 @@
 
 void AudioMixer::DownmixerBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer) {
     //ALOGV("DownmixerBufferProvider::releaseBuffer()");
-    if (this->mTrackBufferProvider != NULL) {
+    if (mTrackBufferProvider != NULL) {
         mTrackBufferProvider->releaseBuffer(pBuffer);
     } else {
         ALOGE("DownmixerBufferProvider::releaseBuffer() error: NULL track buffer provider");
@@ -90,9 +90,9 @@
 
 
 // ----------------------------------------------------------------------------
-bool AudioMixer::isMultichannelCapable = false;
+bool AudioMixer::sIsMultichannelCapable = false;
 
-effect_descriptor_t AudioMixer::dwnmFxDesc;
+effect_descriptor_t AudioMixer::sDwnmFxDesc;
 
 // Ensure mConfiguredNames bitmask is initialized properly on all architectures.
 // The value of 1 << x is undefined in C when x >= 32.
@@ -113,8 +113,6 @@
     // AudioMixer is not yet capable of multi-channel output beyond stereo
     ALOG_ASSERT(2 == MAX_NUM_CHANNELS, "bad MAX_NUM_CHANNELS %d", MAX_NUM_CHANNELS);
 
-    LocalClock lc;
-
     pthread_once(&sOnceControl, &sInitRoutine);
 
     mState.enabledTracks= 0;
@@ -136,27 +134,6 @@
         t++;
     }
 
-    // find multichannel downmix effect if we have to play multichannel content
-    uint32_t numEffects = 0;
-    int ret = EffectQueryNumberEffects(&numEffects);
-    if (ret != 0) {
-        ALOGE("AudioMixer() error %d querying number of effects", ret);
-        return;
-    }
-    ALOGV("EffectQueryNumberEffects() numEffects=%d", numEffects);
-
-    for (uint32_t i = 0 ; i < numEffects ; i++) {
-        if (EffectQueryEffect(i, &dwnmFxDesc) == 0) {
-            ALOGV("effect %d is called %s", i, dwnmFxDesc.name);
-            if (memcmp(&dwnmFxDesc.type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) {
-                ALOGI("found effect \"%s\" from %s",
-                        dwnmFxDesc.name, dwnmFxDesc.implementor);
-                isMultichannelCapable = true;
-                break;
-            }
-        }
-    }
-    ALOGE_IF(!isMultichannelCapable, "unable to find downmix effect");
 }
 
 AudioMixer::~AudioMixer()
@@ -216,6 +193,7 @@
         t->mainBuffer = NULL;
         t->auxBuffer = NULL;
         t->downmixerBufferProvider = NULL;
+        t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
 
         status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask);
         if (status == OK) {
@@ -252,7 +230,7 @@
     return status;
 }
 
-void AudioMixer::unprepareTrackForDownmix(track_t* pTrack, int trackName) {
+void AudioMixer::unprepareTrackForDownmix(track_t* pTrack, int trackName __unused) {
     ALOGV("AudioMixer::unprepareTrackForDownmix(%d)", trackName);
 
     if (pTrack->downmixerBufferProvider != NULL) {
@@ -276,13 +254,13 @@
     DownmixerBufferProvider* pDbp = new DownmixerBufferProvider();
     int32_t status;
 
-    if (!isMultichannelCapable) {
+    if (!sIsMultichannelCapable) {
         ALOGE("prepareTrackForDownmix(%d) fails: mixer doesn't support multichannel content",
                 trackName);
         goto noDownmixForActiveTrack;
     }
 
-    if (EffectCreate(&dwnmFxDesc.uuid,
+    if (EffectCreate(&sDwnmFxDesc.uuid,
             pTrack->sessionId /*sessionId*/, -2 /*ioId not relevant here, using random value*/,
             &pDbp->mDownmixHandle/*pHandle*/) != 0) {
         ALOGE("prepareTrackForDownmix(%d) fails: error creating downmixer effect", trackName);
@@ -421,15 +399,16 @@
     ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
     track_t& track = mState.tracks[name];
 
-    int valueInt = (int)value;
-    int32_t *valueBuf = (int32_t *)value;
+    int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));
+    int32_t *valueBuf = reinterpret_cast<int32_t*>(value);
 
     switch (target) {
 
     case TRACK:
         switch (param) {
         case CHANNEL_MASK: {
-            audio_channel_mask_t mask = (audio_channel_mask_t) value;
+            audio_channel_mask_t mask =
+                static_cast<audio_channel_mask_t>(reinterpret_cast<uintptr_t>(value));
             if (track.channelMask != mask) {
                 uint32_t channelCount = popcount(mask);
                 ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) && channelCount);
@@ -462,6 +441,13 @@
         //         for a specific track? or per mixer?
         /* case DOWNMIX_TYPE:
             break          */
+        case MIXER_FORMAT: {
+            audio_format_t format = static_cast<audio_format_t>(valueInt);
+            if (track.mMixerFormat != format) {
+                track.mMixerFormat = format;
+                ALOGV("setParameter(TRACK, MIXER_FORMAT, %#x)", format);
+            }
+            } break;
         default:
             LOG_FATAL("bad param");
         }
@@ -559,14 +545,14 @@
                 // Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
                 if (!((value == 44100 && devSampleRate == 48000) ||
                       (value == 48000 && devSampleRate == 44100))) {
-                    quality = AudioResampler::LOW_QUALITY;
+                    quality = AudioResampler::DYN_LOW_QUALITY;
                 } else {
                     quality = AudioResampler::DEFAULT_QUALITY;
                 }
                 resampler = AudioResampler::create(
                         format,
                         // the resampler sees the number of channels after the downmixer, if any
-                        downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount,
+                        (int) (downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount),
                         devSampleRate, quality);
                 resampler->setLocalTimeFreq(sLocalTimeFreq);
             }
@@ -667,27 +653,29 @@
         countActiveTracks++;
         track_t& t = state->tracks[i];
         uint32_t n = 0;
+        // FIXME can overflow (mask is only 3 bits)
         n |= NEEDS_CHANNEL_1 + t.channelCount - 1;
-        n |= NEEDS_FORMAT_16;
-        n |= t.doesResample() ? NEEDS_RESAMPLE_ENABLED : NEEDS_RESAMPLE_DISABLED;
+        if (t.doesResample()) {
+            n |= NEEDS_RESAMPLE;
+        }
         if (t.auxLevel != 0 && t.auxBuffer != NULL) {
-            n |= NEEDS_AUX_ENABLED;
+            n |= NEEDS_AUX;
         }
 
         if (t.volumeInc[0]|t.volumeInc[1]) {
             volumeRamp = true;
         } else if (!t.doesResample() && t.volumeRL == 0) {
-            n |= NEEDS_MUTE_ENABLED;
+            n |= NEEDS_MUTE;
         }
         t.needs = n;
 
-        if ((n & NEEDS_MUTE__MASK) == NEEDS_MUTE_ENABLED) {
+        if (n & NEEDS_MUTE) {
             t.hook = track__nop;
         } else {
-            if ((n & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
+            if (n & NEEDS_AUX) {
                 all16BitsStereoNoResample = false;
             }
-            if ((n & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
+            if (n & NEEDS_RESAMPLE) {
                 all16BitsStereoNoResample = false;
                 resampling = true;
                 t.hook = track__genericResample;
@@ -753,7 +741,7 @@
             en &= ~(1<<i);
             track_t& t = state->tracks[i];
             if (!t.doesResample() && t.volumeRL == 0) {
-                t.needs |= NEEDS_MUTE_ENABLED;
+                t.needs |= NEEDS_MUTE;
                 t.hook = track__nop;
             } else {
                 allMuted = false;
@@ -804,8 +792,8 @@
     }
 }
 
-void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp,
-        int32_t* aux)
+void AudioMixer::track__nop(track_t* t __unused, int32_t* out __unused,
+        size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused)
 {
 }
 
@@ -881,8 +869,8 @@
     }
 }
 
-void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
-        int32_t* aux)
+void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount,
+        int32_t* temp __unused, int32_t* aux)
 {
     const int16_t *in = static_cast<const int16_t *>(t->in);
 
@@ -972,8 +960,8 @@
     t->in = in;
 }
 
-void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
-        int32_t* aux)
+void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount,
+        int32_t* temp __unused, int32_t* aux)
 {
     const int16_t *in = static_cast<int16_t const *>(t->in);
 
@@ -1063,7 +1051,7 @@
 void AudioMixer::process__nop(state_t* state, int64_t pts)
 {
     uint32_t e0 = state->enabledTracks;
-    size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS;
+    size_t sampleCount = state->frameCount * MAX_NUM_CHANNELS;
     while (e0) {
         // process by group of tracks with same output buffer to
         // avoid multiple memset() on same buffer
@@ -1082,7 +1070,8 @@
             }
             e0 &= ~(e1);
 
-            memset(t1.mainBuffer, 0, bufSize);
+            memset(t1.mainBuffer, 0, sampleCount
+                    * audio_bytes_per_sample(t1.mMixerFormat));
         }
 
         while (e1) {
@@ -1121,11 +1110,6 @@
         t.bufferProvider->getNextBuffer(&t.buffer, pts);
         t.frameCount = t.buffer.frameCount;
         t.in = t.buffer.raw;
-        // t.in == NULL can happen if the track was flushed just after having
-        // been enabled for mixing.
-        if (t.in == NULL) {
-            enabledTracks &= ~(1<<i);
-        }
     }
 
     e0 = enabledTracks;
@@ -1157,10 +1141,17 @@
                 track_t& t = state->tracks[i];
                 size_t outFrames = BLOCKSIZE;
                 int32_t *aux = NULL;
-                if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) {
+                if (CC_UNLIKELY(t.needs & NEEDS_AUX)) {
                     aux = t.auxBuffer + numFrames;
                 }
                 while (outFrames) {
+                    // t.in == NULL can happen if the track was flushed just after having
+                    // been enabled for mixing.
+                   if (t.in == NULL) {
+                        enabledTracks &= ~(1<<i);
+                        e1 &= ~(1<<i);
+                        break;
+                    }
                     size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
                     if (inFrames > 0) {
                         t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames,
@@ -1188,8 +1179,18 @@
                     }
                 }
             }
-            ditherAndClamp(out, outTemp, BLOCKSIZE);
-            out += BLOCKSIZE;
+            switch (t1.mMixerFormat) {
+            case AUDIO_FORMAT_PCM_FLOAT:
+                memcpy_to_float_from_q19_12(reinterpret_cast<float *>(out), outTemp, BLOCKSIZE * 2);
+                out += BLOCKSIZE * 2; // output is 2 floats/frame.
+                break;
+            case AUDIO_FORMAT_PCM_16_BIT:
+                ditherAndClamp(out, outTemp, BLOCKSIZE);
+                out += BLOCKSIZE; // output is 1 int32_t (2 int16_t samples)/frame
+                break;
+            default:
+                LOG_ALWAYS_FATAL("bad mixer format: %d", t1.mMixerFormat);
+            }
             numFrames += BLOCKSIZE;
         } while (numFrames < state->frameCount);
     }
@@ -1238,14 +1239,14 @@
             e1 &= ~(1<<i);
             track_t& t = state->tracks[i];
             int32_t *aux = NULL;
-            if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) {
+            if (CC_UNLIKELY(t.needs & NEEDS_AUX)) {
                 aux = t.auxBuffer;
             }
 
             // this is a little goofy, on the resampling case we don't
             // acquire/release the buffers because it's done by
             // the resampler.
-            if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
+            if (t.needs & NEEDS_RESAMPLE) {
                 t.resampler->setPTS(pts);
                 t.hook(&t, outTemp, numFrames, state->resampleTemp, aux);
             } else {
@@ -1271,7 +1272,16 @@
                 }
             }
         }
-        ditherAndClamp(out, outTemp, numFrames);
+        switch (t1.mMixerFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            memcpy_to_float_from_q19_12(reinterpret_cast<float*>(out), outTemp, numFrames*2);
+            break;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            ditherAndClamp(out, outTemp, numFrames);
+            break;
+        default:
+            LOG_ALWAYS_FATAL("bad mixer format: %d", t1.mMixerFormat);
+        }
     }
 }
 
@@ -1312,27 +1322,46 @@
         }
         size_t outFrames = b.frameCount;
 
-        if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
-            // volume is boosted, so we might need to clamp even though
-            // we process only one track.
+        switch (t.mMixerFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT: {
+            float *fout = reinterpret_cast<float*>(out);
             do {
                 uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
                 in += 2;
-                int32_t l = mulRL(1, rl, vrl) >> 12;
-                int32_t r = mulRL(0, rl, vrl) >> 12;
-                // clamping...
-                l = clamp16(l);
-                r = clamp16(r);
-                *out++ = (r<<16) | (l & 0xFFFF);
+                int32_t l = mulRL(1, rl, vrl);
+                int32_t r = mulRL(0, rl, vrl);
+                *fout++ = float_from_q19_12(l);
+                *fout++ = float_from_q19_12(r);
+                // Note: In case of later int16_t sink output,
+                // conversion and clamping is done by memcpy_to_i16_from_float().
             } while (--outFrames);
-        } else {
-            do {
-                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
-                in += 2;
-                int32_t l = mulRL(1, rl, vrl) >> 12;
-                int32_t r = mulRL(0, rl, vrl) >> 12;
-                *out++ = (r<<16) | (l & 0xFFFF);
-            } while (--outFrames);
+            } break;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
+                // volume is boosted, so we might need to clamp even though
+                // we process only one track.
+                do {
+                    uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+                    in += 2;
+                    int32_t l = mulRL(1, rl, vrl) >> 12;
+                    int32_t r = mulRL(0, rl, vrl) >> 12;
+                    // clamping...
+                    l = clamp16(l);
+                    r = clamp16(r);
+                    *out++ = (r<<16) | (l & 0xFFFF);
+                } while (--outFrames);
+            } else {
+                do {
+                    uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+                    in += 2;
+                    int32_t l = mulRL(1, rl, vrl) >> 12;
+                    int32_t r = mulRL(0, rl, vrl) >> 12;
+                    *out++ = (r<<16) | (l & 0xFFFF);
+                } while (--outFrames);
+            }
+            break;
+        default:
+            LOG_ALWAYS_FATAL("bad mixer format: %d", t.mMixerFormat);
         }
         numFrames -= b.frameCount;
         t.bufferProvider->releaseBuffer(&b);
@@ -1459,6 +1488,28 @@
 {
     LocalClock lc;
     sLocalTimeFreq = lc.getLocalFreq();
+
+    // find multichannel downmix effect if we have to play multichannel content
+    uint32_t numEffects = 0;
+    int ret = EffectQueryNumberEffects(&numEffects);
+    if (ret != 0) {
+        ALOGE("AudioMixer() error %d querying number of effects", ret);
+        return;
+    }
+    ALOGV("EffectQueryNumberEffects() numEffects=%d", numEffects);
+
+    for (uint32_t i = 0 ; i < numEffects ; i++) {
+        if (EffectQueryEffect(i, &sDwnmFxDesc) == 0) {
+            ALOGV("effect %d is called %s", i, sDwnmFxDesc.name);
+            if (memcmp(&sDwnmFxDesc.type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) {
+                ALOGI("found effect \"%s\" from %s",
+                        sDwnmFxDesc.name, sDwnmFxDesc.implementor);
+                sIsMultichannelCapable = true;
+                break;
+            }
+        }
+    }
+    ALOGW_IF(!sIsMultichannelCapable, "unable to find downmix effect");
 }
 
 // ----------------------------------------------------------------------------

services/audioflinger/AudioMixer.h

diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 43aeb86..e5e120c 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -77,6 +77,7 @@
         MAIN_BUFFER     = 0x4002,
         AUX_BUFFER      = 0x4003,
         DOWNMIX_TYPE    = 0X4004,
+        MIXER_FORMAT    = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
         // for target RESAMPLE
         SAMPLE_RATE     = 0x4100, // Configure sample rate conversion on this track name;
                                   // parameter 'value' is the new sample rate in Hz.
@@ -120,27 +121,19 @@
 private:
 
     enum {
+        // FIXME this representation permits up to 8 channels
         NEEDS_CHANNEL_COUNT__MASK   = 0x00000007,
-        NEEDS_FORMAT__MASK          = 0x000000F0,
-        NEEDS_MUTE__MASK            = 0x00000100,
-        NEEDS_RESAMPLE__MASK        = 0x00001000,
-        NEEDS_AUX__MASK             = 0x00010000,
     };
 
     enum {
-        NEEDS_CHANNEL_1             = 0x00000000,
-        NEEDS_CHANNEL_2             = 0x00000001,
+        NEEDS_CHANNEL_1             = 0x00000000,   // mono
+        NEEDS_CHANNEL_2             = 0x00000001,   // stereo
 
-        NEEDS_FORMAT_16             = 0x00000010,
+        // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
 
-        NEEDS_MUTE_DISABLED         = 0x00000000,
-        NEEDS_MUTE_ENABLED          = 0x00000100,
-
-        NEEDS_RESAMPLE_DISABLED     = 0x00000000,
-        NEEDS_RESAMPLE_ENABLED      = 0x00001000,
-
-        NEEDS_AUX_DISABLED     = 0x00000000,
-        NEEDS_AUX_ENABLED      = 0x00010000,
+        NEEDS_MUTE                  = 0x00000100,
+        NEEDS_RESAMPLE              = 0x00001000,
+        NEEDS_AUX                   = 0x00010000,
     };
 
     struct state_t;
@@ -201,7 +194,9 @@
 
         int32_t     sessionId;
 
-        int32_t     padding[2];
+        audio_format_t mMixerFormat; // at this time: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+
+        int32_t     padding[1];
 
         // 16-byte boundary
 
@@ -224,7 +219,7 @@
         NBLog::Writer*  mLog;
         int32_t         reserved[1];
         // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
-        track_t         tracks[MAX_NUM_TRACKS]; __attribute__((aligned(32)));
+        track_t         tracks[MAX_NUM_TRACKS] __attribute__((aligned(32)));
     };
 
     // AudioBufferProvider that wraps a track AudioBufferProvider by a call to a downmix effect
@@ -256,9 +251,9 @@
     state_t         mState __attribute__((aligned(32)));
 
     // effect descriptor for the downmixer used by the mixer
-    static effect_descriptor_t dwnmFxDesc;
+    static effect_descriptor_t sDwnmFxDesc;
     // indicates whether a downmix effect has been found and is usable by this mixer
-    static bool                isMultichannelCapable;
+    static bool                sIsMultichannelCapable;
 
     // Call after changing either the enabled status of a track, or parameters of an enabled track.
     // OK to call more often than that, but unnecessary.

services/audioflinger/AudioPolicyService.cpp

diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index c5ad2c0..41bd990 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -60,7 +60,7 @@
 // ----------------------------------------------------------------------------
 
 AudioPolicyService::AudioPolicyService()
-    : BnAudioPolicyService() , mpAudioPolicyDev(NULL) , mpAudioPolicy(NULL)
+    : BnAudioPolicyService(), mpAudioPolicyDev(NULL), mpAudioPolicy(NULL)
 {
     char value[PROPERTY_VALUE_MAX];
     const struct hw_module_t *module;
@@ -475,8 +475,9 @@
 
 audio_io_handle_t AudioPolicyService::getOutputForEffect(const effect_descriptor_t *desc)
 {
+    // FIXME change return type to status_t, and return NO_INIT here
     if (mpAudioPolicy == NULL) {
-        return NO_INIT;
+        return 0;
     }
     Mutex::Autolock _l(mLock);
     return mpAudioPolicy->get_output_for_effect(mpAudioPolicy, desc);
@@ -612,7 +613,7 @@
     return NO_ERROR;
 }
 
-status_t AudioPolicyService::dump(int fd, const Vector<String16>& args)
+status_t AudioPolicyService::dump(int fd, const Vector<String16>& args __unused)
 {
     if (!dumpAllowed()) {
         dumpPermissionDenial(fd);
@@ -1441,6 +1442,14 @@
     loadEffects(root, effects);
     loadInputSources(root, effects);
 
+    // delete effects to fix memory leak.
+    // as effects is local var and valgrind would treat this as memory leak
+    // and although it only did in mediaserver init, but free it in case mediaserver reboot
+    size_t i;
+    for (i = 0; i < effects.size(); i++) {
+      delete effects[i];
+    }
+
     config_free(root);
     free(root);
     free(data);
@@ -1452,7 +1461,7 @@
 extern "C" {
 
 
-static audio_module_handle_t aps_load_hw_module(void *service,
+static audio_module_handle_t aps_load_hw_module(void *service __unused,
                                              const char *name)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
@@ -1465,7 +1474,7 @@
 }
 
 // deprecated: replaced by aps_open_output_on_module()
-static audio_io_handle_t aps_open_output(void *service,
+static audio_io_handle_t aps_open_output(void *service __unused,
                                          audio_devices_t *pDevices,
                                          uint32_t *pSamplingRate,
                                          audio_format_t *pFormat,
@@ -1483,7 +1492,7 @@
                           pLatencyMs, flags);
 }
 
-static audio_io_handle_t aps_open_output_on_module(void *service,
+static audio_io_handle_t aps_open_output_on_module(void *service __unused,
                                                    audio_module_handle_t module,
                                                    audio_devices_t *pDevices,
                                                    uint32_t *pSamplingRate,
@@ -1502,7 +1511,7 @@
                           pLatencyMs, flags, offloadInfo);
 }
 
-static audio_io_handle_t aps_open_dup_output(void *service,
+static audio_io_handle_t aps_open_dup_output(void *service __unused,
                                                  audio_io_handle_t output1,
                                                  audio_io_handle_t output2)
 {
@@ -1514,7 +1523,7 @@
     return af->openDuplicateOutput(output1, output2);
 }
 
-static int aps_close_output(void *service, audio_io_handle_t output)
+static int aps_close_output(void *service __unused, audio_io_handle_t output)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
     if (af == 0) {
@@ -1524,7 +1533,7 @@
     return af->closeOutput(output);
 }
 
-static int aps_suspend_output(void *service, audio_io_handle_t output)
+static int aps_suspend_output(void *service __unused, audio_io_handle_t output)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
     if (af == 0) {
@@ -1535,7 +1544,7 @@
     return af->suspendOutput(output);
 }
 
-static int aps_restore_output(void *service, audio_io_handle_t output)
+static int aps_restore_output(void *service __unused, audio_io_handle_t output)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
     if (af == 0) {
@@ -1547,12 +1556,12 @@
 }
 
 // deprecated: replaced by aps_open_input_on_module(), and acoustics parameter is ignored
-static audio_io_handle_t aps_open_input(void *service,
+static audio_io_handle_t aps_open_input(void *service __unused,
                                         audio_devices_t *pDevices,
                                         uint32_t *pSamplingRate,
                                         audio_format_t *pFormat,
                                         audio_channel_mask_t *pChannelMask,
-                                        audio_in_acoustics_t acoustics)
+                                        audio_in_acoustics_t acoustics __unused)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
     if (af == 0) {
@@ -1563,7 +1572,7 @@
     return af->openInput((audio_module_handle_t)0, pDevices, pSamplingRate, pFormat, pChannelMask);
 }
 
-static audio_io_handle_t aps_open_input_on_module(void *service,
+static audio_io_handle_t aps_open_input_on_module(void *service __unused,
                                                   audio_module_handle_t module,
                                                   audio_devices_t *pDevices,
                                                   uint32_t *pSamplingRate,
@@ -1579,7 +1588,7 @@
     return af->openInput(module, pDevices, pSamplingRate, pFormat, pChannelMask);
 }
 
-static int aps_close_input(void *service, audio_io_handle_t input)
+static int aps_close_input(void *service __unused, audio_io_handle_t input)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
     if (af == 0) {
@@ -1589,18 +1598,17 @@
     return af->closeInput(input);
 }
 
-static int aps_set_stream_output(void *service, audio_stream_type_t stream,
-                                     audio_io_handle_t output)
+static int aps_invalidate_stream(void *service __unused, audio_stream_type_t stream)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
     if (af == 0) {
         return PERMISSION_DENIED;
     }
 
-    return af->setStreamOutput(stream, output);
+    return af->invalidateStream(stream);
 }
 
-static int aps_move_effects(void *service, int session,
+static int aps_move_effects(void *service __unused, int session,
                                 audio_io_handle_t src_output,
                                 audio_io_handle_t dst_output)
 {
@@ -1612,7 +1620,7 @@
     return af->moveEffects(session, src_output, dst_output);
 }
 
-static char * aps_get_parameters(void *service, audio_io_handle_t io_handle,
+static char * aps_get_parameters(void *service __unused, audio_io_handle_t io_handle,
                                      const char *keys)
 {
     String8 result = AudioSystem::getParameters(io_handle, String8(keys));
@@ -1663,24 +1671,24 @@
 
 namespace {
     struct audio_policy_service_ops aps_ops = {
-        open_output           : aps_open_output,
-        open_duplicate_output : aps_open_dup_output,
-        close_output          : aps_close_output,
-        suspend_output        : aps_suspend_output,
-        restore_output        : aps_restore_output,
-        open_input            : aps_open_input,
-        close_input           : aps_close_input,
-        set_stream_volume     : aps_set_stream_volume,
-        set_stream_output     : aps_set_stream_output,
-        set_parameters        : aps_set_parameters,
-        get_parameters        : aps_get_parameters,
-        start_tone            : aps_start_tone,
-        stop_tone             : aps_stop_tone,
-        set_voice_volume      : aps_set_voice_volume,
-        move_effects          : aps_move_effects,
-        load_hw_module        : aps_load_hw_module,
-        open_output_on_module : aps_open_output_on_module,
-        open_input_on_module  : aps_open_input_on_module,
+        .open_output           = aps_open_output,
+        .open_duplicate_output = aps_open_dup_output,
+        .close_output          = aps_close_output,
+        .suspend_output        = aps_suspend_output,
+        .restore_output        = aps_restore_output,
+        .open_input            = aps_open_input,
+        .close_input           = aps_close_input,
+        .set_stream_volume     = aps_set_stream_volume,
+        .invalidate_stream     = aps_invalidate_stream,
+        .set_parameters        = aps_set_parameters,
+        .get_parameters        = aps_get_parameters,
+        .start_tone            = aps_start_tone,
+        .stop_tone             = aps_stop_tone,
+        .set_voice_volume      = aps_set_voice_volume,
+        .move_effects          = aps_move_effects,
+        .load_hw_module        = aps_load_hw_module,
+        .open_output_on_module = aps_open_output_on_module,
+        .open_input_on_module  = aps_open_input_on_module,
     };
 }; // namespace <unnamed>
 

services/audioflinger/AudioResampler.cpp

diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 323f1a4..ca98f16 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -25,6 +25,7 @@
 #include "AudioResampler.h"
 #include "AudioResamplerSinc.h"
 #include "AudioResamplerCubic.h"
+#include "AudioResamplerDyn.h"
 
 #ifdef __arm__
 #include <machine/cpu-features.h>
@@ -77,6 +78,9 @@
     int mX0R;
 };
 
+/*static*/
+const double AudioResampler::kPhaseMultiplier = 1L << AudioResampler::kNumPhaseBits;
+
 bool AudioResampler::qualityIsSupported(src_quality quality)
 {
     switch (quality) {
@@ -85,6 +89,9 @@
     case MED_QUALITY:
     case HIGH_QUALITY:
     case VERY_HIGH_QUALITY:
+    case DYN_LOW_QUALITY:
+    case DYN_MED_QUALITY:
+    case DYN_HIGH_QUALITY:
         return true;
     default:
         return false;
@@ -105,7 +112,7 @@
         if (*endptr == '\0') {
             defaultQuality = (src_quality) l;
             ALOGD("forcing AudioResampler quality to %d", defaultQuality);
-            if (defaultQuality < DEFAULT_QUALITY || defaultQuality > VERY_HIGH_QUALITY) {
+            if (defaultQuality < DEFAULT_QUALITY || defaultQuality > DYN_HIGH_QUALITY) {
                 defaultQuality = DEFAULT_QUALITY;
             }
         }
@@ -125,6 +132,12 @@
         return 20;
     case VERY_HIGH_QUALITY:
         return 34;
+    case DYN_LOW_QUALITY:
+        return 4;
+    case DYN_MED_QUALITY:
+        return 6;
+    case DYN_HIGH_QUALITY:
+        return 12;
     }
 }
 
@@ -148,6 +161,16 @@
         atFinalQuality = true;
     }
 
+    /* if the caller requests DEFAULT_QUALITY and af.resampler.property
+     * has not been set, the target resampler quality is set to DYN_MED_QUALITY,
+     * and allowed to "throttle" down to DYN_LOW_QUALITY if necessary
+     * due to estimated CPU load of having too many active resamplers
+     * (the code below the if).
+     */
+    if (quality == DEFAULT_QUALITY) {
+        quality = DYN_MED_QUALITY;
+    }
+
     // naive implementation of CPU load throttling doesn't account for whether resampler is active
     pthread_mutex_lock(&mutex);
     for (;;) {
@@ -162,7 +185,6 @@
         // not enough CPU available for proposed quality level, so try next lowest level
         switch (quality) {
         default:
-        case DEFAULT_QUALITY:
         case LOW_QUALITY:
             atFinalQuality = true;
             break;
@@ -175,6 +197,15 @@
         case VERY_HIGH_QUALITY:
             quality = HIGH_QUALITY;
             break;
+        case DYN_LOW_QUALITY:
+            atFinalQuality = true;
+            break;
+        case DYN_MED_QUALITY:
+            quality = DYN_LOW_QUALITY;
+            break;
+        case DYN_HIGH_QUALITY:
+            quality = DYN_MED_QUALITY;
+            break;
         }
     }
     pthread_mutex_unlock(&mutex);
@@ -183,7 +214,6 @@
 
     switch (quality) {
     default:
-    case DEFAULT_QUALITY:
     case LOW_QUALITY:
         ALOGV("Create linear Resampler");
         resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
@@ -200,6 +230,12 @@
         ALOGV("Create VERY_HIGH_QUALITY sinc Resampler = %d", quality);
         resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate, quality);
         break;
+    case DYN_LOW_QUALITY:
+    case DYN_MED_QUALITY:
+    case DYN_HIGH_QUALITY:
+        ALOGV("Create dynamic Resampler = %d", quality);
+        resampler = new AudioResamplerDyn(bitDepth, inChannelCount, sampleRate, quality);
+        break;
     }
 
     // initialize resampler
@@ -305,7 +341,7 @@
     uint32_t phaseIncrement = mPhaseIncrement;
     size_t outputIndex = 0;
     size_t outputSampleCount = outFrameCount * 2;
-    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
+    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
 
     // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
     //      outFrameCount, inputIndex, phaseFraction, phaseIncrement);
@@ -403,7 +439,7 @@
     uint32_t phaseIncrement = mPhaseIncrement;
     size_t outputIndex = 0;
     size_t outputSampleCount = outFrameCount * 2;
-    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
+    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
 
     // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
     //      outFrameCount, inputIndex, phaseFraction, phaseIncrement);
@@ -516,6 +552,16 @@
             size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
             uint32_t &phaseFraction, uint32_t phaseIncrement)
 {
+    (void)maxOutPt; // remove unused parameter warnings
+    (void)maxInIdx;
+    (void)outputIndex;
+    (void)out;
+    (void)inputIndex;
+    (void)vl;
+    (void)vr;
+    (void)phaseFraction;
+    (void)phaseIncrement;
+    (void)in;
 #define MO_PARAM5   "36"        // offset of parameter 5 (outputIndex)
 
     asm(
@@ -528,7 +574,7 @@
         "   ldr r8, [sp, #" MO_PARAM5 " + 4]\n"     // out
         "   ldr r0, [sp, #" MO_PARAM5 " + 0]\n"     // &outputIndex
         "   ldr r0, [r0]\n"                         // outputIndex
-        "   add r8, r0, asl #2\n"                   // curOut
+        "   add r8, r8, r0, asl #2\n"               // curOut
         "   ldr r9, [sp, #" MO_PARAM5 " + 24]\n"    // phaseIncrement
         "   ldr r10, [sp, #" MO_PARAM5 " + 12]\n"   // vl
         "   ldr r11, [sp, #" MO_PARAM5 " + 16]\n"   // vr
@@ -627,6 +673,16 @@
             size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
             uint32_t &phaseFraction, uint32_t phaseIncrement)
 {
+    (void)maxOutPt; // remove unused parameter warnings
+    (void)maxInIdx;
+    (void)outputIndex;
+    (void)out;
+    (void)inputIndex;
+    (void)vl;
+    (void)vr;
+    (void)phaseFraction;
+    (void)phaseIncrement;
+    (void)in;
 #define ST_PARAM5    "40"     // offset of parameter 5 (outputIndex)
     asm(
         "stmfd  sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, lr}\n"
@@ -638,7 +694,7 @@
         "   ldr r8, [sp, #" ST_PARAM5 " + 4]\n"     // out
         "   ldr r0, [sp, #" ST_PARAM5 " + 0]\n"     // &outputIndex
         "   ldr r0, [r0]\n"                         // outputIndex
-        "   add r8, r0, asl #2\n"                   // curOut
+        "   add r8, r8, r0, asl #2\n"               // curOut
         "   ldr r9, [sp, #" ST_PARAM5 " + 24]\n"    // phaseIncrement
         "   ldr r10, [sp, #" ST_PARAM5 " + 12]\n"   // vl
         "   ldr r11, [sp, #" ST_PARAM5 " + 16]\n"   // vr

services/audioflinger/AudioResampler.h

diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index 33e64ce..0592855 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -41,6 +41,9 @@
         MED_QUALITY=2,
         HIGH_QUALITY=3,
         VERY_HIGH_QUALITY=4,
+        DYN_LOW_QUALITY=5,
+        DYN_MED_QUALITY=6,
+        DYN_HIGH_QUALITY=7,
     };
 
     static AudioResampler* create(int bitDepth, int inChannelCount,
@@ -81,7 +84,7 @@
     static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1;
 
     // multiplier to calculate fixed point phase increment
-    static const double kPhaseMultiplier = 1L << kNumPhaseBits;
+    static const double kPhaseMultiplier;
 
     AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate, src_quality quality);
 
@@ -107,6 +110,38 @@
     uint64_t mLocalTimeFreq;
     int64_t mPTS;
 
+    // returns the inFrameCount required to generate outFrameCount frames.
+    //
+    // Placed here to be a consistent for all resamplers.
+    //
+    // Right now, we use the upper bound without regards to the current state of the
+    // input buffer using integer arithmetic, as follows:
+    //
+    // (static_cast<uint64_t>(outFrameCount)*mInSampleRate + (mSampleRate - 1))/mSampleRate;
+    //
+    // The double precision equivalent (float may not be precise enough):
+    // ceil(static_cast<double>(outFrameCount) * mInSampleRate / mSampleRate);
+    //
+    // this relies on the fact that the mPhaseIncrement is rounded down from
+    // #phases * mInSampleRate/mSampleRate and the fact that Sum(Floor(x)) <= Floor(Sum(x)).
+    // http://www.proofwiki.org/wiki/Sum_of_Floors_Not_Greater_Than_Floor_of_Sums
+    //
+    // (so long as double precision is computed accurately enough to be considered
+    // greater than or equal to the Floor(x) value in int32_t arithmetic; thus this
+    // will not necessarily hold for floats).
+    //
+    // TODO:
+    // Greater accuracy and a tight bound is obtained by:
+    // 1) subtract and adjust for the current state of the AudioBufferProvider buffer.
+    // 2) using the exact integer formula where (ignoring 64b casting)
+    //  inFrameCount = (mPhaseIncrement * (outFrameCount - 1) + mPhaseFraction) / phaseWrapLimit;
+    //  phaseWrapLimit is the wraparound (1 << kNumPhaseBits), if not specified explicitly.
+    //
+    inline size_t getInFrameCountRequired(size_t outFrameCount) {
+        return (static_cast<uint64_t>(outFrameCount)*mInSampleRate
+                + (mSampleRate - 1))/mSampleRate;
+    }
+
 private:
     const src_quality mQuality;
 

services/audioflinger/AudioResamplerCubic.cpp

diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp
index 1f9714b..8f14ff9 100644
--- a/services/audioflinger/AudioResamplerCubic.cpp
+++ b/services/audioflinger/AudioResamplerCubic.cpp
@@ -60,7 +60,7 @@
     uint32_t phaseIncrement = mPhaseIncrement;
     size_t outputIndex = 0;
     size_t outputSampleCount = outFrameCount * 2;
-    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
+    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
 
     // fetch first buffer
     if (mBuffer.frameCount == 0) {
@@ -128,7 +128,7 @@
     uint32_t phaseIncrement = mPhaseIncrement;
     size_t outputIndex = 0;
     size_t outputSampleCount = outFrameCount * 2;
-    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
+    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
 
     // fetch first buffer
     if (mBuffer.frameCount == 0) {

services/audioflinger/AudioResamplerDyn.cpp

diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
new file mode 100644
index 0000000..7e4ca0c
--- /dev/null
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -0,0 +1,559 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioResamplerDyn"
+//#define LOG_NDEBUG 0
+
+#include <malloc.h>
+#include <string.h>
+#include <stdlib.h>
+#include <dlfcn.h>
+#include <math.h>
+
+#include <cutils/compiler.h>
+#include <cutils/properties.h>
+#include <utils/Log.h>
+
+#include "AudioResamplerFirOps.h" // USE_NEON and USE_INLINE_ASSEMBLY defined here
+#include "AudioResamplerFirProcess.h"
+#include "AudioResamplerFirProcessNeon.h"
+#include "AudioResamplerFirGen.h" // requires math.h
+#include "AudioResamplerDyn.h"
+
+//#define DEBUG_RESAMPLER
+
+namespace android {
+
+// generate a unique resample type compile-time constant (constexpr)
+#define RESAMPLETYPE(CHANNELS, LOCKED, STRIDE, COEFTYPE) \
+    ((((CHANNELS)-1)&1) | !!(LOCKED)<<1 | (COEFTYPE)<<2 \
+    | ((STRIDE)==8 ? 1 : (STRIDE)==16 ? 2 : 0)<<3)
+
+/*
+ * InBuffer is a type agnostic input buffer.
+ *
+ * Layout of the state buffer for halfNumCoefs=8.
+ *
+ * [rrrrrrppppppppnnnnnnnnrrrrrrrrrrrrrrrrrrr.... rrrrrrr]
+ *  S            I                                R
+ *
+ * S = mState
+ * I = mImpulse
+ * R = mRingFull
+ * p = past samples, convoluted with the (p)ositive side of sinc()
+ * n = future samples, convoluted with the (n)egative side of sinc()
+ * r = extra space for implementing the ring buffer
+ */
+
+template<typename TI>
+AudioResamplerDyn::InBuffer<TI>::InBuffer()
+    : mState(NULL), mImpulse(NULL), mRingFull(NULL), mStateSize(0) {
+}
+
+template<typename TI>
+AudioResamplerDyn::InBuffer<TI>::~InBuffer() {
+    init();
+}
+
+template<typename TI>
+void AudioResamplerDyn::InBuffer<TI>::init() {
+    free(mState);
+    mState = NULL;
+    mImpulse = NULL;
+    mRingFull = NULL;
+    mStateSize = 0;
+}
+
+// resizes the state buffer to accommodate the appropriate filter length
+template<typename TI>
+void AudioResamplerDyn::InBuffer<TI>::resize(int CHANNELS, int halfNumCoefs) {
+    // calculate desired state size
+    int stateSize = halfNumCoefs * CHANNELS * 2
+            * kStateSizeMultipleOfFilterLength;
+
+    // check if buffer needs resizing
+    if (mState
+            && stateSize == mStateSize
+            && mRingFull-mState == mStateSize-halfNumCoefs*CHANNELS) {
+        return;
+    }
+
+    // create new buffer
+    TI* state = (int16_t*)memalign(32, stateSize*sizeof(*state));
+    memset(state, 0, stateSize*sizeof(*state));
+
+    // attempt to preserve state
+    if (mState) {
+        TI* srcLo = mImpulse - halfNumCoefs*CHANNELS;
+        TI* srcHi = mImpulse + halfNumCoefs*CHANNELS;
+        TI* dst = state;
+
+        if (srcLo < mState) {
+            dst += mState-srcLo;
+            srcLo = mState;
+        }
+        if (srcHi > mState + mStateSize) {
+            srcHi = mState + mStateSize;
+        }
+        memcpy(dst, srcLo, (srcHi - srcLo) * sizeof(*srcLo));
+        free(mState);
+    }
+
+    // set class member vars
+    mState = state;
+    mStateSize = stateSize;
+    mImpulse = mState + halfNumCoefs*CHANNELS; // actually one sample greater than needed
+    mRingFull = mState + mStateSize - halfNumCoefs*CHANNELS;
+}
+
+// copy in the input data into the head (impulse+halfNumCoefs) of the buffer.
+template<typename TI>
+template<int CHANNELS>
+void AudioResamplerDyn::InBuffer<TI>::readAgain(TI*& impulse, const int halfNumCoefs,
+        const TI* const in, const size_t inputIndex) {
+    int16_t* head = impulse + halfNumCoefs*CHANNELS;
+    for (size_t i=0 ; i<CHANNELS ; i++) {
+        head[i] = in[inputIndex*CHANNELS + i];
+    }
+}
+
+// advance the impulse pointer, and load in data into the head (impulse+halfNumCoefs)
+template<typename TI>
+template<int CHANNELS>
+void AudioResamplerDyn::InBuffer<TI>::readAdvance(TI*& impulse, const int halfNumCoefs,
+        const TI* const in, const size_t inputIndex) {
+    impulse += CHANNELS;
+
+    if (CC_UNLIKELY(impulse >= mRingFull)) {
+        const size_t shiftDown = mRingFull - mState - halfNumCoefs*CHANNELS;
+        memcpy(mState, mState+shiftDown, halfNumCoefs*CHANNELS*2*sizeof(TI));
+        impulse -= shiftDown;
+    }
+    readAgain<CHANNELS>(impulse, halfNumCoefs, in, inputIndex);
+}
+
+void AudioResamplerDyn::Constants::set(
+        int L, int halfNumCoefs, int inSampleRate, int outSampleRate)
+{
+    int bits = 0;
+    int lscale = inSampleRate/outSampleRate < 2 ? L - 1 :
+            static_cast<int>(static_cast<uint64_t>(L)*inSampleRate/outSampleRate);
+    for (int i=lscale; i; ++bits, i>>=1)
+        ;
+    mL = L;
+    mShift = kNumPhaseBits - bits;
+    mHalfNumCoefs = halfNumCoefs;
+}
+
+AudioResamplerDyn::AudioResamplerDyn(int bitDepth,
+        int inChannelCount, int32_t sampleRate, src_quality quality)
+    : AudioResampler(bitDepth, inChannelCount, sampleRate, quality),
+    mResampleType(0), mFilterSampleRate(0), mFilterQuality(DEFAULT_QUALITY),
+    mCoefBuffer(NULL)
+{
+    mVolumeSimd[0] = mVolumeSimd[1] = 0;
+    // The AudioResampler base class assumes we are always ready for 1:1 resampling.
+    // We reset mInSampleRate to 0, so setSampleRate() will calculate filters for
+    // setSampleRate() for 1:1. (May be removed if precalculated filters are used.)
+    mInSampleRate = 0;
+    mConstants.set(128, 8, mSampleRate, mSampleRate); // TODO: set better
+}
+
+AudioResamplerDyn::~AudioResamplerDyn() {
+    free(mCoefBuffer);
+}
+
+void AudioResamplerDyn::init() {
+    mFilterSampleRate = 0; // always trigger new filter generation
+    mInBuffer.init();
+}
+
+void AudioResamplerDyn::setVolume(int16_t left, int16_t right) {
+    AudioResampler::setVolume(left, right);
+    mVolumeSimd[0] = static_cast<int32_t>(left)<<16;
+    mVolumeSimd[1] = static_cast<int32_t>(right)<<16;
+}
+
+template <typename T> T max(T a, T b) {return a > b ? a : b;}
+
+template <typename T> T absdiff(T a, T b) {return a > b ? a - b : b - a;}
+
+template<typename T>
+void AudioResamplerDyn::createKaiserFir(Constants &c, double stopBandAtten,
+        int inSampleRate, int outSampleRate, double tbwCheat) {
+    T* buf = reinterpret_cast<T*>(memalign(32, (c.mL+1)*c.mHalfNumCoefs*sizeof(T)));
+    static const double atten = 0.9998;   // to avoid ripple overflow
+    double fcr;
+    double tbw = firKaiserTbw(c.mHalfNumCoefs, stopBandAtten);
+
+    if (inSampleRate < outSampleRate) { // upsample
+        fcr = max(0.5*tbwCheat - tbw/2, tbw/2);
+    } else { // downsample
+        fcr = max(0.5*tbwCheat*outSampleRate/inSampleRate - tbw/2, tbw/2);
+    }
+    // create and set filter
+    firKaiserGen(buf, c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, atten);
+    c.setBuf(buf);
+    if (mCoefBuffer) {
+        free(mCoefBuffer);
+    }
+    mCoefBuffer = buf;
+#ifdef DEBUG_RESAMPLER
+    // print basic filter stats
+    printf("L:%d  hnc:%d  stopBandAtten:%lf  fcr:%lf  atten:%lf  tbw:%lf\n",
+            c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, atten, tbw);
+    // test the filter and report results
+    double fp = (fcr - tbw/2)/c.mL;
+    double fs = (fcr + tbw/2)/c.mL;
+    double passMin, passMax, passRipple;
+    double stopMax, stopRipple;
+    testFir(buf, c.mL, c.mHalfNumCoefs, fp, fs, /*passSteps*/ 1000, /*stopSteps*/ 100000,
+            passMin, passMax, passRipple, stopMax, stopRipple);
+    printf("passband(%lf, %lf): %.8lf %.8lf %.8lf\n", 0., fp, passMin, passMax, passRipple);
+    printf("stopband(%lf, %lf): %.8lf %.3lf\n", fs, 0.5, stopMax, stopRipple);
+#endif
+}
+
+// recursive gcd. Using objdump, it appears the tail recursion is converted to a while loop.
+static int gcd(int n, int m) {
+    if (m == 0) {
+        return n;
+    }
+    return gcd(m, n % m);
+}
+
+static bool isClose(int32_t newSampleRate, int32_t prevSampleRate,
+        int32_t filterSampleRate, int32_t outSampleRate) {
+
+    // different upsampling ratios do not need a filter change.
+    if (filterSampleRate != 0
+            && filterSampleRate < outSampleRate
+            && newSampleRate < outSampleRate)
+        return true;
+
+    // check design criteria again if downsampling is detected.
+    int pdiff = absdiff(newSampleRate, prevSampleRate);
+    int adiff = absdiff(newSampleRate, filterSampleRate);
+
+    // allow up to 6% relative change increments.
+    // allow up to 12% absolute change increments (from filter design)
+    return pdiff < prevSampleRate>>4 && adiff < filterSampleRate>>3;
+}
+
+void AudioResamplerDyn::setSampleRate(int32_t inSampleRate) {
+    if (mInSampleRate == inSampleRate) {
+        return;
+    }
+    int32_t oldSampleRate = mInSampleRate;
+    int32_t oldHalfNumCoefs = mConstants.mHalfNumCoefs;
+    uint32_t oldPhaseWrapLimit = mConstants.mL << mConstants.mShift;
+    bool useS32 = false;
+
+    mInSampleRate = inSampleRate;
+
+    // TODO: Add precalculated Equiripple filters
+
+    if (mFilterQuality != getQuality() ||
+            !isClose(inSampleRate, oldSampleRate, mFilterSampleRate, mSampleRate)) {
+        mFilterSampleRate = inSampleRate;
+        mFilterQuality = getQuality();
+
+        // Begin Kaiser Filter computation
+        //
+        // The quantization floor for S16 is about 96db - 10*log_10(#length) + 3dB.
+        // Keep the stop band attenuation no greater than 84-85dB for 32 length S16 filters
+        //
+        // For s32 we keep the stop band attenuation at the same as 16b resolution, about
+        // 96-98dB
+        //
+
+        double stopBandAtten;
+        double tbwCheat = 1.; // how much we "cheat" into aliasing
+        int halfLength;
+        if (mFilterQuality == DYN_HIGH_QUALITY) {
+            // 32b coefficients, 64 length
+            useS32 = true;
+            stopBandAtten = 98.;
+            if (inSampleRate >= mSampleRate * 4) {
+                halfLength = 48;
+            } else if (inSampleRate >= mSampleRate * 2) {
+                halfLength = 40;
+            } else {
+                halfLength = 32;
+            }
+        } else if (mFilterQuality == DYN_LOW_QUALITY) {
+            // 16b coefficients, 16-32 length
+            useS32 = false;
+            stopBandAtten = 80.;
+            if (inSampleRate >= mSampleRate * 4) {
+                halfLength = 24;
+            } else if (inSampleRate >= mSampleRate * 2) {
+                halfLength = 16;
+            } else {
+                halfLength = 8;
+            }
+            if (inSampleRate <= mSampleRate) {
+                tbwCheat = 1.05;
+            } else {
+                tbwCheat = 1.03;
+            }
+        } else { // DYN_MED_QUALITY
+            // 16b coefficients, 32-64 length
+            // note: > 64 length filters with 16b coefs can have quantization noise problems
+            useS32 = false;
+            stopBandAtten = 84.;
+            if (inSampleRate >= mSampleRate * 4) {
+                halfLength = 32;
+            } else if (inSampleRate >= mSampleRate * 2) {
+                halfLength = 24;
+            } else {
+                halfLength = 16;
+            }
+            if (inSampleRate <= mSampleRate) {
+                tbwCheat = 1.03;
+            } else {
+                tbwCheat = 1.01;
+            }
+        }
+
+        // determine the number of polyphases in the filterbank.
+        // for 16b, it is desirable to have 2^(16/2) = 256 phases.
+        // https://ccrma.stanford.edu/~jos/resample/Relation_Interpolation_Error_Quantization.html
+        //
+        // We are a bit more lax on this.
+
+        int phases = mSampleRate / gcd(mSampleRate, inSampleRate);
+
+        // TODO: Once dynamic sample rate change is an option, the code below
+        // should be modified to execute only when dynamic sample rate change is enabled.
+        //
+        // as above, #phases less than 63 is too few phases for accurate linear interpolation.
+        // we increase the phases to compensate, but more phases means more memory per
+        // filter and more time to compute the filter.
+        //
+        // if we know that the filter will be used for dynamic sample rate changes,
+        // that would allow us skip this part for fixed sample rate resamplers.
+        //
+        while (phases<63) {
+            phases *= 2; // this code only needed to support dynamic rate changes
+        }
+
+        if (phases>=256) {  // too many phases, always interpolate
+            phases = 127;
+        }
+
+        // create the filter
+        mConstants.set(phases, halfLength, inSampleRate, mSampleRate);
+        if (useS32) {
+            createKaiserFir<int32_t>(mConstants, stopBandAtten,
+                    inSampleRate, mSampleRate, tbwCheat);
+        } else {
+            createKaiserFir<int16_t>(mConstants, stopBandAtten,
+                    inSampleRate, mSampleRate, tbwCheat);
+        }
+    } // End Kaiser filter
+
+    // update phase and state based on the new filter.
+    const Constants& c(mConstants);
+    mInBuffer.resize(mChannelCount, c.mHalfNumCoefs);
+    const uint32_t phaseWrapLimit = c.mL << c.mShift;
+    // try to preserve as much of the phase fraction as possible for on-the-fly changes
+    mPhaseFraction = static_cast<unsigned long long>(mPhaseFraction)
+            * phaseWrapLimit / oldPhaseWrapLimit;
+    mPhaseFraction %= phaseWrapLimit; // should not do anything, but just in case.
+    mPhaseIncrement = static_cast<uint32_t>(static_cast<double>(phaseWrapLimit)
+            * inSampleRate / mSampleRate);
+
+    // determine which resampler to use
+    // check if locked phase (works only if mPhaseIncrement has no "fractional phase bits")
+    int locked = (mPhaseIncrement << (sizeof(mPhaseIncrement)*8 - c.mShift)) == 0;
+    int stride = (c.mHalfNumCoefs&7)==0 ? 16 : (c.mHalfNumCoefs&3)==0 ? 8 : 2;
+    if (locked) {
+        mPhaseFraction = mPhaseFraction >> c.mShift << c.mShift; // remove fractional phase
+    }
+
+    mResampleType = RESAMPLETYPE(mChannelCount, locked, stride, !!useS32);
+#ifdef DEBUG_RESAMPLER
+    printf("channels:%d  %s  stride:%d  %s  coef:%d  shift:%d\n",
+            mChannelCount, locked ? "locked" : "interpolated",
+            stride, useS32 ? "S32" : "S16", 2*c.mHalfNumCoefs, c.mShift);
+#endif
+}
+
+void AudioResamplerDyn::resample(int32_t* out, size_t outFrameCount,
+            AudioBufferProvider* provider)
+{
+    // TODO:
+    // 24 cases - this perhaps can be reduced later, as testing might take too long
+    switch (mResampleType) {
+
+    // stride 16 (falls back to stride 2 for machines that do not support NEON)
+    case RESAMPLETYPE(1, true, 16, 0):
+        return resample<1, true, 16>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+    case RESAMPLETYPE(2, true, 16, 0):
+        return resample<2, true, 16>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+    case RESAMPLETYPE(1, false, 16, 0):
+        return resample<1, false, 16>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+    case RESAMPLETYPE(2, false, 16, 0):
+        return resample<2, false, 16>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+    case RESAMPLETYPE(1, true, 16, 1):
+        return resample<1, true, 16>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+    case RESAMPLETYPE(2, true, 16, 1):
+        return resample<2, true, 16>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+    case RESAMPLETYPE(1, false, 16, 1):
+        return resample<1, false, 16>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+    case RESAMPLETYPE(2, false, 16, 1):
+        return resample<2, false, 16>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+#if 0
+    // TODO: Remove these?
+    // stride 8
+    case RESAMPLETYPE(1, true, 8, 0):
+        return resample<1, true, 8>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+    case RESAMPLETYPE(2, true, 8, 0):
+        return resample<2, true, 8>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+    case RESAMPLETYPE(1, false, 8, 0):
+        return resample<1, false, 8>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+    case RESAMPLETYPE(2, false, 8, 0):
+        return resample<2, false, 8>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+    case RESAMPLETYPE(1, true, 8, 1):
+        return resample<1, true, 8>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+    case RESAMPLETYPE(2, true, 8, 1):
+        return resample<2, true, 8>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+    case RESAMPLETYPE(1, false, 8, 1):
+        return resample<1, false, 8>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+    case RESAMPLETYPE(2, false, 8, 1):
+        return resample<2, false, 8>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+    // stride 2 (can handle any filter length)
+    case RESAMPLETYPE(1, true, 2, 0):
+        return resample<1, true, 2>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+    case RESAMPLETYPE(2, true, 2, 0):
+        return resample<2, true, 2>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+    case RESAMPLETYPE(1, false, 2, 0):
+        return resample<1, false, 2>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+    case RESAMPLETYPE(2, false, 2, 0):
+        return resample<2, false, 2>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+    case RESAMPLETYPE(1, true, 2, 1):
+        return resample<1, true, 2>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+    case RESAMPLETYPE(2, true, 2, 1):
+        return resample<2, true, 2>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+    case RESAMPLETYPE(1, false, 2, 1):
+        return resample<1, false, 2>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+    case RESAMPLETYPE(2, false, 2, 1):
+        return resample<2, false, 2>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+#endif
+    default:
+        ; // error
+    }
+}
+
+template<int CHANNELS, bool LOCKED, int STRIDE, typename TC>
+void AudioResamplerDyn::resample(int32_t* out, size_t outFrameCount,
+        const TC* const coefs,  AudioBufferProvider* provider)
+{
+    const Constants& c(mConstants);
+    int16_t* impulse = mInBuffer.getImpulse();
+    size_t inputIndex = mInputIndex;
+    uint32_t phaseFraction = mPhaseFraction;
+    const uint32_t phaseIncrement = mPhaseIncrement;
+    size_t outputIndex = 0;
+    size_t outputSampleCount = outFrameCount * 2;   // stereo output
+    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
+    const uint32_t phaseWrapLimit = c.mL << c.mShift;
+
+    // NOTE: be very careful when modifying the code here. register
+    // pressure is very high and a small change might cause the compiler
+    // to generate far less efficient code.
+    // Always sanity check the result with objdump or test-resample.
+
+    // the following logic is a bit convoluted to keep the main processing loop
+    // as tight as possible with register allocation.
+    while (outputIndex < outputSampleCount) {
+        // buffer is empty, fetch a new one
+        while (mBuffer.frameCount == 0) {
+            mBuffer.frameCount = inFrameCount;
+            provider->getNextBuffer(&mBuffer,
+                    calculateOutputPTS(outputIndex / 2));
+            if (mBuffer.raw == NULL) {
+                goto resample_exit;
+            }
+            if (phaseFraction >= phaseWrapLimit) { // read in data
+                mInBuffer.readAdvance<CHANNELS>(
+                        impulse, c.mHalfNumCoefs, mBuffer.i16, inputIndex);
+                phaseFraction -= phaseWrapLimit;
+                while (phaseFraction >= phaseWrapLimit) {
+                    inputIndex++;
+                    if (inputIndex >= mBuffer.frameCount) {
+                        inputIndex -= mBuffer.frameCount;
+                        provider->releaseBuffer(&mBuffer);
+                        break;
+                    }
+                    mInBuffer.readAdvance<CHANNELS>(
+                            impulse, c.mHalfNumCoefs, mBuffer.i16, inputIndex);
+                    phaseFraction -= phaseWrapLimit;
+                }
+            }
+        }
+        const int16_t* const in = mBuffer.i16;
+        const size_t frameCount = mBuffer.frameCount;
+        const int coefShift = c.mShift;
+        const int halfNumCoefs = c.mHalfNumCoefs;
+        const int32_t* const volumeSimd = mVolumeSimd;
+
+        // reread the last input in.
+        mInBuffer.readAgain<CHANNELS>(impulse, halfNumCoefs, in, inputIndex);
+
+        // main processing loop
+        while (CC_LIKELY(outputIndex < outputSampleCount)) {
+            // caution: fir() is inlined and may be large.
+            // output will be loaded with the appropriate values
+            //
+            // from the input samples in impulse[-halfNumCoefs+1]... impulse[halfNumCoefs]
+            // from the polyphase filter of (phaseFraction / phaseWrapLimit) in coefs.
+            //
+            fir<CHANNELS, LOCKED, STRIDE>(
+                    &out[outputIndex],
+                    phaseFraction, phaseWrapLimit,
+                    coefShift, halfNumCoefs, coefs,
+                    impulse, volumeSimd);
+            outputIndex += 2;
+
+            phaseFraction += phaseIncrement;
+            while (phaseFraction >= phaseWrapLimit) {
+                inputIndex++;
+                if (inputIndex >= frameCount) {
+                    goto done;  // need a new buffer
+                }
+                mInBuffer.readAdvance<CHANNELS>(impulse, halfNumCoefs, in, inputIndex);
+                phaseFraction -= phaseWrapLimit;
+            }
+        }
+done:
+        // often arrives here when input buffer runs out
+        if (inputIndex >= frameCount) {
+            inputIndex -= frameCount;
+            provider->releaseBuffer(&mBuffer);
+            // mBuffer.frameCount MUST be zero here.
+        }
+    }
+
+resample_exit:
+    mInBuffer.setImpulse(impulse);
+    mInputIndex = inputIndex;
+    mPhaseFraction = phaseFraction;
+}
+
+// ----------------------------------------------------------------------------
+}; // namespace android

services/audioflinger/AudioResamplerDyn.h

diff --git a/services/audioflinger/AudioResamplerDyn.h b/services/audioflinger/AudioResamplerDyn.h
new file mode 100644
index 0000000..df1fdbe
--- /dev/null
+++ b/services/audioflinger/AudioResamplerDyn.h
@@ -0,0 +1,124 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIO_RESAMPLER_DYN_H
+#define ANDROID_AUDIO_RESAMPLER_DYN_H
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <cutils/log.h>
+
+#include "AudioResampler.h"
+
+namespace android {
+
+class AudioResamplerDyn: public AudioResampler {
+public:
+    AudioResamplerDyn(int bitDepth, int inChannelCount, int32_t sampleRate,
+            src_quality quality);
+
+    virtual ~AudioResamplerDyn();
+
+    virtual void init();
+
+    virtual void setSampleRate(int32_t inSampleRate);
+
+    virtual void setVolume(int16_t left, int16_t right);
+
+    virtual void resample(int32_t* out, size_t outFrameCount,
+            AudioBufferProvider* provider);
+
+private:
+
+    class Constants { // stores the filter constants.
+    public:
+        Constants() :
+            mL(0), mShift(0), mHalfNumCoefs(0), mFirCoefsS16(NULL)
+        {}
+        void set(int L, int halfNumCoefs,
+                int inSampleRate, int outSampleRate);
+        inline void setBuf(int16_t* buf) {
+            mFirCoefsS16 = buf;
+        }
+        inline void setBuf(int32_t* buf) {
+            mFirCoefsS32 = buf;
+        }
+
+        int mL;       // interpolation phases in the filter.
+        int mShift;   // right shift to get polyphase index
+        unsigned int mHalfNumCoefs; // filter half #coefs
+        union {       // polyphase filter bank
+            const int16_t* mFirCoefsS16;
+            const int32_t* mFirCoefsS32;
+        };
+    };
+
+    // Input buffer management for a given input type TI, now (int16_t)
+    // Is agnostic of the actual type, can work with int32_t and float.
+    template<typename TI>
+    class InBuffer {
+    public:
+        InBuffer();
+        ~InBuffer();
+        void init();
+        void resize(int CHANNELS, int halfNumCoefs);
+
+        // used for direct management of the mImpulse pointer
+        inline TI* getImpulse() {
+            return mImpulse;
+        }
+        inline void setImpulse(TI *impulse) {
+            mImpulse = impulse;
+        }
+        template<int CHANNELS>
+        inline void readAgain(TI*& impulse, const int halfNumCoefs,
+                const TI* const in, const size_t inputIndex);
+        template<int CHANNELS>
+        inline void readAdvance(TI*& impulse, const int halfNumCoefs,
+                const TI* const in, const size_t inputIndex);
+
+    private:
+        // tuning parameter guidelines: 2 <= multiple <= 8
+        static const int kStateSizeMultipleOfFilterLength = 4;
+
+        TI* mState;    // base pointer for the input buffer storage
+        TI* mImpulse;  // current location of the impulse response (centered)
+        TI* mRingFull; // mState <= mImpulse < mRingFull
+        // in general, mRingFull = mState + mStateSize - halfNumCoefs*CHANNELS.
+        size_t mStateSize; // in units of TI.
+    };
+
+    template<int CHANNELS, bool LOCKED, int STRIDE, typename TC>
+    void resample(int32_t* out, size_t outFrameCount,
+            const TC* const coefs, AudioBufferProvider* provider);
+
+    template<typename T>
+    void createKaiserFir(Constants &c, double stopBandAtten,
+            int inSampleRate, int outSampleRate, double tbwCheat);
+
+    InBuffer<int16_t> mInBuffer;
+    Constants mConstants;  // current set of coefficient parameters
+    int32_t __attribute__ ((aligned (8))) mVolumeSimd[2];
+    int32_t mResampleType; // contains the resample type.
+    int32_t mFilterSampleRate; // designed filter sample rate.
+    src_quality mFilterQuality; // designed filter quality.
+    void* mCoefBuffer; // if a filter is created, this is not null
+};
+
+// ----------------------------------------------------------------------------
+}; // namespace android
+
+#endif /*ANDROID_AUDIO_RESAMPLER_DYN_H*/

services/audioflinger/AudioResamplerFirGen.h

diff --git a/services/audioflinger/AudioResamplerFirGen.h b/services/audioflinger/AudioResamplerFirGen.h
new file mode 100644
index 0000000..fac3001
--- /dev/null
+++ b/services/audioflinger/AudioResamplerFirGen.h
@@ -0,0 +1,684 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIO_RESAMPLER_FIR_GEN_H
+#define ANDROID_AUDIO_RESAMPLER_FIR_GEN_H
+
+namespace android {
+
+/*
+ * generates a sine wave at equal steps.
+ *
+ * As most of our functions use sine or cosine at equal steps,
+ * it is very efficient to compute them that way (single multiply and subtract),
+ * rather than invoking the math library sin() or cos() each time.
+ *
+ * SineGen uses Goertzel's Algorithm (as a generator not a filter)
+ * to calculate sine(wstart + n * wstep) or cosine(wstart + n * wstep)
+ * by stepping through 0, 1, ... n.
+ *
+ * e^i(wstart+wstep) = 2cos(wstep) * e^i(wstart) - e^i(wstart-wstep)
+ *
+ * or looking at just the imaginary sine term, as the cosine follows identically:
+ *
+ * sin(wstart+wstep) = 2cos(wstep) * sin(wstart) - sin(wstart-wstep)
+ *
+ * Goertzel's algorithm is more efficient than the angle addition formula,
+ * e^i(wstart+wstep) = e^i(wstart) * e^i(wstep), which takes up to
+ * 4 multiplies and 2 adds (or 3* and 3+) and requires both sine and
+ * cosine generation due to the complex * complex multiply (full rotation).
+ *
+ * See: http://en.wikipedia.org/wiki/Goertzel_algorithm
+ *
+ */
+
+class SineGen {
+public:
+    SineGen(double wstart, double wstep, bool cosine = false) {
+        if (cosine) {
+            mCurrent = cos(wstart);
+            mPrevious = cos(wstart - wstep);
+        } else {
+            mCurrent = sin(wstart);
+            mPrevious = sin(wstart - wstep);
+        }
+        mTwoCos = 2.*cos(wstep);
+    }
+    SineGen(double expNow, double expPrev, double twoCosStep) {
+        mCurrent = expNow;
+        mPrevious = expPrev;
+        mTwoCos = twoCosStep;
+    }
+    inline double value() const {
+        return mCurrent;
+    }
+    inline void advance() {
+        double tmp = mCurrent;
+        mCurrent = mCurrent*mTwoCos - mPrevious;
+        mPrevious = tmp;
+    }
+    inline double valueAdvance() {
+        double tmp = mCurrent;
+        mCurrent = mCurrent*mTwoCos - mPrevious;
+        mPrevious = tmp;
+        return tmp;
+    }
+
+private:
+    double mCurrent; // current value of sine/cosine
+    double mPrevious; // previous value of sine/cosine
+    double mTwoCos; // stepping factor
+};
+
+/*
+ * generates a series of sine generators, phase offset by fixed steps.
+ *
+ * This is used to generate polyphase sine generators, one per polyphase
+ * in the filter code below.
+ *
+ * The SineGen returned by value() starts at innerStart = outerStart + n*outerStep;
+ * increments by innerStep.
+ *
+ */
+
+class SineGenGen {
+public:
+    SineGenGen(double outerStart, double outerStep, double innerStep, bool cosine = false)
+            : mSineInnerCur(outerStart, outerStep, cosine),
+              mSineInnerPrev(outerStart-innerStep, outerStep, cosine)
+    {
+        mTwoCos = 2.*cos(innerStep);
+    }
+    inline SineGen value() {
+        return SineGen(mSineInnerCur.value(), mSineInnerPrev.value(), mTwoCos);
+    }
+    inline void advance() {
+        mSineInnerCur.advance();
+        mSineInnerPrev.advance();
+    }
+    inline SineGen valueAdvance() {
+        return SineGen(mSineInnerCur.valueAdvance(), mSineInnerPrev.valueAdvance(), mTwoCos);
+    }
+
+private:
+    SineGen mSineInnerCur; // generate the inner sine values (stepped by outerStep).
+    SineGen mSineInnerPrev; // generate the inner sine previous values
+                            // (behind by innerStep, stepped by outerStep).
+    double mTwoCos; // the inner stepping factor for the returned SineGen.
+};
+
+static inline double sqr(double x) {
+    return x * x;
+}
+
+/*
+ * rounds a double to the nearest integer for FIR coefficients.
+ *
+ * One variant uses noise shaping, which must keep error history
+ * to work (the err parameter, initialized to 0).
+ * The other variant is a non-noise shaped version for
+ * S32 coefficients (noise shaping doesn't gain much).
+ *
+ * Caution: No bounds saturation is applied, but isn't needed in this case.
+ *
+ * @param x is the value to round.
+ *
+ * @param maxval is the maximum integer scale factor expressed as an int64 (for headroom).
+ * Typically this may be the maximum positive integer+1 (using the fact that double precision
+ * FIR coefficients generated here are never that close to 1.0 to pose an overflow condition).
+ *
+ * @param err is the previous error (actual - rounded) for the previous rounding op.
+ * For 16b coefficients this can improve stopband dB performance by up to 2dB.
+ *
+ * Many variants exist for the noise shaping: http://en.wikipedia.org/wiki/Noise_shaping
+ *
+ */
+
+static inline int64_t toint(double x, int64_t maxval, double& err) {
+    double val = x * maxval;
+    double ival = floor(val + 0.5 + err*0.2);
+    err = val - ival;
+    return static_cast<int64_t>(ival);
+}
+
+static inline int64_t toint(double x, int64_t maxval) {
+    return static_cast<int64_t>(floor(x * maxval + 0.5));
+}
+
+/*
+ * Modified Bessel function of the first kind
+ * http://en.wikipedia.org/wiki/Bessel_function
+ *
+ * The formulas are taken from Abramowitz and Stegun,
+ * _Handbook of Mathematical Functions_ (links below):
+ *
+ * http://people.math.sfu.ca/~cbm/aands/page_375.htm
+ * http://people.math.sfu.ca/~cbm/aands/page_378.htm
+ *
+ * http://dlmf.nist.gov/10.25
+ * http://dlmf.nist.gov/10.40
+ *
+ * Note we assume x is nonnegative (the function is symmetric,
+ * pass in the absolute value as needed).
+ *
+ * Constants are compile time derived with templates I0Term<> and
+ * I0ATerm<> to the precision of the compiler.  The series can be expanded
+ * to any precision needed, but currently set around 24b precision.
+ *
+ * We use a bit of template math here, constexpr would probably be
+ * more appropriate for a C++11 compiler.
+ *
+ * For the intermediate range 3.75 < x < 15, we use minimax polynomial fit.
+ *
+ */
+
+template <int N>
+struct I0Term {
+    static const double value = I0Term<N-1>::value / (4. * N * N);
+};
+
+template <>
+struct I0Term<0> {
+    static const double value = 1.;
+};
+
+template <int N>
+struct I0ATerm {
+    static const double value = I0ATerm<N-1>::value * (2.*N-1.) * (2.*N-1.) / (8. * N);
+};
+
+template <>
+struct I0ATerm<0> { // 1/sqrt(2*PI);
+    static const double value = 0.398942280401432677939946059934381868475858631164934657665925;
+};
+
+#if USE_HORNERS_METHOD
+/* Polynomial evaluation of A + Bx + Cx^2 + Dx^3 + ...
+ * using Horner's Method: http://en.wikipedia.org/wiki/Horner's_method
+ *
+ * This has fewer multiplications than Estrin's method below, but has back to back
+ * floating point dependencies.
+ *
+ * On ARM this appears to work slower, so USE_HORNERS_METHOD is not default enabled.
+ */
+
+inline double Poly2(double A, double B, double x) {
+    return A + x * B;
+}
+
+inline double Poly4(double A, double B, double C, double D, double x) {
+    return A + x * (B + x * (C + x * (D)));
+}
+
+inline double Poly7(double A, double B, double C, double D, double E, double F, double G,
+        double x) {
+    return A + x * (B + x * (C + x * (D + x * (E + x * (F + x * (G))))));
+}
+
+inline double Poly9(double A, double B, double C, double D, double E, double F, double G,
+        double H, double I, double x) {
+    return A + x * (B + x * (C + x * (D + x * (E + x * (F + x * (G + x * (H + x * (I))))))));
+}
+
+#else
+/* Polynomial evaluation of A + Bx + Cx^2 + Dx^3 + ...
+ * using Estrin's Method: http://en.wikipedia.org/wiki/Estrin's_scheme
+ *
+ * This is typically faster, perhaps gains about 5-10% overall on ARM processors
+ * over Horner's method above.
+ */
+
+inline double Poly2(double A, double B, double x) {
+    return A + B * x;
+}
+
+inline double Poly3(double A, double B, double C, double x, double x2) {
+    return Poly2(A, B, x) + C * x2;
+}
+
+inline double Poly3(double A, double B, double C, double x) {
+    return Poly2(A, B, x) + C * x * x;
+}
+
+inline double Poly4(double A, double B, double C, double D, double x, double x2) {
+    return Poly2(A, B, x) + Poly2(C, D, x) * x2; // same as poly2(poly2, poly2, x2);
+}
+
+inline double Poly4(double A, double B, double C, double D, double x) {
+    return Poly4(A, B, C, D, x, x * x);
+}
+
+inline double Poly7(double A, double B, double C, double D, double E, double F, double G,
+        double x) {
+    double x2 = x * x;
+    return Poly4(A, B, C, D, x, x2) + Poly3(E, F, G, x, x2) * (x2 * x2);
+}
+
+inline double Poly8(double A, double B, double C, double D, double E, double F, double G,
+        double H, double x, double x2, double x4) {
+    return Poly4(A, B, C, D, x, x2) + Poly4(E, F, G, H, x, x2) * x4;
+}
+
+inline double Poly9(double A, double B, double C, double D, double E, double F, double G,
+        double H, double I, double x) {
+    double x2 = x * x;
+#if 1
+    // It does not seem faster to explicitly decompose Poly8 into Poly4, but
+    // could depend on compiler floating point scheduling.
+    double x4 = x2 * x2;
+    return Poly8(A, B, C, D, E, F, G, H, x, x2, x4) + I * (x4 * x4);
+#else
+    double val = Poly4(A, B, C, D, x, x2);
+    double x4 = x2 * x2;
+    return val + Poly4(E, F, G, H, x, x2) * x4 + I * (x4 * x4);
+#endif
+}
+#endif
+
+static inline double I0(double x) {
+    if (x < 3.75) {
+        x *= x;
+        return Poly7(I0Term<0>::value, I0Term<1>::value,
+                I0Term<2>::value, I0Term<3>::value,
+                I0Term<4>::value, I0Term<5>::value,
+                I0Term<6>::value, x); // e < 1.6e-7
+    }
+    if (1) {
+        /*
+         * Series expansion coefs are easy to calculate, but are expanded around 0,
+         * so error is unequal over the interval 0 < x < 3.75, the error being
+         * significantly better near 0.
+         *
+         * A better solution is to use precise minimax polynomial fits.
+         *
+         * We use a slightly more complicated solution for 3.75 < x < 15, based on
+         * the tables in Blair and Edwards, "Stable Rational Minimax Approximations
+         * to the Modified Bessel Functions I0(x) and I1(x)", Chalk Hill Nuclear Laboratory,
+         * AECL-4928.
+         *
+         * http://www.iaea.org/inis/collection/NCLCollectionStore/_Public/06/178/6178667.pdf
+         *
+         * See Table 11 for 0 < x < 15; e < 10^(-7.13).
+         *
+         * Note: Beta cannot exceed 15 (hence Stopband cannot exceed 144dB = 24b).
+         *
+         * This speeds up overall computation by about 40% over using the else clause below,
+         * which requires sqrt and exp.
+         *
+         */
+
+        x *= x;
+        double num = Poly9(-0.13544938430e9, -0.33153754512e8,
+                -0.19406631946e7, -0.48058318783e5,
+                -0.63269783360e3, -0.49520779070e1,
+                -0.24970910370e-1, -0.74741159550e-4,
+                -0.18257612460e-6, x);
+        double y = x - 225.; // reflection around 15 (squared)
+        double den = Poly4(-0.34598737196e8, 0.23852643181e6,
+                -0.70699387620e3, 0.10000000000e1, y);
+        return num / den;
+
+#if IO_EXTENDED_BETA
+        /* Table 42 for x > 15; e < 10^(-8.11).
+         * This is used for Beta>15, but is disabled here as
+         * we never use Beta that high.
+         *
+         * NOTE: This should be enabled only for x > 15.
+         */
+
+        double y = 1./x;
+        double z = y - (1./15);
+        double num = Poly2(0.415079861746e1, -0.5149092496e1, z);
+        double den = Poly3(0.103150763823e2, -0.14181687413e2,
+                0.1000000000e1, z);
+        return exp(x) * sqrt(y) * num / den;
+#endif
+    } else {
+        /*
+         * NOT USED, but reference for large Beta.
+         *
+         * Abramowitz and Stegun asymptotic formula.
+         * works for x > 3.75.
+         */
+        double y = 1./x;
+        return exp(x) * sqrt(y) *
+                // note: reciprocal squareroot may be easier!
+                // http://en.wikipedia.org/wiki/Fast_inverse_square_root
+                Poly9(I0ATerm<0>::value, I0ATerm<1>::value,
+                        I0ATerm<2>::value, I0ATerm<3>::value,
+                        I0ATerm<4>::value, I0ATerm<5>::value,
+                        I0ATerm<6>::value, I0ATerm<7>::value,
+                        I0ATerm<8>::value, y); // (... e) < 1.9e-7
+    }
+}
+
+/*
+ * calculates the transition bandwidth for a Kaiser filter
+ *
+ * Formula 3.2.8, Vaidyanathan, _Multirate Systems and Filter Banks_, p. 48
+ * Formula 7.76, Oppenheim and Schafer, _Discrete-time Signal Processing, 3e_, p. 542
+ *
+ * @param halfNumCoef is half the number of coefficients per filter phase.
+ *
+ * @param stopBandAtten is the stop band attenuation desired.
+ *
+ * @return the transition bandwidth in normalized frequency (0 <= f <= 0.5)
+ */
+static inline double firKaiserTbw(int halfNumCoef, double stopBandAtten) {
+    return (stopBandAtten - 7.95)/((2.*14.36)*halfNumCoef);
+}
+
+/*
+ * calculates the fir transfer response of the overall polyphase filter at w.
+ *
+ * Calculates the DTFT transfer coefficient H(w) for 0 <= w <= PI, utilizing the
+ * fact that h[n] is symmetric (cosines only, no complex arithmetic).
+ *
+ * We use Goertzel's algorithm to accelerate the computation to essentially
+ * a single multiply and 2 adds per filter coefficient h[].
+ *
+ * Be careful be careful to consider that h[n] is the overall polyphase filter,
+ * with L phases, so rescaling H(w)/L is probably what you expect for "unity gain",
+ * as you only use one of the polyphases at a time.
+ */
+template <typename T>
+static inline double firTransfer(const T* coef, int L, int halfNumCoef, double w) {
+    double accum = static_cast<double>(coef[0])*0.5;  // "center coefficient" from first bank
+    coef += halfNumCoef;    // skip first filterbank (picked up by the last filterbank).
+#if SLOW_FIRTRANSFER
+    /* Original code for reference.  This is equivalent to the code below, but slower. */
+    for (int i=1 ; i<=L ; ++i) {
+        for (int j=0, ix=i ; j<halfNumCoef ; ++j, ix+=L) {
+            accum += cos(ix*w)*static_cast<double>(*coef++);
+        }
+    }
+#else
+    /*
+     * Our overall filter is stored striped by polyphases, not a contiguous h[n].
+     * We could fetch coefficients in a non-contiguous fashion
+     * but that will not scale to vector processing.
+     *
+     * We apply Goertzel's algorithm directly to each polyphase filter bank instead of
+     * using cosine generation/multiplication, thereby saving one multiply per inner loop.
+     *
+     * See: http://en.wikipedia.org/wiki/Goertzel_algorithm
+     * Also: Oppenheim and Schafer, _Discrete Time Signal Processing, 3e_, p. 720.
+     *
+     * We use the basic recursion to incorporate the cosine steps into real sequence x[n]:
+     * s[n] = x[n] + (2cosw)*s[n-1] + s[n-2]
+     *
+     * y[n] = s[n] - e^(iw)s[n-1]
+     *      = sum_{k=-\infty}^{n} x[k]e^(-iw(n-k))
+     *      = e^(-iwn) sum_{k=0}^{n} x[k]e^(iwk)
+     *
+     * The summation contains the frequency steps we want multiplied by the source
+     * (similar to a DTFT).
+     *
+     * Using symmetry, and just the real part (be careful, this must happen
+     * after any internal complex multiplications), the polyphase filterbank
+     * transfer function is:
+     *
+     * Hpp[n, w, w_0] = sum_{k=0}^{n} x[k] * cos(wk + w_0)
+     *                = Re{ e^(iwn + iw_0) y[n]}
+     *                = cos(wn+w_0) * s[n] - cos(w(n+1)+w_0) * s[n-1]
+     *
+     * using the fact that s[n] of real x[n] is real.
+     *
+     */
+    double dcos = 2. * cos(L*w);
+    int start = ((halfNumCoef)*L + 1);
+    SineGen cc((start - L) * w, w, true); // cosine
+    SineGen cp(start * w, w, true); // cosine
+    for (int i=1 ; i<=L ; ++i) {
+        double sc = 0;
+        double sp = 0;
+        for (int j=0 ; j<halfNumCoef ; ++j) {
+            double tmp = sc;
+            sc  = static_cast<double>(*coef++) + dcos*sc - sp;
+            sp = tmp;
+        }
+        // If we are awfully clever, we can apply Goertzel's algorithm
+        // again on the sc and sp sequences returned here.
+        accum += cc.valueAdvance() * sc - cp.valueAdvance() * sp;
+    }
+#endif
+    return accum*2.;
+}
+
+/*
+ * evaluates the minimum and maximum |H(f)| bound in a band region.
+ *
+ * This is usually done with equally spaced increments in the target band in question.
+ * The passband is often very small, and sampled that way. The stopband is often much
+ * larger.
+ *
+ * We use the fact that the overall polyphase filter has an additional bank at the end
+ * for interpolation; hence it is overspecified for the H(f) computation.  Thus the
+ * first polyphase is never actually checked, excepting its first term.
+ *
+ * In this code we use the firTransfer() evaluator above, which uses Goertzel's
+ * algorithm to calculate the transfer function at each point.
+ *
+ * TODO: An alternative with equal spacing is the FFT/DFT.  An alternative with unequal
+ * spacing is a chirp transform.
+ *
+ * @param coef is the designed polyphase filter banks
+ *
+ * @param L is the number of phases (for interpolation)
+ *
+ * @param halfNumCoef should be half the number of coefficients for a single
+ * polyphase.
+ *
+ * @param fstart is the normalized frequency start.
+ *
+ * @param fend is the normalized frequency end.
+ *
+ * @param steps is the number of steps to take (sampling) between frequency start and end
+ *
+ * @param firMin returns the minimum transfer |H(f)| found
+ *
+ * @param firMax returns the maximum transfer |H(f)| found
+ *
+ * 0 <= f <= 0.5.
+ * This is used to test passband and stopband performance.
+ */
+template <typename T>
+static void testFir(const T* coef, int L, int halfNumCoef,
+        double fstart, double fend, int steps, double &firMin, double &firMax) {
+    double wstart = fstart*(2.*M_PI);
+    double wend = fend*(2.*M_PI);
+    double wstep = (wend - wstart)/steps;
+    double fmax, fmin;
+    double trf = firTransfer(coef, L, halfNumCoef, wstart);
+    if (trf<0) {
+        trf = -trf;
+    }
+    fmin = fmax = trf;
+    wstart += wstep;
+    for (int i=1; i<steps; ++i) {
+        trf = firTransfer(coef, L, halfNumCoef, wstart);
+        if (trf<0) {
+            trf = -trf;
+        }
+        if (trf>fmax) {
+            fmax = trf;
+        }
+        else if (trf<fmin) {
+            fmin = trf;
+        }
+        wstart += wstep;
+    }
+    // renormalize - this is only needed for integer filter types
+    double norm = 1./((1ULL<<(sizeof(T)*8-1))*L);
+
+    firMin = fmin * norm;
+    firMax = fmax * norm;
+}
+
+/*
+ * evaluates the |H(f)| lowpass band characteristics.
+ *
+ * This function tests the lowpass characteristics for the overall polyphase filter,
+ * and is used to verify the design.  For this case, fp should be set to the
+ * passband normalized frequency from 0 to 0.5 for the overall filter (thus it
+ * is the designed polyphase bank value / L).  Likewise for fs.
+ *
+ * @param coef is the designed polyphase filter banks
+ *
+ * @param L is the number of phases (for interpolation)
+ *
+ * @param halfNumCoef should be half the number of coefficients for a single
+ * polyphase.
+ *
+ * @param fp is the passband normalized frequency, 0 < fp < fs < 0.5.
+ *
+ * @param fs is the stopband normalized frequency, 0 < fp < fs < 0.5.
+ *
+ * @param passSteps is the number of passband sampling steps.
+ *
+ * @param stopSteps is the number of stopband sampling steps.
+ *
+ * @param passMin is the minimum value in the passband
+ *
+ * @param passMax is the maximum value in the passband (useful for scaling).  This should
+ * be less than 1., to avoid sine wave test overflow.
+ *
+ * @param passRipple is the passband ripple.  Typically this should be less than 0.1 for
+ * an audio filter.  Generally speaker/headphone device characteristics will dominate
+ * the passband term.
+ *
+ * @param stopMax is the maximum value in the stopband.
+ *
+ * @param stopRipple is the stopband ripple, also known as stopband attenuation.
+ * Typically this should be greater than ~80dB for low quality, and greater than
+ * ~100dB for full 16b quality, otherwise aliasing may become noticeable.
+ *
+ */
+template <typename T>
+static void testFir(const T* coef, int L, int halfNumCoef,
+        double fp, double fs, int passSteps, int stopSteps,
+        double &passMin, double &passMax, double &passRipple,
+        double &stopMax, double &stopRipple) {
+    double fmin, fmax;
+    testFir(coef, L, halfNumCoef, 0., fp, passSteps, fmin, fmax);
+    double d1 = (fmax - fmin)/2.;
+    passMin = fmin;
+    passMax = fmax;
+    passRipple = -20.*log10(1. - d1); // passband ripple
+    testFir(coef, L, halfNumCoef, fs, 0.5, stopSteps, fmin, fmax);
+    // fmin is really not important for the stopband.
+    stopMax = fmax;
+    stopRipple = -20.*log10(fmax); // stopband ripple/attenuation
+}
+
+/*
+ * Calculates the overall polyphase filter based on a windowed sinc function.
+ *
+ * The windowed sinc is an odd length symmetric filter of exactly L*halfNumCoef*2+1
+ * taps for the entire kernel.  This is then decomposed into L+1 polyphase filterbanks.
+ * The last filterbank is used for interpolation purposes (and is mostly composed
+ * of the first bank shifted by one sample), and is unnecessary if one does
+ * not do interpolation.
+ *
+ * We use the last filterbank for some transfer function calculation purposes,
+ * so it needs to be generated anyways.
+ *
+ * @param coef is the caller allocated space for coefficients.  This should be
+ * exactly (L+1)*halfNumCoef in size.
+ *
+ * @param L is the number of phases (for interpolation)
+ *
+ * @param halfNumCoef should be half the number of coefficients for a single
+ * polyphase.
+ *
+ * @param stopBandAtten is the stopband value, should be >50dB.
+ *
+ * @param fcr is cutoff frequency/sampling rate (<0.5).  At this point, the energy
+ * should be 6dB less. (fcr is where the amplitude drops by half).  Use the
+ * firKaiserTbw() to calculate the transition bandwidth.  fcr is the midpoint
+ * between the stop band and the pass band (fstop+fpass)/2.
+ *
+ * @param atten is the attenuation (generally slightly less than 1).
+ */
+
+template <typename T>
+static inline void firKaiserGen(T* coef, int L, int halfNumCoef,
+        double stopBandAtten, double fcr, double atten) {
+    //
+    // Formula 3.2.5, 3.2.7, Vaidyanathan, _Multirate Systems and Filter Banks_, p. 48
+    // Formula 7.75, Oppenheim and Schafer, _Discrete-time Signal Processing, 3e_, p. 542
+    //
+    // See also: http://melodi.ee.washington.edu/courses/ee518/notes/lec17.pdf
+    //
+    // Kaiser window and beta parameter
+    //
+    //         | 0.1102*(A - 8.7)                         A > 50
+    //  beta = | 0.5842*(A - 21)^0.4 + 0.07886*(A - 21)   21 <= A <= 50
+    //         | 0.                                       A < 21
+    //
+    // with A is the desired stop-band attenuation in dBFS
+    //
+    //    30 dB    2.210
+    //    40 dB    3.384
+    //    50 dB    4.538
+    //    60 dB    5.658
+    //    70 dB    6.764
+    //    80 dB    7.865
+    //    90 dB    8.960
+    //   100 dB   10.056
+
+    const int N = L * halfNumCoef; // non-negative half
+    const double beta = 0.1102 * (stopBandAtten - 8.7); // >= 50dB always
+    const double xstep = (2. * M_PI) * fcr / L;
+    const double xfrac = 1. / N;
+    const double yscale = atten * L / (I0(beta) * M_PI);
+
+    // We use sine generators, which computes sines on regular step intervals.
+    // This speeds up overall computation about 40% from computing the sine directly.
+
+    SineGenGen sgg(0., xstep, L*xstep); // generates sine generators (one per polyphase)
+
+    for (int i=0 ; i<=L ; ++i) { // generate an extra set of coefs for interpolation
+
+        // computation for a single polyphase of the overall filter.
+        SineGen sg = sgg.valueAdvance(); // current sine generator for "j" inner loop.
+        double err = 0; // for noise shaping on int16_t coefficients (over each polyphase)
+
+        for (int j=0, ix=i ; j<halfNumCoef ; ++j, ix+=L) {
+            double y;
+            if (CC_LIKELY(ix)) {
+                double x = static_cast<double>(ix);
+
+                // sine generator: sg.valueAdvance() returns sin(ix*xstep);
+                y = I0(beta * sqrt(1.0 - sqr(x * xfrac))) * yscale * sg.valueAdvance() / x;
+            } else {
+                y = 2. * atten * fcr; // center of filter, sinc(0) = 1.
+                sg.advance();
+            }
+
+            // (caution!) float version does not need rounding
+            if (is_same<T, int16_t>::value) { // int16_t needs noise shaping
+                *coef++ = static_cast<T>(toint(y, 1ULL<<(sizeof(T)*8-1), err));
+            } else {
+                *coef++ = static_cast<T>(toint(y, 1ULL<<(sizeof(T)*8-1)));
+            }
+        }
+    }
+}
+
+}; // namespace android
+
+#endif /*ANDROID_AUDIO_RESAMPLER_FIR_GEN_H*/

services/audioflinger/AudioResamplerFirOps.h

diff --git a/services/audioflinger/AudioResamplerFirOps.h b/services/audioflinger/AudioResamplerFirOps.h
new file mode 100644
index 0000000..bf2163f
--- /dev/null
+++ b/services/audioflinger/AudioResamplerFirOps.h
@@ -0,0 +1,163 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIO_RESAMPLER_FIR_OPS_H
+#define ANDROID_AUDIO_RESAMPLER_FIR_OPS_H
+
+namespace android {
+
+#if defined(__arm__) && !defined(__thumb__)
+#define USE_INLINE_ASSEMBLY (true)
+#else
+#define USE_INLINE_ASSEMBLY (false)
+#endif
+
+#if USE_INLINE_ASSEMBLY && defined(__ARM_NEON__)
+#define USE_NEON (true)
+#include <arm_neon.h>
+#else
+#define USE_NEON (false)
+#endif
+
+template<typename T, typename U>
+struct is_same
+{
+    static const bool value = false;
+};
+
+template<typename T>
+struct is_same<T, T>  // partial specialization
+{
+    static const bool value = true;
+};
+
+static inline
+int32_t mulRL(int left, int32_t in, uint32_t vRL)
+{
+#if USE_INLINE_ASSEMBLY
+    int32_t out;
+    if (left) {
+        asm( "smultb %[out], %[in], %[vRL] \n"
+             : [out]"=r"(out)
+             : [in]"%r"(in), [vRL]"r"(vRL)
+             : );
+    } else {
+        asm( "smultt %[out], %[in], %[vRL] \n"
+             : [out]"=r"(out)
+             : [in]"%r"(in), [vRL]"r"(vRL)
+             : );
+    }
+    return out;
+#else
+    int16_t v = left ? static_cast<int16_t>(vRL) : static_cast<int16_t>(vRL>>16);
+    return static_cast<int32_t>((static_cast<int64_t>(in) * v) >> 16);
+#endif
+}
+
+static inline
+int32_t mulAdd(int16_t in, int16_t v, int32_t a)
+{
+#if USE_INLINE_ASSEMBLY
+    int32_t out;
+    asm( "smlabb %[out], %[v], %[in], %[a] \n"
+         : [out]"=r"(out)
+         : [in]"%r"(in), [v]"r"(v), [a]"r"(a)
+         : );
+    return out;
+#else
+    return a + v * in;
+#endif
+}
+
+static inline
+int32_t mulAdd(int16_t in, int32_t v, int32_t a)
+{
+#if USE_INLINE_ASSEMBLY
+    int32_t out;
+    asm( "smlawb %[out], %[v], %[in], %[a] \n"
+         : [out]"=r"(out)
+         : [in]"%r"(in), [v]"r"(v), [a]"r"(a)
+         : );
+    return out;
+#else
+    return a + static_cast<int32_t>((static_cast<int64_t>(v) * in) >> 16);
+#endif
+}
+
+static inline
+int32_t mulAdd(int32_t in, int32_t v, int32_t a)
+{
+#if USE_INLINE_ASSEMBLY
+    int32_t out;
+    asm( "smmla %[out], %[v], %[in], %[a] \n"
+         : [out]"=r"(out)
+         : [in]"%r"(in), [v]"r"(v), [a]"r"(a)
+         : );
+    return out;
+#else
+    return a + static_cast<int32_t>((static_cast<int64_t>(v) * in) >> 32);
+#endif
+}
+
+static inline
+int32_t mulAddRL(int left, uint32_t inRL, int16_t v, int32_t a)
+{
+#if USE_INLINE_ASSEMBLY
+    int32_t out;
+    if (left) {
+        asm( "smlabb %[out], %[v], %[inRL], %[a] \n"
+             : [out]"=r"(out)
+             : [inRL]"%r"(inRL), [v]"r"(v), [a]"r"(a)
+             : );
+    } else {
+        asm( "smlabt %[out], %[v], %[inRL], %[a] \n"
+             : [out]"=r"(out)
+             : [inRL]"%r"(inRL), [v]"r"(v), [a]"r"(a)
+             : );
+    }
+    return out;
+#else
+    int16_t s = left ? static_cast<int16_t>(inRL) : static_cast<int16_t>(inRL>>16);
+    return a + v * s;
+#endif
+}
+
+static inline
+int32_t mulAddRL(int left, uint32_t inRL, int32_t v, int32_t a)
+{
+#if USE_INLINE_ASSEMBLY
+    int32_t out;
+    if (left) {
+        asm( "smlawb %[out], %[v], %[inRL], %[a] \n"
+             : [out]"=r"(out)
+             : [inRL]"%r"(inRL), [v]"r"(v), [a]"r"(a)
+             : );
+    } else {
+        asm( "smlawt %[out], %[v], %[inRL], %[a] \n"
+             : [out]"=r"(out)
+             : [inRL]"%r"(inRL), [v]"r"(v), [a]"r"(a)
+             : );
+    }
+    return out;
+#else
+    int16_t s = left ? static_cast<int16_t>(inRL) : static_cast<int16_t>(inRL>>16);
+    return a + static_cast<int32_t>((static_cast<int64_t>(v) * s) >> 16);
+#endif
+}
+
+}; // namespace android
+
+#endif /*ANDROID_AUDIO_RESAMPLER_FIR_OPS_H*/

services/audioflinger/AudioResamplerFirProcess.h

diff --git a/services/audioflinger/AudioResamplerFirProcess.h b/services/audioflinger/AudioResamplerFirProcess.h
new file mode 100644
index 0000000..38e387c
--- /dev/null
+++ b/services/audioflinger/AudioResamplerFirProcess.h
@@ -0,0 +1,256 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_H
+#define ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_H
+
+namespace android {
+
+// depends on AudioResamplerFirOps.h
+
+template<int CHANNELS, typename TC>
+static inline
+void mac(
+        int32_t& l, int32_t& r,
+        const TC coef,
+        const int16_t* samples)
+{
+    if (CHANNELS == 2) {
+        uint32_t rl = *reinterpret_cast<const uint32_t*>(samples);
+        l = mulAddRL(1, rl, coef, l);
+        r = mulAddRL(0, rl, coef, r);
+    } else {
+        r = l = mulAdd(samples[0], coef, l);
+    }
+}
+
+template<int CHANNELS, typename TC>
+static inline
+void interpolate(
+        int32_t& l, int32_t& r,
+        const TC coef_0, const TC coef_1,
+        const int16_t lerp, const int16_t* samples)
+{
+    TC sinc;
+
+    if (is_same<TC, int16_t>::value) {
+        sinc = (lerp * ((coef_1-coef_0)<<1)>>16) + coef_0;
+    } else {
+        sinc = mulAdd(lerp, (coef_1-coef_0)<<1, coef_0);
+    }
+    if (CHANNELS == 2) {
+        uint32_t rl = *reinterpret_cast<const uint32_t*>(samples);
+        l = mulAddRL(1, rl, sinc, l);
+        r = mulAddRL(0, rl, sinc, r);
+    } else {
+        r = l = mulAdd(samples[0], sinc, l);
+    }
+}
+
+/*
+ * Calculates a single output sample (two stereo frames).
+ *
+ * This function computes both the positive half FIR dot product and
+ * the negative half FIR dot product, accumulates, and then applies the volume.
+ *
+ * This is a locked phase filter (it does not compute the interpolation).
+ *
+ * Use fir() to compute the proper coefficient pointers for a polyphase
+ * filter bank.
+ */
+
+template <int CHANNELS, int STRIDE, typename TC>
+static inline
+void ProcessL(int32_t* const out,
+        int count,
+        const TC* coefsP,
+        const TC* coefsN,
+        const int16_t* sP,
+        const int16_t* sN,
+        const int32_t* const volumeLR)
+{
+    int32_t l = 0;
+    int32_t r = 0;
+    do {
+        mac<CHANNELS>(l, r, *coefsP++, sP);
+        sP -= CHANNELS;
+        mac<CHANNELS>(l, r, *coefsN++, sN);
+        sN += CHANNELS;
+    } while (--count > 0);
+    out[0] += 2 * mulRL(0, l, volumeLR[0]); // Note: only use top 16b
+    out[1] += 2 * mulRL(0, r, volumeLR[1]); // Note: only use top 16b
+}
+
+/*
+ * Calculates a single output sample (two stereo frames) interpolating phase.
+ *
+ * This function computes both the positive half FIR dot product and
+ * the negative half FIR dot product, accumulates, and then applies the volume.
+ *
+ * This is an interpolated phase filter.
+ *
+ * Use fir() to compute the proper coefficient pointers for a polyphase
+ * filter bank.
+ */
+
+template <int CHANNELS, int STRIDE, typename TC>
+static inline
+void Process(int32_t* const out,
+        int count,
+        const TC* coefsP,
+        const TC* coefsN,
+        const TC* coefsP1,
+        const TC* coefsN1,
+        const int16_t* sP,
+        const int16_t* sN,
+        uint32_t lerpP,
+        const int32_t* const volumeLR)
+{
+    (void) coefsP1; // suppress unused parameter warning
+    (void) coefsN1;
+    if (sizeof(*coefsP)==4) {
+        lerpP >>= 16;   // ensure lerpP is 16b
+    }
+    int32_t l = 0;
+    int32_t r = 0;
+    for (size_t i = 0; i < count; ++i) {
+        interpolate<CHANNELS>(l, r, coefsP[0], coefsP[count], lerpP, sP);
+        coefsP++;
+        sP -= CHANNELS;
+        interpolate<CHANNELS>(l, r, coefsN[count], coefsN[0], lerpP, sN);
+        coefsN++;
+        sN += CHANNELS;
+    }
+    out[0] += 2 * mulRL(0, l, volumeLR[0]); // Note: only use top 16b
+    out[1] += 2 * mulRL(0, r, volumeLR[1]); // Note: only use top 16b
+}
+
+/*
+ * Calculates a single output sample (two stereo frames) from input sample pointer.
+ *
+ * This sets up the params for the accelerated Process() and ProcessL()
+ * functions to do the appropriate dot products.
+ *
+ * @param out should point to the output buffer with at least enough space for 2 output frames.
+ *
+ * @param phase is the fractional distance between input samples for interpolation:
+ * phase >= 0  && phase < phaseWrapLimit.  It can be thought of as a rational fraction
+ * of phase/phaseWrapLimit.
+ *
+ * @param phaseWrapLimit is #polyphases<<coefShift, where #polyphases is the number of polyphases
+ * in the polyphase filter. Likewise, #polyphases can be obtained as (phaseWrapLimit>>coefShift).
+ *
+ * @param coefShift gives the bit alignment of the polyphase index in the phase parameter.
+ *
+ * @param halfNumCoefs is the half the number of coefficients per polyphase filter. Since the
+ * overall filterbank is odd-length symmetric, only halfNumCoefs need be stored.
+ *
+ * @param coefs is the polyphase filter bank, starting at from polyphase index 0, and ranging to
+ * and including the #polyphases.  Each polyphase of the filter has half-length halfNumCoefs
+ * (due to symmetry).  The total size of the filter bank in coefficients is
+ * (#polyphases+1)*halfNumCoefs.
+ *
+ * The filter bank coefs should be aligned to a minimum of 16 bytes (preferrably to cache line).
+ *
+ * The coefs should be attenuated (to compensate for passband ripple)
+ * if storing back into the native format.
+ *
+ * @param samples are unaligned input samples.  The position is in the "middle" of the
+ * sample array with respect to the FIR filter:
+ * the negative half of the filter is dot product from samples+1 to samples+halfNumCoefs;
+ * the positive half of the filter is dot product from samples to samples-halfNumCoefs+1.
+ *
+ * @param volumeLR is a pointer to an array of two 32 bit volume values, one per stereo channel,
+ * expressed as a S32 integer.  A negative value inverts the channel 180 degrees.
+ * The pointer volumeLR should be aligned to a minimum of 8 bytes.
+ * A typical value for volume is 0x1000 to align to a unity gain output of 20.12.
+ *
+ * In between calls to filterCoefficient, the phase is incremented by phaseIncrement, where
+ * phaseIncrement is calculated as inputSampling * phaseWrapLimit / outputSampling.
+ *
+ * The filter polyphase index is given by indexP = phase >> coefShift. Due to
+ * odd length symmetric filter, the polyphase index of the negative half depends on
+ * whether interpolation is used.
+ *
+ * The fractional siting between the polyphase indices is given by the bits below coefShift:
+ *
+ * lerpP = phase << 32 - coefShift >> 1;  // for 32 bit unsigned phase multiply
+ * lerpP = phase << 32 - coefShift >> 17; // for 16 bit unsigned phase multiply
+ *
+ * For integer types, this is expressed as:
+ *
+ * lerpP = phase << sizeof(phase)*8 - coefShift
+ *              >> (sizeof(phase)-sizeof(*coefs))*8 + 1;
+ *
+ */
+
+template<int CHANNELS, bool LOCKED, int STRIDE, typename TC>
+static inline
+void fir(int32_t* const out,
+        const uint32_t phase, const uint32_t phaseWrapLimit,
+        const int coefShift, const int halfNumCoefs, const TC* const coefs,
+        const int16_t* const samples, const int32_t* const volumeLR)
+{
+    // NOTE: be very careful when modifying the code here. register
+    // pressure is very high and a small change might cause the compiler
+    // to generate far less efficient code.
+    // Always sanity check the result with objdump or test-resample.
+
+    if (LOCKED) {
+        // locked polyphase (no interpolation)
+        // Compute the polyphase filter index on the positive and negative side.
+        uint32_t indexP = phase >> coefShift;
+        uint32_t indexN = (phaseWrapLimit - phase) >> coefShift;
+        const TC* coefsP = coefs + indexP*halfNumCoefs;
+        const TC* coefsN = coefs + indexN*halfNumCoefs;
+        const int16_t* sP = samples;
+        const int16_t* sN = samples + CHANNELS;
+
+        // dot product filter.
+        ProcessL<CHANNELS, STRIDE>(out,
+                halfNumCoefs, coefsP, coefsN, sP, sN, volumeLR);
+    } else {
+        // interpolated polyphase
+        // Compute the polyphase filter index on the positive and negative side.
+        uint32_t indexP = phase >> coefShift;
+        uint32_t indexN = (phaseWrapLimit - phase - 1) >> coefShift; // one's complement.
+        const TC* coefsP = coefs + indexP*halfNumCoefs;
+        const TC* coefsN = coefs + indexN*halfNumCoefs;
+        const TC* coefsP1 = coefsP + halfNumCoefs;
+        const TC* coefsN1 = coefsN + halfNumCoefs;
+        const int16_t* sP = samples;
+        const int16_t* sN = samples + CHANNELS;
+
+        // Interpolation fraction lerpP derived by shifting all the way up and down
+        // to clear the appropriate bits and align to the appropriate level
+        // for the integer multiply.  The constants should resolve in compile time.
+        //
+        // The interpolated filter coefficient is derived as follows for the pos/neg half:
+        //
+        // interpolated[P] = index[P]*lerpP + index[P+1]*(1-lerpP)
+        // interpolated[N] = index[N+1]*lerpP + index[N]*(1-lerpP)
+        uint32_t lerpP = phase << (sizeof(phase)*8 - coefShift)
+                >> ((sizeof(phase)-sizeof(*coefs))*8 + 1);
+
+        // on-the-fly interpolated dot product filter
+        Process<CHANNELS, STRIDE>(out,
+                halfNumCoefs, coefsP, coefsN, coefsP1, coefsN1, sP, sN, lerpP, volumeLR);
+    }
+}
+
+}; // namespace android
+
+#endif /*ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_H*/

services/audioflinger/AudioResamplerFirProcessNeon.h

diff --git a/services/audioflinger/AudioResamplerFirProcessNeon.h b/services/audioflinger/AudioResamplerFirProcessNeon.h
new file mode 100644
index 0000000..f311cef
--- /dev/null
+++ b/services/audioflinger/AudioResamplerFirProcessNeon.h
@@ -0,0 +1,1149 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_NEON_H
+#define ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_NEON_H
+
+namespace android {
+
+// depends on AudioResamplerFirOps.h, AudioResamplerFirProcess.h
+
+#if USE_NEON
+//
+// NEON specializations are enabled for Process() and ProcessL()
+//
+// TODO: Stride 16 and Stride 8 can be combined with one pass stride 8 (if necessary)
+// and looping stride 16 (or vice versa). This has some polyphase coef data alignment
+// issues with S16 coefs. Consider this later.
+
+// Macros to save a mono/stereo accumulator sample in q0 (and q4) as stereo out.
+#define ASSEMBLY_ACCUMULATE_MONO \
+        "vld1.s32       {d2}, [%[vLR]:64]        \n"/* (1) load volumes */\
+        "vld1.s32       {d3}, %[out]             \n"/* (2) unaligned load the output */\
+        "vpadd.s32      d0, d0, d1               \n"/* (1) add all 4 partial sums */\
+        "vpadd.s32      d0, d0, d0               \n"/* (1+4d) and replicate L/R */\
+        "vqrdmulh.s32   d0, d0, d2               \n"/* (2+3d) apply volume */\
+        "vqadd.s32      d3, d3, d0               \n"/* (1+4d) accumulate result (saturating) */\
+        "vst1.s32       {d3}, %[out]             \n"/* (2+2d) store result */
+
+#define ASSEMBLY_ACCUMULATE_STEREO \
+        "vld1.s32       {d2}, [%[vLR]:64]        \n"/* (1) load volumes*/\
+        "vld1.s32       {d3}, %[out]             \n"/* (2) unaligned load the output*/\
+        "vpadd.s32      d0, d0, d1               \n"/* (1) add all 4 partial sums from q0*/\
+        "vpadd.s32      d8, d8, d9               \n"/* (1) add all 4 partial sums from q4*/\
+        "vpadd.s32      d0, d0, d8               \n"/* (1+4d) combine into L/R*/\
+        "vqrdmulh.s32   d0, d0, d2               \n"/* (2+3d) apply volume*/\
+        "vqadd.s32      d3, d3, d0               \n"/* (1+4d) accumulate result (saturating)*/\
+        "vst1.s32       {d3}, %[out]             \n"/* (2+2d)store result*/
+
+template <>
+inline void ProcessL<1, 16>(int32_t* const out,
+        int count,
+        const int16_t* coefsP,
+        const int16_t* coefsN,
+        const int16_t* sP,
+        const int16_t* sN,
+        const int32_t* const volumeLR)
+{
+    const int CHANNELS = 1; // template specialization does not preserve params
+    const int STRIDE = 16;
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    asm (
+        "veor           q0, q0, q0               \n"// (0 - combines+) accumulator = 0
+
+        "1:                                      \n"
+
+        "vld1.16        {q2}, [%[sP]]            \n"// (2+0d) load 8 16-bits mono samples
+        "vld1.16        {q3}, [%[sN]]!           \n"// (2) load 8 16-bits mono samples
+        "vld1.16        {q8}, [%[coefsP0]:128]!  \n"// (1) load 8 16-bits coefs
+        "vld1.16        {q10}, [%[coefsN0]:128]! \n"// (1) load 8 16-bits coefs
+
+        "vrev64.16      q2, q2                   \n"// (1) reverse s3, s2, s1, s0, s7, s6, s5, s4
+
+        // reordering the vmal to do d6, d7 before d4, d5 is slower(?)
+        "vmlal.s16      q0, d4, d17              \n"// (1+0d) multiply (reversed)samples by coef
+        "vmlal.s16      q0, d5, d16              \n"// (1) multiply (reversed)samples by coef
+        "vmlal.s16      q0, d6, d20              \n"// (1) multiply neg samples
+        "vmlal.s16      q0, d7, d21              \n"// (1) multiply neg samples
+
+        // moving these ARM instructions before neon above seems to be slower
+        "subs           %[count], %[count], #8   \n"// (1) update loop counter
+        "sub            %[sP], %[sP], #16        \n"// (0) move pointer to next set of samples
+
+        // sP used after branch (warning)
+        "bne            1b                       \n"// loop
+
+         ASSEMBLY_ACCUMULATE_MONO
+
+        : [out]     "=Uv" (out[0]),
+          [count]   "+r" (count),
+          [coefsP0] "+r" (coefsP),
+          [coefsN0] "+r" (coefsN),
+          [sP]      "+r" (sP),
+          [sN]      "+r" (sN)
+        : [vLR]     "r" (volumeLR)
+        : "cc", "memory",
+          "q0", "q1", "q2", "q3",
+          "q8", "q10"
+    );
+}
+
+template <>
+inline void ProcessL<2, 16>(int32_t* const out,
+        int count,
+        const int16_t* coefsP,
+        const int16_t* coefsN,
+        const int16_t* sP,
+        const int16_t* sN,
+        const int32_t* const volumeLR)
+{
+    const int CHANNELS = 2; // template specialization does not preserve params
+    const int STRIDE = 16;
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    asm (
+        "veor           q0, q0, q0               \n"// (1) acc_L = 0
+        "veor           q4, q4, q4               \n"// (0 combines+) acc_R = 0
+
+        "1:                                      \n"
+
+        "vld2.16        {q2, q3}, [%[sP]]        \n"// (3+0d) load 8 16-bits stereo samples
+        "vld2.16        {q5, q6}, [%[sN]]!       \n"// (3) load 8 16-bits stereo samples
+        "vld1.16        {q8}, [%[coefsP0]:128]!  \n"// (1) load 8 16-bits coefs
+        "vld1.16        {q10}, [%[coefsN0]:128]! \n"// (1) load 8 16-bits coefs
+
+        "vrev64.16      q2, q2                   \n"// (1) reverse 8 frames of the left positive
+        "vrev64.16      q3, q3                   \n"// (0 combines+) reverse right positive
+
+        "vmlal.s16      q0, d4, d17              \n"// (1) multiply (reversed) samples left
+        "vmlal.s16      q0, d5, d16              \n"// (1) multiply (reversed) samples left
+        "vmlal.s16      q4, d6, d17              \n"// (1) multiply (reversed) samples right
+        "vmlal.s16      q4, d7, d16              \n"// (1) multiply (reversed) samples right
+        "vmlal.s16      q0, d10, d20             \n"// (1) multiply samples left
+        "vmlal.s16      q0, d11, d21             \n"// (1) multiply samples left
+        "vmlal.s16      q4, d12, d20             \n"// (1) multiply samples right
+        "vmlal.s16      q4, d13, d21             \n"// (1) multiply samples right
+
+        // moving these ARM before neon seems to be slower
+        "subs           %[count], %[count], #8   \n"// (1) update loop counter
+        "sub            %[sP], %[sP], #32        \n"// (0) move pointer to next set of samples
+
+        // sP used after branch (warning)
+        "bne            1b                       \n"// loop
+
+        ASSEMBLY_ACCUMULATE_STEREO
+
+        : [out] "=Uv" (out[0]),
+          [count] "+r" (count),
+          [coefsP0] "+r" (coefsP),
+          [coefsN0] "+r" (coefsN),
+          [sP] "+r" (sP),
+          [sN] "+r" (sN)
+        : [vLR] "r" (volumeLR)
+        : "cc", "memory",
+          "q0", "q1", "q2", "q3",
+          "q4", "q5", "q6",
+          "q8", "q10"
+     );
+}
+
+template <>
+inline void Process<1, 16>(int32_t* const out,
+        int count,
+        const int16_t* coefsP,
+        const int16_t* coefsN,
+        const int16_t* coefsP1,
+        const int16_t* coefsN1,
+        const int16_t* sP,
+        const int16_t* sN,
+        uint32_t lerpP,
+        const int32_t* const volumeLR)
+{
+    const int CHANNELS = 1; // template specialization does not preserve params
+    const int STRIDE = 16;
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    asm (
+        "vmov.32        d2[0], %[lerpP]          \n"// load the positive phase S32 Q15
+        "veor           q0, q0, q0               \n"// (0 - combines+) accumulator = 0
+
+        "1:                                      \n"
+
+        "vld1.16        {q2}, [%[sP]]            \n"// (2+0d) load 8 16-bits mono samples
+        "vld1.16        {q3}, [%[sN]]!           \n"// (2) load 8 16-bits mono samples
+        "vld1.16        {q8}, [%[coefsP0]:128]!  \n"// (1) load 8 16-bits coefs
+        "vld1.16        {q9}, [%[coefsP1]:128]!  \n"// (1) load 8 16-bits coefs for interpolation
+        "vld1.16        {q10}, [%[coefsN1]:128]! \n"// (1) load 8 16-bits coefs
+        "vld1.16        {q11}, [%[coefsN0]:128]! \n"// (1) load 8 16-bits coefs for interpolation
+
+        "vsub.s16       q9, q9, q8               \n"// (1) interpolate (step1) 1st set of coefs
+        "vsub.s16       q11, q11, q10            \n"// (1) interpolate (step1) 2nd set of coets
+
+        "vqrdmulh.s16   q9, q9, d2[0]            \n"// (2) interpolate (step2) 1st set of coefs
+        "vqrdmulh.s16   q11, q11, d2[0]          \n"// (2) interpolate (step2) 2nd set of coefs
+
+        "vrev64.16      q2, q2                   \n"// (1) reverse s3, s2, s1, s0, s7, s6, s5, s4
+
+        "vadd.s16       q8, q8, q9               \n"// (1+2d) interpolate (step3) 1st set
+        "vadd.s16       q10, q10, q11            \n"// (1+1d) interpolate (step3) 2nd set
+
+        // reordering the vmal to do d6, d7 before d4, d5 is slower(?)
+        "vmlal.s16      q0, d4, d17              \n"// (1+0d) multiply reversed samples by coef
+        "vmlal.s16      q0, d5, d16              \n"// (1) multiply reversed samples by coef
+        "vmlal.s16      q0, d6, d20              \n"// (1) multiply neg samples
+        "vmlal.s16      q0, d7, d21              \n"// (1) multiply neg samples
+
+        // moving these ARM instructions before neon above seems to be slower
+        "subs           %[count], %[count], #8   \n"// (1) update loop counter
+        "sub            %[sP], %[sP], #16        \n"// (0) move pointer to next set of samples
+
+        // sP used after branch (warning)
+        "bne            1b                       \n"// loop
+
+        ASSEMBLY_ACCUMULATE_MONO
+
+        : [out]     "=Uv" (out[0]),
+          [count]   "+r" (count),
+          [coefsP0] "+r" (coefsP),
+          [coefsN0] "+r" (coefsN),
+          [coefsP1] "+r" (coefsP1),
+          [coefsN1] "+r" (coefsN1),
+          [sP]      "+r" (sP),
+          [sN]      "+r" (sN)
+        : [lerpP]   "r" (lerpP),
+          [vLR]     "r" (volumeLR)
+        : "cc", "memory",
+          "q0", "q1", "q2", "q3",
+          "q8", "q9", "q10", "q11"
+    );
+}
+
+template <>
+inline void Process<2, 16>(int32_t* const out,
+        int count,
+        const int16_t* coefsP,
+        const int16_t* coefsN,
+        const int16_t* coefsP1,
+        const int16_t* coefsN1,
+        const int16_t* sP,
+        const int16_t* sN,
+        uint32_t lerpP,
+        const int32_t* const volumeLR)
+{
+    const int CHANNELS = 2; // template specialization does not preserve params
+    const int STRIDE = 16;
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    asm (
+        "vmov.32        d2[0], %[lerpP]          \n"// load the positive phase
+        "veor           q0, q0, q0               \n"// (1) acc_L = 0
+        "veor           q4, q4, q4               \n"// (0 combines+) acc_R = 0
+
+        "1:                                      \n"
+
+        "vld2.16        {q2, q3}, [%[sP]]        \n"// (3+0d) load 8 16-bits stereo samples
+        "vld2.16        {q5, q6}, [%[sN]]!       \n"// (3) load 8 16-bits stereo samples
+        "vld1.16        {q8}, [%[coefsP0]:128]!  \n"// (1) load 8 16-bits coefs
+        "vld1.16        {q9}, [%[coefsP1]:128]!  \n"// (1) load 8 16-bits coefs for interpolation
+        "vld1.16        {q10}, [%[coefsN1]:128]! \n"// (1) load 8 16-bits coefs
+        "vld1.16        {q11}, [%[coefsN0]:128]! \n"// (1) load 8 16-bits coefs for interpolation
+
+        "vsub.s16       q9, q9, q8               \n"// (1) interpolate (step1) 1st set of coefs
+        "vsub.s16       q11, q11, q10            \n"// (1) interpolate (step1) 2nd set of coets
+
+        "vqrdmulh.s16   q9, q9, d2[0]            \n"// (2) interpolate (step2) 1st set of coefs
+        "vqrdmulh.s16   q11, q11, d2[0]          \n"// (2) interpolate (step2) 2nd set of coefs
+
+        "vrev64.16      q2, q2                   \n"// (1) reverse 8 frames of the left positive
+        "vrev64.16      q3, q3                   \n"// (1) reverse 8 frames of the right positive
+
+        "vadd.s16       q8, q8, q9               \n"// (1+1d) interpolate (step3) 1st set
+        "vadd.s16       q10, q10, q11            \n"// (1+1d) interpolate (step3) 2nd set
+
+        "vmlal.s16      q0, d4, d17              \n"// (1) multiply reversed samples left
+        "vmlal.s16      q0, d5, d16              \n"// (1) multiply reversed samples left
+        "vmlal.s16      q4, d6, d17              \n"// (1) multiply reversed samples right
+        "vmlal.s16      q4, d7, d16              \n"// (1) multiply reversed samples right
+        "vmlal.s16      q0, d10, d20             \n"// (1) multiply samples left
+        "vmlal.s16      q0, d11, d21             \n"// (1) multiply samples left
+        "vmlal.s16      q4, d12, d20             \n"// (1) multiply samples right
+        "vmlal.s16      q4, d13, d21             \n"// (1) multiply samples right
+
+        // moving these ARM before neon seems to be slower
+        "subs           %[count], %[count], #8   \n"// (1) update loop counter
+        "sub            %[sP], %[sP], #32        \n"// (0) move pointer to next set of samples
+
+        // sP used after branch (warning)
+        "bne            1b                       \n"// loop
+
+        ASSEMBLY_ACCUMULATE_STEREO
+
+        : [out] "=Uv" (out[0]),
+          [count] "+r" (count),
+          [coefsP0] "+r" (coefsP),
+          [coefsN0] "+r" (coefsN),
+          [coefsP1] "+r" (coefsP1),
+          [coefsN1] "+r" (coefsN1),
+          [sP] "+r" (sP),
+          [sN] "+r" (sN)
+        : [lerpP]   "r" (lerpP),
+          [vLR] "r" (volumeLR)
+        : "cc", "memory",
+          "q0", "q1", "q2", "q3",
+          "q4", "q5", "q6",
+          "q8", "q9", "q10", "q11"
+    );
+}
+
+template <>
+inline void ProcessL<1, 16>(int32_t* const out,
+        int count,
+        const int32_t* coefsP,
+        const int32_t* coefsN,
+        const int16_t* sP,
+        const int16_t* sN,
+        const int32_t* const volumeLR)
+{
+    const int CHANNELS = 1; // template specialization does not preserve params
+    const int STRIDE = 16;
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    asm (
+        "veor           q0, q0, q0                    \n"// result, initialize to 0
+
+        "1:                                           \n"
+
+        "vld1.16        {q2}, [%[sP]]                 \n"// load 8 16-bits mono samples
+        "vld1.16        {q3}, [%[sN]]!                \n"// load 8 16-bits mono samples
+        "vld1.32        {q8, q9}, [%[coefsP0]:128]!   \n"// load 8 32-bits coefs
+        "vld1.32        {q10, q11}, [%[coefsN0]:128]! \n"// load 8 32-bits coefs
+
+        "vrev64.16      q2, q2                        \n"// reverse 8 frames of the positive side
+
+        "vshll.s16      q12, d4, #15                  \n"// extend samples to 31 bits
+        "vshll.s16      q13, d5, #15                  \n"// extend samples to 31 bits
+
+        "vshll.s16      q14, d6, #15                  \n"// extend samples to 31 bits
+        "vshll.s16      q15, d7, #15                  \n"// extend samples to 31 bits
+
+        "vqrdmulh.s32   q12, q12, q9                  \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q13, q13, q8                  \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q14, q14, q10                 \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q15, q15, q11                 \n"// multiply samples by interpolated coef
+
+        "vadd.s32       q0, q0, q12                   \n"// accumulate result
+        "vadd.s32       q13, q13, q14                 \n"// accumulate result
+        "vadd.s32       q0, q0, q15                   \n"// accumulate result
+        "vadd.s32       q0, q0, q13                   \n"// accumulate result
+
+        "sub            %[sP], %[sP], #16             \n"// move pointer to next set of samples
+        "subs           %[count], %[count], #8        \n"// update loop counter
+
+        "bne            1b                            \n"// loop
+
+        ASSEMBLY_ACCUMULATE_MONO
+
+        : [out]     "=Uv" (out[0]),
+          [count]   "+r" (count),
+          [coefsP0] "+r" (coefsP),
+          [coefsN0] "+r" (coefsN),
+          [sP]      "+r" (sP),
+          [sN]      "+r" (sN)
+        : [vLR]     "r" (volumeLR)
+        : "cc", "memory",
+          "q0", "q1", "q2", "q3",
+          "q8", "q9", "q10", "q11",
+          "q12", "q13", "q14", "q15"
+    );
+}
+
+template <>
+inline void ProcessL<2, 16>(int32_t* const out,
+        int count,
+        const int32_t* coefsP,
+        const int32_t* coefsN,
+        const int16_t* sP,
+        const int16_t* sN,
+        const int32_t* const volumeLR)
+{
+    const int CHANNELS = 2; // template specialization does not preserve params
+    const int STRIDE = 16;
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    asm (
+        "veor           q0, q0, q0                    \n"// result, initialize to 0
+        "veor           q4, q4, q4                    \n"// result, initialize to 0
+
+        "1:                                           \n"
+
+        "vld2.16        {q2, q3}, [%[sP]]             \n"// load 4 16-bits stereo samples
+        "vld2.16        {q5, q6}, [%[sN]]!            \n"// load 4 16-bits stereo samples
+        "vld1.32        {q8, q9}, [%[coefsP0]:128]!   \n"// load 4 32-bits coefs
+        "vld1.32        {q10, q11}, [%[coefsN0]:128]! \n"// load 4 32-bits coefs
+
+        "vrev64.16      q2, q2                        \n"// reverse 8 frames of the positive side
+        "vrev64.16      q3, q3                        \n"// reverse 8 frames of the positive side
+
+        "vshll.s16      q12,  d4, #15                 \n"// extend samples to 31 bits
+        "vshll.s16      q13,  d5, #15                 \n"// extend samples to 31 bits
+
+        "vshll.s16      q14,  d10, #15                \n"// extend samples to 31 bits
+        "vshll.s16      q15,  d11, #15                \n"// extend samples to 31 bits
+
+        "vqrdmulh.s32   q12, q12, q9                  \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q13, q13, q8                  \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q14, q14, q10                 \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q15, q15, q11                 \n"// multiply samples by interpolated coef
+
+        "vadd.s32       q0, q0, q12                   \n"// accumulate result
+        "vadd.s32       q13, q13, q14                 \n"// accumulate result
+        "vadd.s32       q0, q0, q15                   \n"// (+1) accumulate result
+        "vadd.s32       q0, q0, q13                   \n"// (+1) accumulate result
+
+        "vshll.s16      q12,  d6, #15                 \n"// extend samples to 31 bits
+        "vshll.s16      q13,  d7, #15                 \n"// extend samples to 31 bits
+
+        "vshll.s16      q14,  d12, #15                \n"// extend samples to 31 bits
+        "vshll.s16      q15,  d13, #15                \n"// extend samples to 31 bits
+
+        "vqrdmulh.s32   q12, q12, q9                  \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q13, q13, q8                  \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q14, q14, q10                 \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q15, q15, q11                 \n"// multiply samples by interpolated coef
+
+        "vadd.s32       q4, q4, q12                   \n"// accumulate result
+        "vadd.s32       q13, q13, q14                 \n"// accumulate result
+        "vadd.s32       q4, q4, q15                   \n"// (+1) accumulate result
+        "vadd.s32       q4, q4, q13                   \n"// (+1) accumulate result
+
+        "subs           %[count], %[count], #8        \n"// update loop counter
+        "sub            %[sP], %[sP], #32             \n"// move pointer to next set of samples
+
+        "bne            1b                            \n"// loop
+
+        ASSEMBLY_ACCUMULATE_STEREO
+
+        : [out]     "=Uv" (out[0]),
+          [count]   "+r" (count),
+          [coefsP0] "+r" (coefsP),
+          [coefsN0] "+r" (coefsN),
+          [sP]      "+r" (sP),
+          [sN]      "+r" (sN)
+        : [vLR]     "r" (volumeLR)
+        : "cc", "memory",
+          "q0", "q1", "q2", "q3",
+          "q4", "q5", "q6",
+          "q8", "q9", "q10", "q11",
+          "q12", "q13", "q14", "q15"
+    );
+}
+
+template <>
+inline void Process<1, 16>(int32_t* const out,
+        int count,
+        const int32_t* coefsP,
+        const int32_t* coefsN,
+        const int32_t* coefsP1,
+        const int32_t* coefsN1,
+        const int16_t* sP,
+        const int16_t* sN,
+        uint32_t lerpP,
+        const int32_t* const volumeLR)
+{
+    const int CHANNELS = 1; // template specialization does not preserve params
+    const int STRIDE = 16;
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    asm (
+        "vmov.32        d2[0], %[lerpP]               \n"// load the positive phase
+        "veor           q0, q0, q0                    \n"// result, initialize to 0
+
+        "1:                                           \n"
+
+        "vld1.16        {q2}, [%[sP]]                 \n"// load 8 16-bits mono samples
+        "vld1.16        {q3}, [%[sN]]!                \n"// load 8 16-bits mono samples
+        "vld1.32        {q8, q9}, [%[coefsP0]:128]!   \n"// load 8 32-bits coefs
+        "vld1.32        {q12, q13}, [%[coefsP1]:128]! \n"// load 8 32-bits coefs
+        "vld1.32        {q10, q11}, [%[coefsN1]:128]! \n"// load 8 32-bits coefs
+        "vld1.32        {q14, q15}, [%[coefsN0]:128]! \n"// load 8 32-bits coefs
+
+        "vsub.s32       q12, q12, q8                  \n"// interpolate (step1)
+        "vsub.s32       q13, q13, q9                  \n"// interpolate (step1)
+        "vsub.s32       q14, q14, q10                 \n"// interpolate (step1)
+        "vsub.s32       q15, q15, q11                 \n"// interpolate (step1)
+
+        "vqrdmulh.s32   q12, q12, d2[0]               \n"// interpolate (step2)
+        "vqrdmulh.s32   q13, q13, d2[0]               \n"// interpolate (step2)
+        "vqrdmulh.s32   q14, q14, d2[0]               \n"// interpolate (step2)
+        "vqrdmulh.s32   q15, q15, d2[0]               \n"// interpolate (step2)
+
+        "vadd.s32       q8, q8, q12                   \n"// interpolate (step3)
+        "vadd.s32       q9, q9, q13                   \n"// interpolate (step3)
+        "vadd.s32       q10, q10, q14                 \n"// interpolate (step3)
+        "vadd.s32       q11, q11, q15                 \n"// interpolate (step3)
+
+        "vrev64.16      q2, q2                        \n"// reverse 8 frames of the positive side
+
+        "vshll.s16      q12,  d4, #15                 \n"// extend samples to 31 bits
+        "vshll.s16      q13,  d5, #15                 \n"// extend samples to 31 bits
+
+        "vshll.s16      q14,  d6, #15                 \n"// extend samples to 31 bits
+        "vshll.s16      q15,  d7, #15                 \n"// extend samples to 31 bits
+
+        "vqrdmulh.s32   q12, q12, q9                  \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q13, q13, q8                  \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q14, q14, q10                 \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q15, q15, q11                 \n"// multiply samples by interpolated coef
+
+        "vadd.s32       q0, q0, q12                   \n"// accumulate result
+        "vadd.s32       q13, q13, q14                 \n"// accumulate result
+        "vadd.s32       q0, q0, q15                   \n"// accumulate result
+        "vadd.s32       q0, q0, q13                   \n"// accumulate result
+
+        "sub            %[sP], %[sP], #16             \n"// move pointer to next set of samples
+        "subs           %[count], %[count], #8        \n"// update loop counter
+
+        "bne            1b                            \n"// loop
+
+        ASSEMBLY_ACCUMULATE_MONO
+
+        : [out]     "=Uv" (out[0]),
+          [count]   "+r" (count),
+          [coefsP0] "+r" (coefsP),
+          [coefsN0] "+r" (coefsN),
+          [coefsP1] "+r" (coefsP1),
+          [coefsN1] "+r" (coefsN1),
+          [sP]      "+r" (sP),
+          [sN]      "+r" (sN)
+        : [lerpP]   "r" (lerpP),
+          [vLR]     "r" (volumeLR)
+        : "cc", "memory",
+          "q0", "q1", "q2", "q3",
+          "q8", "q9", "q10", "q11",
+          "q12", "q13", "q14", "q15"
+    );
+}
+
+template <>
+inline void Process<2, 16>(int32_t* const out,
+        int count,
+        const int32_t* coefsP,
+        const int32_t* coefsN,
+        const int32_t* coefsP1,
+        const int32_t* coefsN1,
+        const int16_t* sP,
+        const int16_t* sN,
+        uint32_t lerpP,
+        const int32_t* const volumeLR)
+{
+    const int CHANNELS = 2; // template specialization does not preserve params
+    const int STRIDE = 16;
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    asm (
+        "vmov.32        d2[0], %[lerpP]               \n"// load the positive phase
+        "veor           q0, q0, q0                    \n"// result, initialize to 0
+        "veor           q4, q4, q4                    \n"// result, initialize to 0
+
+        "1:                                           \n"
+
+        "vld2.16        {q2, q3}, [%[sP]]             \n"// load 4 16-bits stereo samples
+        "vld2.16        {q5, q6}, [%[sN]]!            \n"// load 4 16-bits stereo samples
+        "vld1.32        {q8, q9}, [%[coefsP0]:128]!   \n"// load 8 32-bits coefs
+        "vld1.32        {q12, q13}, [%[coefsP1]:128]! \n"// load 8 32-bits coefs
+        "vld1.32        {q10, q11}, [%[coefsN1]:128]! \n"// load 8 32-bits coefs
+        "vld1.32        {q14, q15}, [%[coefsN0]:128]! \n"// load 8 32-bits coefs
+
+        "vsub.s32       q12, q12, q8                  \n"// interpolate (step1)
+        "vsub.s32       q13, q13, q9                  \n"// interpolate (step1)
+        "vsub.s32       q14, q14, q10                 \n"// interpolate (step1)
+        "vsub.s32       q15, q15, q11                 \n"// interpolate (step1)
+
+        "vqrdmulh.s32   q12, q12, d2[0]               \n"// interpolate (step2)
+        "vqrdmulh.s32   q13, q13, d2[0]               \n"// interpolate (step2)
+        "vqrdmulh.s32   q14, q14, d2[0]               \n"// interpolate (step2)
+        "vqrdmulh.s32   q15, q15, d2[0]               \n"// interpolate (step2)
+
+        "vadd.s32       q8, q8, q12                   \n"// interpolate (step3)
+        "vadd.s32       q9, q9, q13                   \n"// interpolate (step3)
+        "vadd.s32       q10, q10, q14                 \n"// interpolate (step3)
+        "vadd.s32       q11, q11, q15                 \n"// interpolate (step3)
+
+        "vrev64.16      q2, q2                        \n"// reverse 8 frames of the positive side
+        "vrev64.16      q3, q3                        \n"// reverse 8 frames of the positive side
+
+        "vshll.s16      q12,  d4, #15                 \n"// extend samples to 31 bits
+        "vshll.s16      q13,  d5, #15                 \n"// extend samples to 31 bits
+
+        "vshll.s16      q14,  d10, #15                \n"// extend samples to 31 bits
+        "vshll.s16      q15,  d11, #15                \n"// extend samples to 31 bits
+
+        "vqrdmulh.s32   q12, q12, q9                  \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q13, q13, q8                  \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q14, q14, q10                 \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q15, q15, q11                 \n"// multiply samples by interpolated coef
+
+        "vadd.s32       q0, q0, q12                   \n"// accumulate result
+        "vadd.s32       q13, q13, q14                 \n"// accumulate result
+        "vadd.s32       q0, q0, q15                   \n"// (+1) accumulate result
+        "vadd.s32       q0, q0, q13                   \n"// (+1) accumulate result
+
+        "vshll.s16      q12,  d6, #15                 \n"// extend samples to 31 bits
+        "vshll.s16      q13,  d7, #15                 \n"// extend samples to 31 bits
+
+        "vshll.s16      q14,  d12, #15                \n"// extend samples to 31 bits
+        "vshll.s16      q15,  d13, #15                \n"// extend samples to 31 bits
+
+        "vqrdmulh.s32   q12, q12, q9                  \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q13, q13, q8                  \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q14, q14, q10                 \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q15, q15, q11                 \n"// multiply samples by interpolated coef
+
+        "vadd.s32       q4, q4, q12                   \n"// accumulate result
+        "vadd.s32       q13, q13, q14                 \n"// accumulate result
+        "vadd.s32       q4, q4, q15                   \n"// (+1) accumulate result
+        "vadd.s32       q4, q4, q13                   \n"// (+1) accumulate result
+
+        "subs           %[count], %[count], #8        \n"// update loop counter
+        "sub            %[sP], %[sP], #32             \n"// move pointer to next set of samples
+
+        "bne            1b                            \n"// loop
+
+        ASSEMBLY_ACCUMULATE_STEREO
+
+        : [out]     "=Uv" (out[0]),
+          [count]   "+r" (count),
+          [coefsP0] "+r" (coefsP),
+          [coefsN0] "+r" (coefsN),
+          [coefsP1] "+r" (coefsP1),
+          [coefsN1] "+r" (coefsN1),
+          [sP]      "+r" (sP),
+          [sN]      "+r" (sN)
+        : [lerpP]   "r" (lerpP),
+          [vLR]     "r" (volumeLR)
+        : "cc", "memory",
+          "q0", "q1", "q2", "q3",
+          "q4", "q5", "q6",
+          "q8", "q9", "q10", "q11",
+          "q12", "q13", "q14", "q15"
+    );
+}
+
+template <>
+inline void ProcessL<1, 8>(int32_t* const out,
+        int count,
+        const int16_t* coefsP,
+        const int16_t* coefsN,
+        const int16_t* sP,
+        const int16_t* sN,
+        const int32_t* const volumeLR)
+{
+    const int CHANNELS = 1; // template specialization does not preserve params
+    const int STRIDE = 8;
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    asm (
+        "veor           q0, q0, q0               \n"// (0 - combines+) accumulator = 0
+
+        "1:                                      \n"
+
+        "vld1.16        {d4}, [%[sP]]            \n"// (2+0d) load 4 16-bits mono samples
+        "vld1.16        {d6}, [%[sN]]!           \n"// (2) load 4 16-bits mono samples
+        "vld1.16        {d16}, [%[coefsP0]:64]!  \n"// (1) load 4 16-bits coefs
+        "vld1.16        {d20}, [%[coefsN0]:64]!  \n"// (1) load 4 16-bits coefs
+
+        "vrev64.16      d4, d4                   \n"// (1) reversed s3, s2, s1, s0, s7, s6, s5, s4
+
+        // reordering the vmal to do d6, d7 before d4, d5 is slower(?)
+        "vmlal.s16      q0, d4, d16              \n"// (1) multiply (reversed)samples by coef
+        "vmlal.s16      q0, d6, d20              \n"// (1) multiply neg samples
+
+        // moving these ARM instructions before neon above seems to be slower
+        "subs           %[count], %[count], #4   \n"// (1) update loop counter
+        "sub            %[sP], %[sP], #8         \n"// (0) move pointer to next set of samples
+
+        // sP used after branch (warning)
+        "bne            1b                       \n"// loop
+
+        ASSEMBLY_ACCUMULATE_MONO
+
+        : [out]     "=Uv" (out[0]),
+          [count]   "+r" (count),
+          [coefsP0] "+r" (coefsP),
+          [coefsN0] "+r" (coefsN),
+          [sP]      "+r" (sP),
+          [sN]      "+r" (sN)
+        : [vLR]     "r" (volumeLR)
+        : "cc", "memory",
+          "q0", "q1", "q2", "q3",
+          "q8", "q10"
+    );
+}
+
+template <>
+inline void ProcessL<2, 8>(int32_t* const out,
+        int count,
+        const int16_t* coefsP,
+        const int16_t* coefsN,
+        const int16_t* sP,
+        const int16_t* sN,
+        const int32_t* const volumeLR)
+{
+    const int CHANNELS = 2; // template specialization does not preserve params
+    const int STRIDE = 8;
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    asm (
+        "veor           q0, q0, q0               \n"// (1) acc_L = 0
+        "veor           q4, q4, q4               \n"// (0 combines+) acc_R = 0
+
+        "1:                                      \n"
+
+        "vld2.16        {d4, d5}, [%[sP]]        \n"// (2+0d) load 8 16-bits stereo samples
+        "vld2.16        {d6, d7}, [%[sN]]!       \n"// (2) load 8 16-bits stereo samples
+        "vld1.16        {d16}, [%[coefsP0]:64]!  \n"// (1) load 8 16-bits coefs
+        "vld1.16        {d20}, [%[coefsN0]:64]!  \n"// (1) load 8 16-bits coefs
+
+        "vrev64.16      q2, q2                   \n"// (1) reverse 8 frames of the left positive
+
+        "vmlal.s16      q0, d4, d16              \n"// (1) multiply (reversed) samples left
+        "vmlal.s16      q4, d5, d16              \n"// (1) multiply (reversed) samples right
+        "vmlal.s16      q0, d6, d20              \n"// (1) multiply samples left
+        "vmlal.s16      q4, d7, d20              \n"// (1) multiply samples right
+
+        // moving these ARM before neon seems to be slower
+        "subs           %[count], %[count], #4   \n"// (1) update loop counter
+        "sub            %[sP], %[sP], #16        \n"// (0) move pointer to next set of samples
+
+        // sP used after branch (warning)
+        "bne            1b                       \n"// loop
+
+        ASSEMBLY_ACCUMULATE_STEREO
+
+        : [out] "=Uv" (out[0]),
+          [count] "+r" (count),
+          [coefsP0] "+r" (coefsP),
+          [coefsN0] "+r" (coefsN),
+          [sP] "+r" (sP),
+          [sN] "+r" (sN)
+        : [vLR] "r" (volumeLR)
+        : "cc", "memory",
+          "q0", "q1", "q2", "q3",
+          "q4", "q5", "q6",
+          "q8", "q10"
+     );
+}
+
+template <>
+inline void Process<1, 8>(int32_t* const out,
+        int count,
+        const int16_t* coefsP,
+        const int16_t* coefsN,
+        const int16_t* coefsP1,
+        const int16_t* coefsN1,
+        const int16_t* sP,
+        const int16_t* sN,
+        uint32_t lerpP,
+        const int32_t* const volumeLR)
+{
+    const int CHANNELS = 1; // template specialization does not preserve params
+    const int STRIDE = 8;
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    asm (
+        "vmov.32        d2[0], %[lerpP]          \n"// load the positive phase S32 Q15
+        "veor           q0, q0, q0               \n"// (0 - combines+) accumulator = 0
+
+        "1:                                      \n"
+
+        "vld1.16        {d4}, [%[sP]]            \n"// (2+0d) load 4 16-bits mono samples
+        "vld1.16        {d6}, [%[sN]]!           \n"// (2) load 4 16-bits mono samples
+        "vld1.16        {d16}, [%[coefsP0]:64]!  \n"// (1) load 4 16-bits coefs
+        "vld1.16        {d17}, [%[coefsP1]:64]!  \n"// (1) load 4 16-bits coefs for interpolation
+        "vld1.16        {d20}, [%[coefsN1]:64]!  \n"// (1) load 4 16-bits coefs
+        "vld1.16        {d21}, [%[coefsN0]:64]!  \n"// (1) load 4 16-bits coefs for interpolation
+
+        "vsub.s16       d17, d17, d16            \n"// (1) interpolate (step1) 1st set of coefs
+        "vsub.s16       d21, d21, d20            \n"// (1) interpolate (step1) 2nd set of coets
+
+        "vqrdmulh.s16   d17, d17, d2[0]          \n"// (2) interpolate (step2) 1st set of coefs
+        "vqrdmulh.s16   d21, d21, d2[0]          \n"// (2) interpolate (step2) 2nd set of coefs
+
+        "vrev64.16      d4, d4                   \n"// (1) reverse s3, s2, s1, s0, s7, s6, s5, s4
+
+        "vadd.s16       d16, d16, d17            \n"// (1+2d) interpolate (step3) 1st set
+        "vadd.s16       d20, d20, d21            \n"// (1+1d) interpolate (step3) 2nd set
+
+        // reordering the vmal to do d6, d7 before d4, d5 is slower(?)
+        "vmlal.s16      q0, d4, d16              \n"// (1+0d) multiply (reversed)by coef
+        "vmlal.s16      q0, d6, d20              \n"// (1) multiply neg samples
+
+        // moving these ARM instructions before neon above seems to be slower
+        "subs           %[count], %[count], #4   \n"// (1) update loop counter
+        "sub            %[sP], %[sP], #8        \n"// move pointer to next set of samples
+
+        // sP used after branch (warning)
+        "bne            1b                       \n"// loop
+
+        ASSEMBLY_ACCUMULATE_MONO
+
+        : [out]     "=Uv" (out[0]),
+          [count]   "+r" (count),
+          [coefsP0] "+r" (coefsP),
+          [coefsN0] "+r" (coefsN),
+          [coefsP1] "+r" (coefsP1),
+          [coefsN1] "+r" (coefsN1),
+          [sP]      "+r" (sP),
+          [sN]      "+r" (sN)
+        : [lerpP]   "r" (lerpP),
+          [vLR]     "r" (volumeLR)
+        : "cc", "memory",
+          "q0", "q1", "q2", "q3",
+          "q8", "q9", "q10", "q11"
+    );
+}
+
+template <>
+inline void Process<2, 8>(int32_t* const out,
+        int count,
+        const int16_t* coefsP,
+        const int16_t* coefsN,
+        const int16_t* coefsP1,
+        const int16_t* coefsN1,
+        const int16_t* sP,
+        const int16_t* sN,
+        uint32_t lerpP,
+        const int32_t* const volumeLR)
+{
+    const int CHANNELS = 2; // template specialization does not preserve params
+    const int STRIDE = 8;
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    asm (
+        "vmov.32        d2[0], %[lerpP]          \n"// load the positive phase
+        "veor           q0, q0, q0               \n"// (1) acc_L = 0
+        "veor           q4, q4, q4               \n"// (0 combines+) acc_R = 0
+
+        "1:                                      \n"
+
+        "vld2.16        {d4, d5}, [%[sP]]        \n"// (3+0d) load 8 16-bits stereo samples
+        "vld2.16        {d6, d7}, [%[sN]]!       \n"// (3) load 8 16-bits stereo samples
+        "vld1.16        {d16}, [%[coefsP0]:64]!  \n"// (1) load 8 16-bits coefs
+        "vld1.16        {d17}, [%[coefsP1]:64]!  \n"// (1) load 8 16-bits coefs for interpolation
+        "vld1.16        {d20}, [%[coefsN1]:64]!  \n"// (1) load 8 16-bits coefs
+        "vld1.16        {d21}, [%[coefsN0]:64]!  \n"// (1) load 8 16-bits coefs for interpolation
+
+        "vsub.s16       d17, d17, d16            \n"// (1) interpolate (step1) 1st set of coefs
+        "vsub.s16       d21, d21, d20            \n"// (1) interpolate (step1) 2nd set of coets
+
+        "vqrdmulh.s16   d17, d17, d2[0]          \n"// (2) interpolate (step2) 1st set of coefs
+        "vqrdmulh.s16   d21, d21, d2[0]          \n"// (2) interpolate (step2) 2nd set of coefs
+
+        "vrev64.16      q2, q2                   \n"// (1) reverse 8 frames of the left positive
+
+        "vadd.s16       d16, d16, d17            \n"// (1+1d) interpolate (step3) 1st set
+        "vadd.s16       d20, d20, d21            \n"// (1+1d) interpolate (step3) 2nd set
+
+        "vmlal.s16      q0, d4, d16              \n"// (1) multiply (reversed) samples left
+        "vmlal.s16      q4, d5, d16              \n"// (1) multiply (reversed) samples right
+        "vmlal.s16      q0, d6, d20              \n"// (1) multiply samples left
+        "vmlal.s16      q4, d7, d20              \n"// (1) multiply samples right
+
+        // moving these ARM before neon seems to be slower
+        "subs           %[count], %[count], #4   \n"// (1) update loop counter
+        "sub            %[sP], %[sP], #16        \n"// move pointer to next set of samples
+
+        // sP used after branch (warning)
+        "bne            1b                       \n"// loop
+
+        ASSEMBLY_ACCUMULATE_STEREO
+
+        : [out] "=Uv" (out[0]),
+          [count] "+r" (count),
+          [coefsP0] "+r" (coefsP),
+          [coefsN0] "+r" (coefsN),
+          [coefsP1] "+r" (coefsP1),
+          [coefsN1] "+r" (coefsN1),
+          [sP] "+r" (sP),
+          [sN] "+r" (sN)
+        : [lerpP]   "r" (lerpP),
+          [vLR] "r" (volumeLR)
+        : "cc", "memory",
+          "q0", "q1", "q2", "q3",
+          "q4", "q5", "q6",
+          "q8", "q9", "q10", "q11"
+    );
+}
+
+template <>
+inline void ProcessL<1, 8>(int32_t* const out,
+        int count,
+        const int32_t* coefsP,
+        const int32_t* coefsN,
+        const int16_t* sP,
+        const int16_t* sN,
+        const int32_t* const volumeLR)
+{
+    const int CHANNELS = 1; // template specialization does not preserve params
+    const int STRIDE = 8;
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    asm (
+        "veor           q0, q0, q0               \n"// result, initialize to 0
+
+        "1:                                      \n"
+
+        "vld1.16        {d4}, [%[sP]]            \n"// load 4 16-bits mono samples
+        "vld1.16        {d6}, [%[sN]]!           \n"// load 4 16-bits mono samples
+        "vld1.32        {q8}, [%[coefsP0]:128]!  \n"// load 4 32-bits coefs
+        "vld1.32        {q10}, [%[coefsN0]:128]! \n"// load 4 32-bits coefs
+
+        "vrev64.16      d4, d4                   \n"// reverse 2 frames of the positive side
+
+        "vshll.s16      q12, d4, #15             \n"// (stall) extend samples to 31 bits
+        "vshll.s16      q14, d6, #15             \n"// extend samples to 31 bits
+
+        "vqrdmulh.s32   q12, q12, q8             \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q14, q14, q10            \n"// multiply samples by interpolated coef
+
+        "vadd.s32       q0, q0, q12              \n"// accumulate result
+        "vadd.s32       q0, q0, q14              \n"// (stall) accumulate result
+
+        "subs           %[count], %[count], #4   \n"// update loop counter
+        "sub            %[sP], %[sP], #8         \n"// move pointer to next set of samples
+
+        "bne            1b                       \n"// loop
+
+        ASSEMBLY_ACCUMULATE_MONO
+
+        : [out] "=Uv" (out[0]),
+          [count] "+r" (count),
+          [coefsP0] "+r" (coefsP),
+          [coefsN0] "+r" (coefsN),
+          [sP] "+r" (sP),
+          [sN] "+r" (sN)
+        : [vLR] "r" (volumeLR)
+        : "cc", "memory",
+          "q0", "q1", "q2", "q3",
+          "q8", "q9", "q10", "q11",
+          "q12", "q14"
+    );
+}
+
+template <>
+inline void ProcessL<2, 8>(int32_t* const out,
+        int count,
+        const int32_t* coefsP,
+        const int32_t* coefsN,
+        const int16_t* sP,
+        const int16_t* sN,
+        const int32_t* const volumeLR)
+{
+    const int CHANNELS = 2; // template specialization does not preserve params
+    const int STRIDE = 8;
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    asm (
+        "veor           q0, q0, q0               \n"// result, initialize to 0
+        "veor           q4, q4, q4               \n"// result, initialize to 0
+
+        "1:                                      \n"
+
+        "vld2.16        {d4, d5}, [%[sP]]        \n"// load 4 16-bits stereo samples
+        "vld2.16        {d6, d7}, [%[sN]]!       \n"// load 4 16-bits stereo samples
+        "vld1.32        {q8}, [%[coefsP0]:128]!  \n"// load 4 32-bits coefs
+        "vld1.32        {q10}, [%[coefsN0]:128]! \n"// load 4 32-bits coefs
+
+        "vrev64.16      q2, q2                   \n"// reverse 2 frames of the positive side
+
+        "vshll.s16      q12, d4, #15             \n"// extend samples to 31 bits
+        "vshll.s16      q13, d5, #15             \n"// extend samples to 31 bits
+
+        "vshll.s16      q14, d6, #15             \n"// extend samples to 31 bits
+        "vshll.s16      q15, d7, #15             \n"// extend samples to 31 bits
+
+        "vqrdmulh.s32   q12, q12, q8             \n"// multiply samples by coef
+        "vqrdmulh.s32   q13, q13, q8             \n"// multiply samples by coef
+        "vqrdmulh.s32   q14, q14, q10            \n"// multiply samples by coef
+        "vqrdmulh.s32   q15, q15, q10            \n"// multiply samples by coef
+
+        "vadd.s32       q0, q0, q12              \n"// accumulate result
+        "vadd.s32       q4, q4, q13              \n"// accumulate result
+        "vadd.s32       q0, q0, q14              \n"// accumulate result
+        "vadd.s32       q4, q4, q15              \n"// accumulate result
+
+        "subs           %[count], %[count], #4   \n"// update loop counter
+        "sub            %[sP], %[sP], #16        \n"// move pointer to next set of samples
+
+        "bne            1b                       \n"// loop
+
+        ASSEMBLY_ACCUMULATE_STEREO
+
+        : [out]     "=Uv" (out[0]),
+          [count]   "+r" (count),
+          [coefsP0] "+r" (coefsP),
+          [coefsN0] "+r" (coefsN),
+          [sP]      "+r" (sP),
+          [sN]      "+r" (sN)
+        : [vLR]     "r" (volumeLR)
+        : "cc", "memory",
+          "q0", "q1", "q2", "q3", "q4",
+          "q8", "q9", "q10", "q11",
+          "q12", "q13", "q14", "q15"
+    );
+}
+
+template <>
+inline void Process<1, 8>(int32_t* const out,
+        int count,
+        const int32_t* coefsP,
+        const int32_t* coefsN,
+        const int32_t* coefsP1,
+        const int32_t* coefsN1,
+        const int16_t* sP,
+        const int16_t* sN,
+        uint32_t lerpP,
+        const int32_t* const volumeLR)
+{
+    const int CHANNELS = 1; // template specialization does not preserve params
+    const int STRIDE = 8;
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    asm (
+        "vmov.32        d2[0], %[lerpP]          \n"// load the positive phase
+        "veor           q0, q0, q0               \n"// result, initialize to 0
+
+        "1:                                      \n"
+
+        "vld1.16        {d4}, [%[sP]]            \n"// load 4 16-bits mono samples
+        "vld1.16        {d6}, [%[sN]]!           \n"// load 4 16-bits mono samples
+        "vld1.32        {q8}, [%[coefsP0]:128]!  \n"// load 4 32-bits coefs
+        "vld1.32        {q9}, [%[coefsP1]:128]!  \n"// load 4 32-bits coefs for interpolation
+        "vld1.32        {q10}, [%[coefsN1]:128]! \n"// load 4 32-bits coefs
+        "vld1.32        {q11}, [%[coefsN0]:128]! \n"// load 4 32-bits coefs for interpolation
+
+        "vrev64.16      d4, d4                   \n"// reverse 2 frames of the positive side
+
+        "vsub.s32       q9, q9, q8               \n"// interpolate (step1) 1st set of coefs
+        "vsub.s32       q11, q11, q10            \n"// interpolate (step1) 2nd set of coets
+        "vshll.s16      q12, d4, #15             \n"// extend samples to 31 bits
+
+        "vqrdmulh.s32   q9, q9, d2[0]            \n"// interpolate (step2) 1st set of coefs
+        "vqrdmulh.s32   q11, q11, d2[0]          \n"// interpolate (step2) 2nd set of coefs
+        "vshll.s16      q14, d6, #15             \n"// extend samples to 31 bits
+
+        "vadd.s32       q8, q8, q9               \n"// interpolate (step3) 1st set
+        "vadd.s32       q10, q10, q11            \n"// interpolate (step4) 2nd set
+
+        "vqrdmulh.s32   q12, q12, q8             \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q14, q14, q10            \n"// multiply samples by interpolated coef
+
+        "vadd.s32       q0, q0, q12              \n"// accumulate result
+        "vadd.s32       q0, q0, q14              \n"// accumulate result
+
+        "subs           %[count], %[count], #4   \n"// update loop counter
+        "sub            %[sP], %[sP], #8         \n"// move pointer to next set of samples
+
+        "bne            1b                       \n"// loop
+
+        ASSEMBLY_ACCUMULATE_MONO
+
+        : [out]     "=Uv" (out[0]),
+          [count]   "+r" (count),
+          [coefsP0] "+r" (coefsP),
+          [coefsP1] "+r" (coefsP1),
+          [coefsN0] "+r" (coefsN),
+          [coefsN1] "+r" (coefsN1),
+          [sP]      "+r" (sP),
+          [sN]      "+r" (sN)
+        : [lerpP]   "r" (lerpP),
+          [vLR]     "r" (volumeLR)
+        : "cc", "memory",
+          "q0", "q1", "q2", "q3",
+          "q8", "q9", "q10", "q11",
+          "q12", "q14"
+    );
+}
+
+template <>
+inline
+void Process<2, 8>(int32_t* const out,
+        int count,
+        const int32_t* coefsP,
+        const int32_t* coefsN,
+        const int32_t* coefsP1,
+        const int32_t* coefsN1,
+        const int16_t* sP,
+        const int16_t* sN,
+        uint32_t lerpP,
+        const int32_t* const volumeLR)
+{
+    const int CHANNELS = 2; // template specialization does not preserve params
+    const int STRIDE = 8;
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    asm (
+        "vmov.32        d2[0], %[lerpP]          \n"// load the positive phase
+        "veor           q0, q0, q0               \n"// result, initialize to 0
+        "veor           q4, q4, q4               \n"// result, initialize to 0
+
+        "1:                                      \n"
+        "vld2.16        {d4, d5}, [%[sP]]        \n"// load 4 16-bits stereo samples
+        "vld2.16        {d6, d7}, [%[sN]]!       \n"// load 4 16-bits stereo samples
+        "vld1.32        {q8}, [%[coefsP0]:128]!  \n"// load 4 32-bits coefs
+        "vld1.32        {q9}, [%[coefsP1]:128]!  \n"// load 4 32-bits coefs for interpolation
+        "vld1.32        {q10}, [%[coefsN1]:128]! \n"// load 4 32-bits coefs
+        "vld1.32        {q11}, [%[coefsN0]:128]! \n"// load 4 32-bits coefs for interpolation
+
+        "vrev64.16      q2, q2                   \n"// (reversed) 2 frames of the positive side
+
+        "vsub.s32       q9, q9, q8               \n"// interpolate (step1) 1st set of coefs
+        "vsub.s32       q11, q11, q10            \n"// interpolate (step1) 2nd set of coets
+        "vshll.s16      q12, d4, #15             \n"// extend samples to 31 bits
+        "vshll.s16      q13, d5, #15             \n"// extend samples to 31 bits
+
+        "vqrdmulh.s32   q9, q9, d2[0]            \n"// interpolate (step2) 1st set of coefs
+        "vqrdmulh.s32   q11, q11, d2[1]          \n"// interpolate (step3) 2nd set of coefs
+        "vshll.s16      q14, d6, #15             \n"// extend samples to 31 bits
+        "vshll.s16      q15, d7, #15             \n"// extend samples to 31 bits
+
+        "vadd.s32       q8, q8, q9               \n"// interpolate (step3) 1st set
+        "vadd.s32       q10, q10, q11            \n"// interpolate (step4) 2nd set
+
+        "vqrdmulh.s32   q12, q12, q8             \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q13, q13, q8             \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q14, q14, q10            \n"// multiply samples by interpolated coef
+        "vqrdmulh.s32   q15, q15, q10            \n"// multiply samples by interpolated coef
+
+        "vadd.s32       q0, q0, q12              \n"// accumulate result
+        "vadd.s32       q4, q4, q13              \n"// accumulate result
+        "vadd.s32       q0, q0, q14              \n"// accumulate result
+        "vadd.s32       q4, q4, q15              \n"// accumulate result
+
+        "subs           %[count], %[count], #4   \n"// update loop counter
+        "sub            %[sP], %[sP], #16        \n"// move pointer to next set of samples
+
+        "bne            1b                       \n"// loop
+
+        ASSEMBLY_ACCUMULATE_STEREO
+
+        : [out]     "=Uv" (out[0]),
+          [count]   "+r" (count),
+          [coefsP0] "+r" (coefsP),
+          [coefsP1] "+r" (coefsP1),
+          [coefsN0] "+r" (coefsN),
+          [coefsN1] "+r" (coefsN1),
+          [sP]      "+r" (sP),
+          [sN]      "+r" (sN)
+        : [lerpP]   "r" (lerpP),
+          [vLR]     "r" (volumeLR)
+        : "cc", "memory",
+          "q0", "q1", "q2", "q3", "q4",
+          "q8", "q9", "q10", "q11",
+          "q12", "q13", "q14", "q15"
+    );
+}
+
+#endif //USE_NEON
+
+}; // namespace android
+
+#endif /*ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_NEON_H*/

services/audioflinger/AudioResamplerSinc.cpp

diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index 207f26b..d0a7a58 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -540,7 +540,7 @@
     uint32_t phaseIncrement = mPhaseIncrement;
     size_t outputIndex = 0;
     size_t outputSampleCount = outFrameCount * 2;
-    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
+    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
 
     while (outputIndex < outputSampleCount) {
         // buffer is empty, fetch a new one

services/audioflinger/Configuration.h

diff --git a/services/audioflinger/Configuration.h b/services/audioflinger/Configuration.h
index bc2038a..0754d9d 100644
--- a/services/audioflinger/Configuration.h
+++ b/services/audioflinger/Configuration.h
@@ -32,9 +32,6 @@
 // uncomment to enable fast mixer to take performance samples for later statistical analysis
 #define FAST_MIXER_STATISTICS
 
-// uncomment to allow fast tracks at non-native sample rate
-//#define FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE
-
 // uncomment for debugging timing problems related to StateQueue::push()
 //#define STATE_QUEUE_DUMP
 

services/audioflinger/Effects.cpp

diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index c6e78a6..29b56db 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -805,7 +805,112 @@
     return mOffloaded;
 }
 
-void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
+String8 effectFlagsToString(uint32_t flags) {
+    String8 s;
+
+    s.append("conn. mode: ");
+    switch (flags & EFFECT_FLAG_TYPE_MASK) {
+    case EFFECT_FLAG_TYPE_INSERT: s.append("insert"); break;
+    case EFFECT_FLAG_TYPE_AUXILIARY: s.append("auxiliary"); break;
+    case EFFECT_FLAG_TYPE_REPLACE: s.append("replace"); break;
+    case EFFECT_FLAG_TYPE_PRE_PROC: s.append("preproc"); break;
+    case EFFECT_FLAG_TYPE_POST_PROC: s.append("postproc"); break;
+    default: s.append("unknown/reserved"); break;
+    }
+    s.append(", ");
+
+    s.append("insert pref: ");
+    switch (flags & EFFECT_FLAG_INSERT_MASK) {
+    case EFFECT_FLAG_INSERT_ANY: s.append("any"); break;
+    case EFFECT_FLAG_INSERT_FIRST: s.append("first"); break;
+    case EFFECT_FLAG_INSERT_LAST: s.append("last"); break;
+    case EFFECT_FLAG_INSERT_EXCLUSIVE: s.append("exclusive"); break;
+    default: s.append("unknown/reserved"); break;
+    }
+    s.append(", ");
+
+    s.append("volume mgmt: ");
+    switch (flags & EFFECT_FLAG_VOLUME_MASK) {
+    case EFFECT_FLAG_VOLUME_NONE: s.append("none"); break;
+    case EFFECT_FLAG_VOLUME_CTRL: s.append("implements control"); break;
+    case EFFECT_FLAG_VOLUME_IND: s.append("requires indication"); break;
+    default: s.append("unknown/reserved"); break;
+    }
+    s.append(", ");
+
+    uint32_t devind = flags & EFFECT_FLAG_DEVICE_MASK;
+    if (devind) {
+        s.append("device indication: ");
+        switch (devind) {
+        case EFFECT_FLAG_DEVICE_IND: s.append("requires updates"); break;
+        default: s.append("unknown/reserved"); break;
+        }
+        s.append(", ");
+    }
+
+    s.append("input mode: ");
+    switch (flags & EFFECT_FLAG_INPUT_MASK) {
+    case EFFECT_FLAG_INPUT_DIRECT: s.append("direct"); break;
+    case EFFECT_FLAG_INPUT_PROVIDER: s.append("provider"); break;
+    case EFFECT_FLAG_INPUT_BOTH: s.append("direct+provider"); break;
+    default: s.append("not set"); break;
+    }
+    s.append(", ");
+
+    s.append("output mode: ");
+    switch (flags & EFFECT_FLAG_OUTPUT_MASK) {
+    case EFFECT_FLAG_OUTPUT_DIRECT: s.append("direct"); break;
+    case EFFECT_FLAG_OUTPUT_PROVIDER: s.append("provider"); break;
+    case EFFECT_FLAG_OUTPUT_BOTH: s.append("direct+provider"); break;
+    default: s.append("not set"); break;
+    }
+    s.append(", ");
+
+    uint32_t accel = flags & EFFECT_FLAG_HW_ACC_MASK;
+    if (accel) {
+        s.append("hardware acceleration: ");
+        switch (accel) {
+        case EFFECT_FLAG_HW_ACC_SIMPLE: s.append("non-tunneled"); break;
+        case EFFECT_FLAG_HW_ACC_TUNNEL: s.append("tunneled"); break;
+        default: s.append("unknown/reserved"); break;
+        }
+        s.append(", ");
+    }
+
+    uint32_t modeind = flags & EFFECT_FLAG_AUDIO_MODE_MASK;
+    if (modeind) {
+        s.append("mode indication: ");
+        switch (modeind) {
+        case EFFECT_FLAG_AUDIO_MODE_IND: s.append("required"); break;
+        default: s.append("unknown/reserved"); break;
+        }
+        s.append(", ");
+    }
+
+    uint32_t srcind = flags & EFFECT_FLAG_AUDIO_SOURCE_MASK;
+    if (srcind) {
+        s.append("source indication: ");
+        switch (srcind) {
+        case EFFECT_FLAG_AUDIO_SOURCE_IND: s.append("required"); break;
+        default: s.append("unknown/reserved"); break;
+        }
+        s.append(", ");
+    }
+
+    if (flags & EFFECT_FLAG_OFFLOAD_MASK) {
+        s.append("offloadable, ");
+    }
+
+    int len = s.length();
+    if (s.length() > 2) {
+        char *str = s.lockBuffer(len);
+        s.unlockBuffer(len - 2);
+    }
+    return s;
+}
+
+
+void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args __unused)
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
@@ -821,8 +926,8 @@
     }
 
     result.append("\t\tSession Status State Engine:\n");
-    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
-            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
+    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   %p\n",
+            mSessionId, mStatus, mState, mEffectInterface);
     result.append(buffer);
 
     result.append("\t\tDescriptor:\n");
@@ -839,9 +944,10 @@
                     mDescriptor.type.node[2],
                 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
     result.append(buffer);
-    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
+    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X (%s)\n",
             mDescriptor.apiVersion,
-            mDescriptor.flags);
+            mDescriptor.flags,
+            effectFlagsToString(mDescriptor.flags).string());
     result.append(buffer);
     snprintf(buffer, SIZE, "\t\t- name: %s\n",
             mDescriptor.name);
@@ -851,38 +957,38 @@
     result.append(buffer);
 
     result.append("\t\t- Input configuration:\n");
-    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
-    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
-            (uint32_t)mConfig.inputCfg.buffer.raw,
+    result.append("\t\t\tFrames  Smp rate Channels Format Buffer\n");
+    snprintf(buffer, SIZE, "\t\t\t%05zu   %05d    %08x %6d (%s) %p\n",
             mConfig.inputCfg.buffer.frameCount,
             mConfig.inputCfg.samplingRate,
             mConfig.inputCfg.channels,
-            mConfig.inputCfg.format);
+            mConfig.inputCfg.format,
+            formatToString((audio_format_t)mConfig.inputCfg.format),
+            mConfig.inputCfg.buffer.raw);
     result.append(buffer);
 
     result.append("\t\t- Output configuration:\n");
     result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
-    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
-            (uint32_t)mConfig.outputCfg.buffer.raw,
+    snprintf(buffer, SIZE, "\t\t\t%p %05zu   %05d    %08x %d (%s)\n",
+            mConfig.outputCfg.buffer.raw,
             mConfig.outputCfg.buffer.frameCount,
             mConfig.outputCfg.samplingRate,
             mConfig.outputCfg.channels,
-            mConfig.outputCfg.format);
+            mConfig.outputCfg.format,
+            formatToString((audio_format_t)mConfig.outputCfg.format));
     result.append(buffer);
 
-    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
+    snprintf(buffer, SIZE, "\t\t%zu Clients:\n", mHandles.size());
     result.append(buffer);
-    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
+    result.append("\t\t\t  Pid Priority Ctrl Locked client server\n");
     for (size_t i = 0; i < mHandles.size(); ++i) {
         EffectHandle *handle = mHandles[i];
         if (handle != NULL && !handle->destroyed_l()) {
-            handle->dump(buffer, SIZE);
+            handle->dumpToBuffer(buffer, SIZE);
             result.append(buffer);
         }
     }
 
-    result.append("\n");
-
     write(fd, result.string(), result.length());
 
     if (locked) {
@@ -912,18 +1018,15 @@
     }
     int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
     mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
-    if (mCblkMemory != 0) {
-        mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
-
-        if (mCblk != NULL) {
-            new(mCblk) effect_param_cblk_t();
-            mBuffer = (uint8_t *)mCblk + bufOffset;
-        }
-    } else {
+    if (mCblkMemory == 0 ||
+            (mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer())) == NULL) {
         ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE +
                 sizeof(effect_param_cblk_t));
+        mCblkMemory.clear();
         return;
     }
+    new(mCblk) effect_param_cblk_t();
+    mBuffer = (uint8_t *)mCblk + bufOffset;
 }
 
 AudioFlinger::EffectHandle::~EffectHandle()
@@ -940,6 +1043,11 @@
     disconnect(false);
 }
 
+status_t AudioFlinger::EffectHandle::initCheck()
+{
+    return mClient == 0 || mCblkMemory != 0 ? OK : NO_MEMORY;
+}
+
 status_t AudioFlinger::EffectHandle::enable()
 {
     ALOGV("enable %p", this);
@@ -973,13 +1081,20 @@
         }
         mEnabled = false;
     } else {
-        if (thread != 0 && !mEffect->isOffloadable()) {
-            if ((thread->type() == ThreadBase::OFFLOAD)) {
+        if (thread != 0) {
+            if (thread->type() == ThreadBase::OFFLOAD) {
                 PlaybackThread *t = (PlaybackThread *)thread.get();
-                t->invalidateTracks(AUDIO_STREAM_MUSIC);
+                Mutex::Autolock _l(t->mLock);
+                t->broadcast_l();
             }
-            if (mEffect->sessionId() == AUDIO_SESSION_OUTPUT_MIX) {
-                thread->mAudioFlinger->onNonOffloadableGlobalEffectEnable();
+            if (!mEffect->isOffloadable()) {
+                if (thread->type() == ThreadBase::OFFLOAD) {
+                    PlaybackThread *t = (PlaybackThread *)thread.get();
+                    t->invalidateTracks(AUDIO_STREAM_MUSIC);
+                }
+                if (mEffect->sessionId() == AUDIO_SESSION_OUTPUT_MIX) {
+                    thread->mAudioFlinger->onNonOffloadableGlobalEffectEnable();
+                }
             }
         }
     }
@@ -1010,6 +1125,11 @@
     sp<ThreadBase> thread = mEffect->thread().promote();
     if (thread != 0) {
         thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
+        if (thread->type() == ThreadBase::OFFLOAD) {
+            PlaybackThread *t = (PlaybackThread *)thread.get();
+            Mutex::Autolock _l(t->mLock);
+            t->broadcast_l();
+        }
     }
 
     return status;
@@ -1168,15 +1288,15 @@
 }
 
 
-void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
+void AudioFlinger::EffectHandle::dumpToBuffer(char* buffer, size_t size)
 {
     bool locked = mCblk != NULL && AudioFlinger::dumpTryLock(mCblk->lock);
 
-    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
+    snprintf(buffer, size, "\t\t\t%5d    %5d  %3s    %3s  %5u  %5u\n",
             (mClient == 0) ? getpid_cached : mClient->pid(),
             mPriority,
-            mHasControl,
-            !locked,
+            mHasControl ? "yes" : "no",
+            locked ? "yes" : "no",
             mCblk ? mCblk->clientIndex : 0,
             mCblk ? mCblk->serverIndex : 0
             );
@@ -1280,9 +1400,10 @@
     }
     bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
             (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
-    // always process effects unless no more tracks are on the session and the effect tail
-    // has been rendered
-    bool doProcess = true;
+    // never process effects when:
+    // - on an OFFLOAD thread
+    // - no more tracks are on the session and the effect tail has been rendered
+    bool doProcess = (thread->type() != ThreadBase::OFFLOAD);
     if (!isGlobalSession) {
         bool tracksOnSession = (trackCnt() != 0);
 
@@ -1556,33 +1677,35 @@
     char buffer[SIZE];
     String8 result;
 
-    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
+    size_t numEffects = mEffects.size();
+    snprintf(buffer, SIZE, "    %d effects for session %d\n", numEffects, mSessionId);
     result.append(buffer);
 
-    bool locked = AudioFlinger::dumpTryLock(mLock);
-    // failed to lock - AudioFlinger is probably deadlocked
-    if (!locked) {
-        result.append("\tCould not lock mutex:\n");
-    }
-
-    result.append("\tNum fx In buffer   Out buffer   Active tracks:\n");
-    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %d\n",
-            mEffects.size(),
-            (uint32_t)mInBuffer,
-            (uint32_t)mOutBuffer,
-            mActiveTrackCnt);
-    result.append(buffer);
-    write(fd, result.string(), result.size());
-
-    for (size_t i = 0; i < mEffects.size(); ++i) {
-        sp<EffectModule> effect = mEffects[i];
-        if (effect != 0) {
-            effect->dump(fd, args);
+    if (numEffects) {
+        bool locked = AudioFlinger::dumpTryLock(mLock);
+        // failed to lock - AudioFlinger is probably deadlocked
+        if (!locked) {
+            result.append("\tCould not lock mutex:\n");
         }
-    }
 
-    if (locked) {
-        mLock.unlock();
+        result.append("\tIn buffer   Out buffer   Active tracks:\n");
+        snprintf(buffer, SIZE, "\t%p  %p   %d\n",
+                mInBuffer,
+                mOutBuffer,
+                mActiveTrackCnt);
+        result.append(buffer);
+        write(fd, result.string(), result.size());
+
+        for (size_t i = 0; i < numEffects; ++i) {
+            sp<EffectModule> effect = mEffects[i];
+            if (effect != 0) {
+                effect->dump(fd, args);
+            }
+        }
+
+        if (locked) {
+            mLock.unlock();
+        }
     }
 }
 

services/audioflinger/Effects.h

diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index b717857..ccc4825 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -169,6 +169,7 @@
             const sp<IEffectClient>& effectClient,
             int32_t priority);
     virtual ~EffectHandle();
+    virtual status_t initCheck();
 
     // IEffect
     virtual status_t enable();
@@ -208,7 +209,7 @@
     // destroyed_l() must be called with the associated EffectModule mLock held
     bool destroyed_l() const { return mDestroyed; }
 
-    void dump(char* buffer, size_t size);
+    void dumpToBuffer(char* buffer, size_t size);
 
 protected:
     friend class AudioFlinger;          // for mEffect, mHasControl, mEnabled

services/audioflinger/FastMixer.cpp

diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 7126e92..90122e0 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -236,10 +236,9 @@
                     sampleRate = Format_sampleRate(format);
                     ALOG_ASSERT(Format_channelCount(format) == FCC_2);
                 }
-                dumpState->mSampleRate = sampleRate;
             }
 
-            if ((format != previousFormat) || (frameCount != previous->mFrameCount)) {
+            if ((!Format_isEqual(format, previousFormat)) || (frameCount != previous->mFrameCount)) {
                 // FIXME to avoid priority inversion, don't delete here
                 delete mixer;
                 mixer = NULL;
@@ -321,12 +320,8 @@
                         mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
                                 (void *) mixBuffer);
                         // newly allocated track names default to full scale volume
-                        if (fastTrack->mSampleRate != 0 && fastTrack->mSampleRate != sampleRate) {
-                            mixer->setParameter(name, AudioMixer::RESAMPLE,
-                                    AudioMixer::SAMPLE_RATE, (void*) fastTrack->mSampleRate);
-                        }
                         mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK,
-                                (void *) fastTrack->mChannelMask);
+                                (void *)(uintptr_t)fastTrack->mChannelMask);
                         mixer->enable(name);
                     }
                     generations[i] = fastTrack->mGeneration;
@@ -353,16 +348,10 @@
                                 mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1,
                                         (void *)0x1000);
                             }
-                            if (fastTrack->mSampleRate != 0 &&
-                                    fastTrack->mSampleRate != sampleRate) {
-                                mixer->setParameter(name, AudioMixer::RESAMPLE,
-                                        AudioMixer::SAMPLE_RATE, (void*) fastTrack->mSampleRate);
-                            } else {
-                                mixer->setParameter(name, AudioMixer::RESAMPLE,
-                                        AudioMixer::REMOVE, NULL);
-                            }
+                            mixer->setParameter(name, AudioMixer::RESAMPLE,
+                                    AudioMixer::REMOVE, NULL);
                             mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK,
-                                    (void *) fastTrack->mChannelMask);
+                                    (void *)(uintptr_t) fastTrack->mChannelMask);
                             // already enabled
                         }
                         generations[i] = fastTrack->mGeneration;
@@ -392,16 +381,8 @@
 
                 // Refresh the per-track timestamp
                 if (timestampStatus == NO_ERROR) {
-                    uint32_t trackFramesWrittenButNotPresented;
-                    uint32_t trackSampleRate = fastTrack->mSampleRate;
-                    // There is currently no sample rate conversion for fast tracks currently
-                    if (trackSampleRate != 0 && trackSampleRate != sampleRate) {
-                        trackFramesWrittenButNotPresented =
-                                ((int64_t) nativeFramesWrittenButNotPresented * trackSampleRate) /
-                                sampleRate;
-                    } else {
-                        trackFramesWrittenButNotPresented = nativeFramesWrittenButNotPresented;
-                    }
+                    uint32_t trackFramesWrittenButNotPresented =
+                        nativeFramesWrittenButNotPresented;
                     uint32_t trackFramesWritten = fastTrack->mBufferProvider->framesReleased();
                     // Can't provide an AudioTimestamp before first frame presented,
                     // or during the brief 32-bit wraparound window
@@ -419,9 +400,9 @@
                 if (fastTrack->mVolumeProvider != NULL) {
                     uint32_t vlr = fastTrack->mVolumeProvider->getVolumeLR();
                     mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0,
-                            (void *)(vlr & 0xFFFF));
+                            (void *)(uintptr_t)(vlr & 0xFFFF));
                     mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1,
-                            (void *)(vlr >> 16));
+                            (void *)(uintptr_t)(vlr >> 16));
                 }
                 // FIXME The current implementation of framesReady() for fast tracks
                 // takes a tryLock, which can block
@@ -715,7 +696,7 @@
 void FastMixerDumpState::dump(int fd) const
 {
     if (mCommand == FastMixerState::INITIAL) {
-        fdprintf(fd, "FastMixer not initialized\n");
+        fdprintf(fd, "  FastMixer not initialized\n");
         return;
     }
 #define COMMAND_MAX 32
@@ -749,10 +730,10 @@
     double measuredWarmupMs = (mMeasuredWarmupTs.tv_sec * 1000.0) +
             (mMeasuredWarmupTs.tv_nsec / 1000000.0);
     double mixPeriodSec = (double) mFrameCount / (double) mSampleRate;
-    fdprintf(fd, "FastMixer command=%s writeSequence=%u framesWritten=%u\n"
-                 "          numTracks=%u writeErrors=%u underruns=%u overruns=%u\n"
-                 "          sampleRate=%u frameCount=%u measuredWarmup=%.3g ms, warmupCycles=%u\n"
-                 "          mixPeriod=%.2f ms\n",
+    fdprintf(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,
@@ -803,14 +784,20 @@
         previousCpukHz = sampleCpukHz;
 #endif
     }
-    fdprintf(fd, "Simple moving statistics over last %.1f seconds:\n", wall.n() * mixPeriodSec);
-    fdprintf(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);
-    fdprintf(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);
+    if (n) {
+        fdprintf(fd, "  Simple moving statistics over last %.1f seconds:\n",
+                     wall.n() * mixPeriodSec);
+        fdprintf(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);
+        fdprintf(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 {
+        fdprintf(fd, "  No FastMixer statistics available currently\n");
+    }
 #ifdef CPU_FREQUENCY_STATISTICS
     fdprintf(fd, "  CPU clock frequency in MHz:\n"
                  "    mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
@@ -828,9 +815,9 @@
             left.sample(tail[i]);
             right.sample(tail[n - (i + 1)]);
         }
-        fdprintf(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",
+        fdprintf(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);
@@ -843,9 +830,9 @@
     // Instead we always display all tracks, with an indication
     // of whether we think the track is active.
     uint32_t trackMask = mTrackMask;
-    fdprintf(fd, "Fast tracks: kMaxFastTracks=%u activeMask=%#x\n",
+    fdprintf(fd, "  Fast tracks: kMaxFastTracks=%u activeMask=%#x\n",
             FastMixerState::kMaxFastTracks, trackMask);
-    fdprintf(fd, "Index Active Full Partial Empty  Recent Ready\n");
+    fdprintf(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];
@@ -865,7 +852,7 @@
             mostRecent = "?";
             break;
         }
-        fdprintf(fd, "%5u %6s %4u %7u %5u %7s %5u\n", i, isActive ? "yes" : "no",
+        fdprintf(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,

services/audioflinger/FastMixerState.cpp

diff --git a/services/audioflinger/FastMixerState.cpp b/services/audioflinger/FastMixerState.cpp
index 737de97..43ff233 100644
--- a/services/audioflinger/FastMixerState.cpp
+++ b/services/audioflinger/FastMixerState.cpp
@@ -20,7 +20,7 @@
 namespace android {
 
 FastTrack::FastTrack() :
-    mBufferProvider(NULL), mVolumeProvider(NULL), mSampleRate(0),
+    mBufferProvider(NULL), mVolumeProvider(NULL),
     mChannelMask(AUDIO_CHANNEL_OUT_STEREO), mGeneration(0)
 {
 }

services/audioflinger/FastMixerState.h

diff --git a/services/audioflinger/FastMixerState.h b/services/audioflinger/FastMixerState.h
index f6e7903..9739fe9 100644
--- a/services/audioflinger/FastMixerState.h
+++ b/services/audioflinger/FastMixerState.h
@@ -43,7 +43,6 @@
 
     ExtendedAudioBufferProvider* mBufferProvider; // must be NULL if inactive, or non-NULL if active
     VolumeProvider*         mVolumeProvider; // optional; if NULL then full-scale
-    unsigned                mSampleRate;     // optional; if zero then use mixer sample rate
     audio_channel_mask_t    mChannelMask;    // AUDIO_CHANNEL_OUT_MONO or AUDIO_CHANNEL_OUT_STEREO
     int                     mGeneration;     // increment when any field is assigned
 };

services/audioflinger/PlaybackTracks.h

diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index be4b811..b5e763d 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -31,12 +31,13 @@
                                 size_t frameCount,
                                 const sp<IMemory>& sharedBuffer,
                                 int sessionId,
+                                int uid,
                                 IAudioFlinger::track_flags_t flags);
     virtual             ~Track();
     virtual status_t    initCheck() const;
 
     static  void        appendDumpHeader(String8& result);
-            void        dump(char* buffer, size_t size);
+            void        dump(char* buffer, size_t size, bool active);
     virtual status_t    start(AudioSystem::sync_event_t event =
                                     AudioSystem::SYNC_EVENT_NONE,
                              int triggerSession = 0);
@@ -61,6 +62,7 @@
             int16_t     *mainBuffer() const { return mMainBuffer; }
             int         auxEffectId() const { return mAuxEffectId; }
     virtual status_t    getTimestamp(AudioTimestamp& timestamp);
+            void        signal();
 
 // implement FastMixerState::VolumeProvider interface
     virtual uint32_t    getVolumeLR();
@@ -92,6 +94,8 @@
     bool isReady() const;
     void setPaused() { mState = PAUSED; }
     void reset();
+    bool isFlushPending() const { return mFlushHwPending; }
+    void flushAck();
 
     bool isOutputTrack() const {
         return (mStreamType == AUDIO_STREAM_CNT);
@@ -153,6 +157,7 @@
     bool                mIsInvalid; // non-resettable latch, set by invalidate()
     AudioTrackServerProxy*  mAudioTrackServerProxy;
     bool                mResumeToStopping; // track was paused in stopping state.
+    bool                mFlushHwPending; // track requests for thread flush
 };  // end of Track
 
 class TimedTrack : public Track {
@@ -165,7 +170,8 @@
                                  audio_channel_mask_t channelMask,
                                  size_t frameCount,
                                  const sp<IMemory>& sharedBuffer,
-                                 int sessionId);
+                                 int sessionId,
+                                 int uid);
     virtual ~TimedTrack();
 
     class TimedBuffer {
@@ -208,7 +214,8 @@
                audio_channel_mask_t channelMask,
                size_t frameCount,
                const sp<IMemory>& sharedBuffer,
-               int sessionId);
+               int sessionId,
+               int uid);
 
     void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer);
     void timedYieldSilence_l(uint32_t numFrames,
@@ -255,7 +262,8 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                size_t frameCount);
+                                size_t frameCount,
+                                int uid);
     virtual             ~OutputTrack();
 
     virtual status_t    start(AudioSystem::sync_event_t event =

services/audioflinger/RecordTracks.h

diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index a4e1810..3ec9889 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -28,7 +28,8 @@
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
                                 size_t frameCount,
-                                int sessionId);
+                                int sessionId,
+                                int uid);
     virtual             ~RecordTrack();
 
     virtual status_t    start(AudioSystem::sync_event_t event, int triggerSession);
@@ -44,7 +45,10 @@
                                                 return tmp; }
 
     static  void        appendDumpHeader(String8& result);
-            void        dump(char* buffer, size_t size);
+            void        dump(char* buffer, size_t size, bool active);
+
+            void        handleSyncStartEvent(const sp<SyncEvent>& event);
+            void        clearSyncStartEvent();
 
 private:
     friend class AudioFlinger;  // for mState
@@ -58,4 +62,33 @@
     // releaseBuffer() not overridden
 
     bool                mOverflow;  // overflow on most recent attempt to fill client buffer
+
+           // updated by RecordThread::readInputParameters_l()
+            AudioResampler                      *mResampler;
+
+            // interleaved stereo pairs of fixed-point signed Q19.12
+            int32_t                             *mRsmpOutBuffer;
+            // current allocated frame count for the above, which may be larger than needed
+            size_t                              mRsmpOutFrameCount;
+
+            size_t                              mRsmpInUnrel;   // unreleased frames remaining from
+                                                                // most recent getNextBuffer
+                                                                // for debug only
+
+            // rolling counter that is never cleared
+            int32_t                             mRsmpInFront;   // next available frame
+
+            AudioBufferProvider::Buffer mSink;  // references client's buffer sink in shared memory
+
+            // sync event triggering actual audio capture. Frames read before this event will
+            // be dropped and therefore not read by the application.
+            sp<SyncEvent>                       mSyncStartEvent;
+
+            // number of captured frames to drop after the start sync event has been received.
+            // when < 0, maximum frames to drop before starting capture even if sync event is
+            // not received
+            ssize_t                             mFramesToDrop;
+
+            // used by resampler to find source frames
+            ResamplerBufferProvider *mResamplerBufferProvider;
 };

services/audioflinger/StateQueue.cpp

diff --git a/services/audioflinger/StateQueue.cpp b/services/audioflinger/StateQueue.cpp
index c2d3bbd..48399c0 100644
--- a/services/audioflinger/StateQueue.cpp
+++ b/services/audioflinger/StateQueue.cpp
@@ -58,7 +58,11 @@
 
 template<typename T> const T* StateQueue<T>::poll()
 {
+#ifdef __LP64__
+    const T *next = (const T *) android_atomic_acquire_load64((volatile int64_t *) &mNext);
+#else
     const T *next = (const T *) android_atomic_acquire_load((volatile int32_t *) &mNext);
+#endif
     if (next != mCurrent) {
         mAck = next;    // no additional barrier needed
         mCurrent = next;
@@ -140,7 +144,11 @@
         }
 
         // publish
+#ifdef __LP64__
+        android_atomic_release_store64((int64_t) mMutating, (volatile int64_t *) &mNext);
+#else
         android_atomic_release_store((int32_t) mMutating, (volatile int32_t *) &mNext);
+#endif
         mExpecting = mMutating;
 
         // copy with circular wraparound

services/audioflinger/Threads.cpp

diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index aca6c18..690d0d6 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -104,10 +104,10 @@
 // maximum divider applied to the active sleep time in the mixer thread loop
 static const uint32_t kMaxThreadSleepTimeShift = 2;
 
-// minimum normal mix buffer size, expressed in milliseconds rather than frames
-static const uint32_t kMinNormalMixBufferSizeMs = 20;
-// maximum normal mix buffer size
-static const uint32_t kMaxNormalMixBufferSizeMs = 24;
+// minimum normal sink buffer size, expressed in milliseconds rather than frames
+static const uint32_t kMinNormalSinkBufferSizeMs = 20;
+// maximum normal sink buffer size
+static const uint32_t kMaxNormalSinkBufferSizeMs = 24;
 
 // Offloaded output thread standby delay: allows track transition without going to standby
 static const nsecs_t kOffloadStandbyDelayNs = seconds(1);
@@ -135,12 +135,12 @@
 
 // IAudioFlinger::createTrack() reports back to client the total size of shared memory area
 // for the track.  The client then sub-divides this into smaller buffers for its use.
-// Currently the client uses double-buffering by default, but doesn't tell us about that.
-// So for now we just assume that client is double-buffered.
-// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or
-// N-buffering, so AudioFlinger could allocate the right amount of memory.
+// Currently the client uses N-buffering by default, but doesn't tell us about the value of N.
+// So for now we just assume that client is double-buffered for fast tracks.
+// FIXME It would be better for client to tell AudioFlinger the value of N,
+// so AudioFlinger could allocate the right amount of memory.
 // See the client's minBufCount and mNotificationFramesAct calculations for details.
-static const int kFastTrackMultiplier = 1;
+static const int kFastTrackMultiplier = 2;
 
 // ----------------------------------------------------------------------------
 
@@ -185,7 +185,11 @@
 {
 }
 
-void CpuStats::sample(const String8 &title) {
+void CpuStats::sample(const String8 &title
+#ifndef DEBUG_CPU_USAGE
+                __unused
+#endif
+        ) {
 #ifdef DEBUG_CPU_USAGE
     // get current thread's delta CPU time in wall clock ns
     double wcNs;
@@ -270,8 +274,10 @@
         mType(type),
         mAudioFlinger(audioFlinger),
         // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize
-        // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters()
+        // are set by PlaybackThread::readOutputParameters_l() or
+        // RecordThread::readInputParameters_l()
         mParamStatus(NO_ERROR),
+        //FIXME: mStandby should be true here. Is this some kind of hack?
         mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
         mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
         // mName will be set by concrete (non-virtual) subclass
@@ -420,7 +426,54 @@
     }
 }
 
-void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
+String8 channelMaskToString(audio_channel_mask_t mask, bool output) {
+    String8 s;
+    if (output) {
+        if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, ");
+        if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, ");
+        if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, ");
+        if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, ");
+        if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, ");
+        if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, ");
+        if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, ");
+        if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, ");
+        if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, ");
+        if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, ");
+        if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, ");
+        if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,");
+        if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, ");
+        if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, ");
+        if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, ");
+        if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, ");
+        if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " );
+        if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " );
+        if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown,  ");
+    } else {
+        if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, ");
+        if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, ");
+        if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, ");
+        if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, ");
+        if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, ");
+        if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, ");
+        if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, ");
+        if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, ");
+        if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, ");
+        if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, ");
+        if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, ");
+        if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, ");
+        if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, ");
+        if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, ");
+        if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown,  ");
+    }
+    int len = s.length();
+    if (s.length() > 2) {
+        char *str = s.lockBuffer(len);
+        s.unlockBuffer(len - 2);
+    }
+    return s;
+}
+
+void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused)
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
@@ -428,49 +481,43 @@
 
     bool locked = AudioFlinger::dumpTryLock(mLock);
     if (!locked) {
-        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
-        write(fd, buffer, strlen(buffer));
+        fdprintf(fd, "thread %p maybe dead locked\n", this);
     }
 
-    snprintf(buffer, SIZE, "io handle: %d\n", mId);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "TID: %d\n", getTid());
-    result.append(buffer);
-    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize);
-    result.append(buffer);
-
-    snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
-    result.append(buffer);
-    result.append(" Index Command");
-    for (size_t i = 0; i < mNewParameters.size(); ++i) {
-        snprintf(buffer, SIZE, "\n %02d    ", i);
-        result.append(buffer);
-        result.append(mNewParameters[i]);
+    fdprintf(fd, "  I/O handle: %d\n", mId);
+    fdprintf(fd, "  TID: %d\n", getTid());
+    fdprintf(fd, "  Standby: %s\n", mStandby ? "yes" : "no");
+    fdprintf(fd, "  Sample rate: %u\n", mSampleRate);
+    fdprintf(fd, "  HAL frame count: %zu\n", mFrameCount);
+    fdprintf(fd, "  HAL buffer size: %u bytes\n", mBufferSize);
+    fdprintf(fd, "  Channel Count: %u\n", mChannelCount);
+    fdprintf(fd, "  Channel Mask: 0x%08x (%s)\n", mChannelMask,
+            channelMaskToString(mChannelMask, mType != RECORD).string());
+    fdprintf(fd, "  Format: 0x%x (%s)\n", mFormat, formatToString(mFormat));
+    fdprintf(fd, "  Frame size: %zu\n", mFrameSize);
+    fdprintf(fd, "  Pending setParameters commands:");
+    size_t numParams = mNewParameters.size();
+    if (numParams) {
+        fdprintf(fd, "\n   Index Command");
+        for (size_t i = 0; i < numParams; ++i) {
+            fdprintf(fd, "\n   %02zu    ", i);
+            fdprintf(fd, mNewParameters[i]);
+        }
+        fdprintf(fd, "\n");
+    } else {
+        fdprintf(fd, " none\n");
     }
-
-    snprintf(buffer, SIZE, "\n\nPending config events: \n");
-    result.append(buffer);
-    for (size_t i = 0; i < mConfigEvents.size(); i++) {
-        mConfigEvents[i]->dump(buffer, SIZE);
-        result.append(buffer);
+    fdprintf(fd, "  Pending config events:");
+    size_t numConfig = mConfigEvents.size();
+    if (numConfig) {
+        for (size_t i = 0; i < numConfig; i++) {
+            mConfigEvents[i]->dump(buffer, SIZE);
+            fdprintf(fd, "\n    %s", buffer);
+        }
+        fdprintf(fd, "\n");
+    } else {
+        fdprintf(fd, " none\n");
     }
-    result.append("\n");
-
-    write(fd, result.string(), result.size());
 
     if (locked) {
         mLock.unlock();
@@ -483,10 +530,11 @@
     char buffer[SIZE];
     String8 result;
 
-    snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
+    size_t numEffectChains = mEffectChains.size();
+    snprintf(buffer, SIZE, "  %zu Effect Chains\n", numEffectChains);
     write(fd, buffer, strlen(buffer));
 
-    for (size_t i = 0; i < mEffectChains.size(); ++i) {
+    for (size_t i = 0; i < numEffectChains; ++i) {
         sp<EffectChain> chain = mEffectChains[i];
         if (chain != 0) {
             chain->dump(fd, args);
@@ -494,31 +542,49 @@
     }
 }
 
-void AudioFlinger::ThreadBase::acquireWakeLock()
+void AudioFlinger::ThreadBase::acquireWakeLock(int uid)
 {
     Mutex::Autolock _l(mLock);
-    acquireWakeLock_l();
+    acquireWakeLock_l(uid);
 }
 
-void AudioFlinger::ThreadBase::acquireWakeLock_l()
+String16 AudioFlinger::ThreadBase::getWakeLockTag()
 {
-    if (mPowerManager == 0) {
-        // use checkService() to avoid blocking if power service is not up yet
-        sp<IBinder> binder =
-            defaultServiceManager()->checkService(String16("power"));
-        if (binder == 0) {
-            ALOGW("Thread %s cannot connect to the power manager service", mName);
-        } else {
-            mPowerManager = interface_cast<IPowerManager>(binder);
-            binder->linkToDeath(mDeathRecipient);
-        }
+    switch (mType) {
+        case MIXER:
+            return String16("AudioMix");
+        case DIRECT:
+            return String16("AudioDirectOut");
+        case DUPLICATING:
+            return String16("AudioDup");
+        case RECORD:
+            return String16("AudioIn");
+        case OFFLOAD:
+            return String16("AudioOffload");
+        default:
+            ALOG_ASSERT(false);
+            return String16("AudioUnknown");
     }
+}
+
+void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid)
+{
+    getPowerManager_l();
     if (mPowerManager != 0) {
         sp<IBinder> binder = new BBinder();
-        status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
-                                                         binder,
-                                                         String16(mName),
-                                                         String16("media"));
+        status_t status;
+        if (uid >= 0) {
+            status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK,
+                    binder,
+                    getWakeLockTag(),
+                    String16("media"),
+                    uid);
+        } else {
+            status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
+                    binder,
+                    getWakeLockTag(),
+                    String16("media"));
+        }
         if (status == NO_ERROR) {
             mWakeLockToken = binder;
         }
@@ -543,6 +609,41 @@
     }
 }
 
+void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) {
+    Mutex::Autolock _l(mLock);
+    updateWakeLockUids_l(uids);
+}
+
+void AudioFlinger::ThreadBase::getPowerManager_l() {
+
+    if (mPowerManager == 0) {
+        // use checkService() to avoid blocking if power service is not up yet
+        sp<IBinder> binder =
+            defaultServiceManager()->checkService(String16("power"));
+        if (binder == 0) {
+            ALOGW("Thread %s cannot connect to the power manager service", mName);
+        } else {
+            mPowerManager = interface_cast<IPowerManager>(binder);
+            binder->linkToDeath(mDeathRecipient);
+        }
+    }
+}
+
+void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) {
+
+    getPowerManager_l();
+    if (mWakeLockToken == NULL) {
+        ALOGE("no wake lock to update!");
+        return;
+    }
+    if (mPowerManager != 0) {
+        sp<IBinder> binder = new BBinder();
+        status_t status;
+        status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array());
+        ALOGV("acquireWakeLock_l() %s status %d", mName, status);
+    }
+}
+
 void AudioFlinger::ThreadBase::clearPowerManager()
 {
     Mutex::Autolock _l(mLock);
@@ -550,7 +651,7 @@
     mPowerManager.clear();
 }
 
-void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
+void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused)
 {
     sp<ThreadBase> thread = mThread.promote();
     if (thread != 0) {
@@ -792,7 +893,10 @@
         }
         // create effect handle and connect it to effect module
         handle = new EffectHandle(effect, client, effectClient, priority);
-        lStatus = effect->addHandle(handle.get());
+        lStatus = handle->initCheck();
+        if (lStatus == OK) {
+            lStatus = effect->addHandle(handle.get());
+        }
         if (enabled != NULL) {
             *enabled = (int)effect->isEnabled();
         }
@@ -962,8 +1066,14 @@
                                              audio_devices_t device,
                                              type_t type)
     :   ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),
-        mNormalFrameCount(0), mMixBuffer(NULL),
+        mNormalFrameCount(0), mSinkBuffer(NULL),
+        mMixerBufferEnabled(false),
+        mMixerBuffer(NULL),
+        mMixerBufferSize(0),
+        mMixerBufferFormat(AUDIO_FORMAT_INVALID),
+        mMixerBufferValid(false),
         mSuspended(0), mBytesWritten(0),
+        mActiveTracksGeneration(0),
         // mStreamTypes[] initialized in constructor body
         mOutput(output),
         mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
@@ -1004,7 +1114,7 @@
         }
     }
 
-    readOutputParameters();
+    readOutputParameters_l();
 
     // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor
     // There is no AUDIO_STREAM_MIN, and ++ operator does not compile
@@ -1020,7 +1130,8 @@
 AudioFlinger::PlaybackThread::~PlaybackThread()
 {
     mAudioFlinger->unregisterWriter(mNBLogWriter);
-    delete[] mMixBuffer;
+    delete[] mSinkBuffer;
+    free(mMixerBuffer);
 }
 
 void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
@@ -1030,13 +1141,13 @@
     dumpEffectChains(fd, args);
 }
 
-void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
+void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused)
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
     String8 result;
 
-    result.appendFormat("Output thread %p stream volumes in dB:\n    ", this);
+    result.appendFormat("  Stream volumes in dB: ");
     for (int i = 0; i < AUDIO_STREAM_CNT; ++i) {
         const stream_type_t *st = &mStreamTypes[i];
         if (i > 0) {
@@ -1051,60 +1162,63 @@
     write(fd, result.string(), result.length());
     result.clear();
 
-    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
-    result.append(buffer);
-    Track::appendDumpHeader(result);
-    for (size_t i = 0; i < mTracks.size(); ++i) {
-        sp<Track> track = mTracks[i];
-        if (track != 0) {
-            track->dump(buffer, SIZE);
-            result.append(buffer);
-        }
-    }
-
-    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
-    result.append(buffer);
-    Track::appendDumpHeader(result);
-    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
-        sp<Track> track = mActiveTracks[i].promote();
-        if (track != 0) {
-            track->dump(buffer, SIZE);
-            result.append(buffer);
-        }
-    }
-    write(fd, result.string(), result.size());
-
     // These values are "raw"; they will wrap around.  See prepareTracks_l() for a better way.
     FastTrackUnderruns underruns = getFastTrackUnderruns(0);
-    fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n",
+    fdprintf(fd, "  Normal mixer raw underrun counters: partial=%u empty=%u\n",
             underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty);
+
+    size_t numtracks = mTracks.size();
+    size_t numactive = mActiveTracks.size();
+    fdprintf(fd, "  %d Tracks", numtracks);
+    size_t numactiveseen = 0;
+    if (numtracks) {
+        fdprintf(fd, " of which %d are active\n", numactive);
+        Track::appendDumpHeader(result);
+        for (size_t i = 0; i < numtracks; ++i) {
+            sp<Track> track = mTracks[i];
+            if (track != 0) {
+                bool active = mActiveTracks.indexOf(track) >= 0;
+                if (active) {
+                    numactiveseen++;
+                }
+                track->dump(buffer, SIZE, active);
+                result.append(buffer);
+            }
+        }
+    } else {
+        result.append("\n");
+    }
+    if (numactiveseen != numactive) {
+        // some tracks in the active list were not in the tracks list
+        snprintf(buffer, SIZE, "  The following tracks are in the active list but"
+                " not in the track list\n");
+        result.append(buffer);
+        Track::appendDumpHeader(result);
+        for (size_t i = 0; i < numactive; ++i) {
+            sp<Track> track = mActiveTracks[i].promote();
+            if (track != 0 && mTracks.indexOf(track) < 0) {
+                track->dump(buffer, SIZE, true);
+                result.append(buffer);
+            }
+        }
+    }
+
+    write(fd, result.string(), result.size());
+
 }
 
 void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
 {
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n",
-            ns2ms(systemTime() - mLastWriteTime));
-    result.append(buffer);
-    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
-    result.append(buffer);
-    write(fd, result.string(), result.size());
-    fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask);
+    fdprintf(fd, "\nOutput thread %p:\n", this);
+    fdprintf(fd, "  Normal frame count: %zu\n", mNormalFrameCount);
+    fdprintf(fd, "  Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
+    fdprintf(fd, "  Total writes: %d\n", mNumWrites);
+    fdprintf(fd, "  Delayed writes: %d\n", mNumDelayedWrites);
+    fdprintf(fd, "  Blocked in write: %s\n", mInWrite ? "yes" : "no");
+    fdprintf(fd, "  Suspend count: %d\n", mSuspended);
+    fdprintf(fd, "  Sink buffer : %p\n", mSinkBuffer);
+    fdprintf(fd, "  Mixer buffer: %p\n", mMixerBuffer);
+    fdprintf(fd, "  Fast track availMask=%#x\n", mFastTrackAvailMask);
 
     dumpBase(fd, args);
 }
@@ -1132,13 +1246,15 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        size_t frameCount,
+        size_t *pFrameCount,
         const sp<IMemory>& sharedBuffer,
         int sessionId,
         IAudioFlinger::track_flags_t *flags,
         pid_t tid,
+        int uid,
         status_t *status)
 {
+    size_t frameCount = *pFrameCount;
     sp<Track> track;
     status_t lStatus;
 
@@ -1159,7 +1275,7 @@
               (
                 (tid != -1) &&
                 ((frameCount == 0) ||
-                (frameCount >= (mFrameCount * kFastTrackMultiplier)))
+                (frameCount >= mFrameCount))
               )
             ) &&
             // PCM data
@@ -1167,10 +1283,8 @@
             // mono or stereo
             ( (channelMask == AUDIO_CHANNEL_OUT_MONO) ||
               (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) &&
-#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE
             // hardware sample rate
             (sampleRate == mSampleRate) &&
-#endif
             // normal mixer has an associated fast mixer
             hasFastMixer() &&
             // there are sufficient fast track slots available
@@ -1207,12 +1321,13 @@
         }
       }
     }
+    *pFrameCount = frameCount;
 
     if (mType == DIRECT) {
         if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
             if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
-                ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x "
-                        "for output %p with format %d",
+                ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x "
+                        "for output %p with format %#x",
                         sampleRate, format, channelMask, mOutput, mFormat);
                 lStatus = BAD_VALUE;
                 goto Exit;
@@ -1220,16 +1335,16 @@
         }
     } else if (mType == OFFLOAD) {
         if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
-            ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
-                    "for output %p with format %d",
+            ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \""
+                    "for output %p with format %#x",
                     sampleRate, format, channelMask, mOutput, mFormat);
             lStatus = BAD_VALUE;
             goto Exit;
         }
     } else {
         if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) {
-                ALOGE("createTrack_l() Bad parameter: format %d \""
-                        "for output %p with format %d",
+                ALOGE("createTrack_l() Bad parameter: format %#x \""
+                        "for output %p with format %#x",
                         format, mOutput, mFormat);
                 lStatus = BAD_VALUE;
                 goto Exit;
@@ -1270,17 +1385,18 @@
 
         if (!isTimed) {
             track = new Track(this, client, streamType, sampleRate, format,
-                    channelMask, frameCount, sharedBuffer, sessionId, *flags);
+                    channelMask, frameCount, sharedBuffer, sessionId, uid, *flags);
         } else {
             track = TimedTrack::create(this, client, streamType, sampleRate, format,
-                    channelMask, frameCount, sharedBuffer, sessionId);
+                    channelMask, frameCount, sharedBuffer, sessionId, uid);
         }
 
         // new Track always returns non-NULL,
         // but TimedTrack::create() is a factory that could fail by returning NULL
         lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
         if (lStatus != NO_ERROR) {
-            track.clear();
+            ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus);
+            // track must be cleared from the caller as the caller has the AF lock
             goto Exit;
         }
 
@@ -1411,6 +1527,9 @@
         track->mResetDone = false;
         track->mPresentationCompleteFrames = 0;
         mActiveTracks.add(track);
+        mWakeLockUids.add(track->uid());
+        mActiveTracksGeneration++;
+        mLatestActiveTrack = track;
         sp<EffectChain> chain = getEffectChain_l(track->sessionId());
         if (chain != 0) {
             ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(),
@@ -1421,9 +1540,7 @@
         status = NO_ERROR;
     }
 
-    ALOGV("signal playback thread");
-    broadcast_l();
-
+    onAddNewTrack_l();
     return status;
 }
 
@@ -1549,7 +1666,7 @@
 
 // static
 int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event,
-                                                void *param,
+                                                void *param __unused,
                                                 void *cookie)
 {
     AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie;
@@ -1568,7 +1685,7 @@
     return 0;
 }
 
-void AudioFlinger::PlaybackThread::readOutputParameters()
+void AudioFlinger::PlaybackThread::readOutputParameters_l()
 {
     // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL
     mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
@@ -1583,10 +1700,10 @@
     mChannelCount = popcount(mChannelMask);
     mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
     if (!audio_is_valid_format(mFormat)) {
-        LOG_FATAL("HAL format %d not valid for output", mFormat);
+        LOG_FATAL("HAL format %#x not valid for output", mFormat);
     }
     if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) {
-        LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT",
+        LOG_FATAL("HAL format %#x not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT",
                 mFormat);
     }
     mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
@@ -1602,15 +1719,16 @@
         if (mOutput->stream->set_callback(mOutput->stream,
                                       AudioFlinger::PlaybackThread::asyncCallback, this) == 0) {
             mUseAsyncWrite = true;
+            mCallbackThread = new AudioFlinger::AsyncCallbackThread(this);
         }
     }
 
-    // Calculate size of normal mix buffer relative to the HAL output buffer size
+    // Calculate size of normal sink buffer relative to the HAL output buffer size
     double multiplier = 1.0;
     if (mType == MIXER && (kUseFastMixer == FastMixer_Static ||
             kUseFastMixer == FastMixer_Dynamic)) {
-        size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000;
-        size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000;
+        size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000;
+        size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000;
         // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer
         minNormalFrameCount = (minNormalFrameCount + 15) & ~15;
         maxNormalFrameCount = maxNormalFrameCount & ~15;
@@ -1628,7 +1746,7 @@
             }
         } else {
             // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL
-            // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast
+            // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast
             // track, but we sometimes have to do this to satisfy the maximum frame count
             // constraint)
             // FIXME this rounding up should not be done if no HAL SRC
@@ -1644,18 +1762,29 @@
     mNormalFrameCount = multiplier * mFrameCount;
     // round up to nearest 16 frames to satisfy AudioMixer
     mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
-    ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount,
+    ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount,
             mNormalFrameCount);
 
-    delete[] mMixBuffer;
+    delete[] mSinkBuffer;
     size_t normalBufferSize = mNormalFrameCount * mFrameSize;
-    // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1)
-    mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1];
-    memset(mMixBuffer, 0, normalBufferSize);
+    // For historical reasons mSinkBuffer is int16_t[], but mFrameSize can be odd (such as 1)
+    mSinkBuffer = new int16_t[(normalBufferSize + 1) >> 1];
+    memset(mSinkBuffer, 0, normalBufferSize);
+
+    // We resize the mMixerBuffer according to the requirements of the sink buffer which
+    // drives the output.
+    free(mMixerBuffer);
+    mMixerBuffer = NULL;
+    if (mMixerBufferEnabled) {
+        mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT.
+        mMixerBufferSize = mNormalFrameCount * mChannelCount
+                * audio_bytes_per_sample(mMixerBufferFormat);
+        (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize);
+    }
 
     // force reconfiguration of effect chains and engines to take new buffer size and audio
     // parameters into account
-    // Note that mLock is not held when readOutputParameters() is called from the constructor
+    // Note that mLock is not held when readOutputParameters_l() is called from the constructor
     // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
     // matter.
     // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
@@ -1666,7 +1795,7 @@
 }
 
 
-status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames)
+status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
 {
     if (halFrames == NULL || dspFrames == NULL) {
         return BAD_VALUE;
@@ -1684,7 +1813,11 @@
         *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0;
         return NO_ERROR;
     } else {
-        return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
+        status_t status;
+        uint32_t frames;
+        status = mOutput->stream->get_render_position(mOutput->stream, &frames);
+        *dspFrames = (size_t)frames;
+        return status;
     }
 }
 
@@ -1843,7 +1976,7 @@
                         (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
             }
         }
-        ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count);
+        ssize_t framesWritten = mNormalSink->write(mSinkBuffer + offset, count);
         ATRACE_END();
         if (framesWritten > 0) {
             bytesWritten = framesWritten << mBitShift;
@@ -1861,7 +1994,7 @@
     // otherwise use the HAL / AudioStreamOut directly
     } else {
         // Direct output and offload threads
-        size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t);
+        size_t offset = (mCurrentWriteLength - mBytesRemaining);
         if (mUseAsyncWrite) {
             ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request");
             mWriteAckSequence += 2;
@@ -1872,7 +2005,7 @@
         // FIXME We should have an implementation of timestamps for direct output threads.
         // They are used e.g for multichannel PCM playback over HDMI.
         bytesWritten = mOutput->stream->write(mOutput->stream,
-                                                   mMixBuffer + offset, mBytesRemaining);
+                                                   (char *)mSinkBuffer + offset, mBytesRemaining);
         if (mUseAsyncWrite &&
                 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
             // do not wait for async callback in case of error of full write
@@ -1884,7 +2017,7 @@
 
     mNumWrites++;
     mInWrite = false;
-
+    mStandby = false;
     return bytesWritten;
 }
 
@@ -1911,7 +2044,7 @@
 
 /*
 The derived values that are cached:
- - mixBufferSize from frame count * frame size
+ - mSinkBufferSize from frame count * frame size
  - activeSleepTime from activeSleepTimeUs()
  - idleSleepTime from idleSleepTimeUs()
  - standbyDelay from mActiveSleepTimeUs (DIRECT only)
@@ -1930,7 +2063,7 @@
 
 void AudioFlinger::PlaybackThread::cacheParameters_l()
 {
-    mixBufferSize = mNormalFrameCount * mFrameSize;
+    mSinkBufferSize = mNormalFrameCount * mFrameSize;
     activeSleepTime = activeSleepTimeUs();
     idleSleepTime = idleSleepTimeUs();
 }
@@ -1953,13 +2086,13 @@
 status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
 {
     int session = chain->sessionId();
-    int16_t *buffer = mMixBuffer;
+    int16_t *buffer = mSinkBuffer;
     bool ownsBuffer = false;
 
     ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
     if (session > 0) {
         // Only one effect chain can be present in direct output thread and it uses
-        // the mix buffer as input
+        // the sink buffer as input
         if (mType != DIRECT) {
             size_t numSamples = mNormalFrameCount * mChannelCount;
             buffer = new int16_t[numSamples];
@@ -1993,7 +2126,7 @@
     }
 
     chain->setInBuffer(buffer, ownsBuffer);
-    chain->setOutBuffer(mMixBuffer);
+    chain->setOutBuffer(mSinkBuffer);
     // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
     // chains list in order to be processed last as it contains output stage effects
     // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
@@ -2043,7 +2176,7 @@
             for (size_t i = 0; i < mTracks.size(); ++i) {
                 sp<Track> track = mTracks[i];
                 if (session == track->sessionId()) {
-                    track->setMainBuffer(mMixBuffer);
+                    track->setMainBuffer(mSinkBuffer);
                     chain->decTrackCnt();
                 }
             }
@@ -2106,6 +2239,8 @@
     // FIXME could this be made local to while loop?
     writeFrames = 0;
 
+    int lastGeneration = 0;
+
     cacheParameters_l();
     sleepTime = idleSleepTime;
 
@@ -2162,6 +2297,8 @@
                     break;
                 }
                 releaseWakeLock_l();
+                mWakeLockUids.clear();
+                mActiveTracksGeneration++;
                 ALOGV("wait async completion");
                 mWaitWorkCV.wait(mLock);
                 ALOGV("async completion/wake");
@@ -2192,6 +2329,8 @@
                     }
 
                     releaseWakeLock_l();
+                    mWakeLockUids.clear();
+                    mActiveTracksGeneration++;
                     // wait until we have something to do...
                     ALOGV("%s going to sleep", myName.string());
                     mWaitWorkCV.wait(mLock);
@@ -2216,41 +2355,74 @@
             // mMixerStatusIgnoringFastTracks is also updated internally
             mMixerStatus = prepareTracks_l(&tracksToRemove);
 
+            // compare with previously applied list
+            if (lastGeneration != mActiveTracksGeneration) {
+                // update wakelock
+                updateWakeLockUids_l(mWakeLockUids);
+                lastGeneration = mActiveTracksGeneration;
+            }
+
             // prevent any changes in effect chain list and in each effect chain
             // during mixing and effect process as the audio buffers could be deleted
             // or modified if an effect is created or deleted
             lockEffectChains_l(effectChains);
-        }
+        } // mLock scope ends
 
         if (mBytesRemaining == 0) {
             mCurrentWriteLength = 0;
             if (mMixerStatus == MIXER_TRACKS_READY) {
                 // threadLoop_mix() sets mCurrentWriteLength
                 threadLoop_mix();
+
+                // Merge mMixerBuffer data into mSinkBuffer
+                // This is done pre-effects computation; if effects change to
+                // support higher precision, this needs to move.
+                //
+                // mMixerBufferValid is only set true by MixerThread::prepareTracks_l().
+                if (mMixerBufferValid) {
+                    if (mMixerBufferFormat == AUDIO_FORMAT_PCM_FLOAT) {
+                        memcpy_to_i16_from_float(mSinkBuffer,
+                                reinterpret_cast<float*>(mMixerBuffer),
+                                mNormalFrameCount * mChannelCount);
+                    } else { // mMixerBufferFormat == AUDIO_FORMAT_PCM_16_BIT
+                        memcpy(mSinkBuffer,
+                                mMixerBuffer,
+                                mNormalFrameCount * mChannelCount * sizeof(int16_t));
+                    }
+                }
             } else if ((mMixerStatus != MIXER_DRAIN_TRACK)
                         && (mMixerStatus != MIXER_DRAIN_ALL)) {
                 // threadLoop_sleepTime sets sleepTime to 0 if data
                 // must be written to HAL
                 threadLoop_sleepTime();
                 if (sleepTime == 0) {
-                    mCurrentWriteLength = mixBufferSize;
+                    mCurrentWriteLength = mSinkBufferSize;
                 }
             }
             mBytesRemaining = mCurrentWriteLength;
             if (isSuspended()) {
                 sleepTime = suspendSleepTimeUs();
                 // simulate write to HAL when suspended
-                mBytesWritten += mixBufferSize;
+                mBytesWritten += mSinkBufferSize;
                 mBytesRemaining = 0;
             }
 
             // only process effects if we're going to write
-            if (sleepTime == 0) {
+            if (sleepTime == 0 && mType != OFFLOAD) {
                 for (size_t i = 0; i < effectChains.size(); i ++) {
                     effectChains[i]->process_l();
                 }
             }
         }
+        // Process effect chains for offloaded thread even if no audio
+        // was read from audio track: process only updates effect state
+        // and thus does have to be synchronized with audio writes but may have
+        // to be called while waiting for async write callback
+        if (mType == OFFLOAD) {
+            for (size_t i = 0; i < effectChains.size(); i ++) {
+                effectChains[i]->process_l();
+            }
+        }
 
         // enable changes in effect chain
         unlockEffectChains(effectChains);
@@ -2270,22 +2442,21 @@
                         (mMixerStatus == MIXER_DRAIN_ALL)) {
                     threadLoop_drain();
                 }
-if (mType == MIXER) {
-                // write blocked detection
-                nsecs_t now = systemTime();
-                nsecs_t delta = now - mLastWriteTime;
-                if (!mStandby && delta > maxPeriod) {
-                    mNumDelayedWrites++;
-                    if ((now - lastWarning) > kWarningThrottleNs) {
-                        ATRACE_NAME("underrun");
-                        ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
-                                ns2ms(delta), mNumDelayedWrites, this);
-                        lastWarning = now;
+                if (mType == MIXER) {
+                    // write blocked detection
+                    nsecs_t now = systemTime();
+                    nsecs_t delta = now - mLastWriteTime;
+                    if (!mStandby && delta > maxPeriod) {
+                        mNumDelayedWrites++;
+                        if ((now - lastWarning) > kWarningThrottleNs) {
+                            ATRACE_NAME("underrun");
+                            ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
+                                    ns2ms(delta), mNumDelayedWrites, this);
+                            lastWarning = now;
+                        }
                     }
                 }
-}
 
-                mStandby = false;
             } else {
                 usleep(sleepTime);
             }
@@ -2321,6 +2492,8 @@
     }
 
     releaseWakeLock();
+    mWakeLockUids.clear();
+    mActiveTracksGeneration++;
 
     ALOGV("Thread %p type %d exiting", this, mType);
     return false;
@@ -2334,6 +2507,8 @@
         for (size_t i=0 ; i<count ; i++) {
             const sp<Track>& track = tracksToRemove.itemAt(i);
             mActiveTracks.remove(track);
+            mWakeLockUids.remove(track->uid());
+            mActiveTracksGeneration++;
             ALOGV("removeTracks_l removing track on session %d", track->sessionId());
             sp<EffectChain> chain = getEffectChain_l(track->sessionId());
             if (chain != 0) {
@@ -2632,12 +2807,6 @@
     PlaybackThread::threadLoop_standby();
 }
 
-// Empty implementation for standard mixer
-// Overridden for offloaded playback
-void AudioFlinger::PlaybackThread::flushOutput_l()
-{
-}
-
 bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l()
 {
     return false;
@@ -2669,6 +2838,12 @@
     }
 }
 
+void AudioFlinger::PlaybackThread::onAddNewTrack_l()
+{
+    ALOGV("signal playback thread");
+    broadcast_l();
+}
+
 void AudioFlinger::MixerThread::threadLoop_mix()
 {
     // obtain the presentation timestamp of the next output buffer
@@ -2687,7 +2862,7 @@
 
     // mix buffers...
     mAudioMixer->process(pts);
-    mCurrentWriteLength = mixBufferSize;
+    mCurrentWriteLength = mSinkBufferSize;
     // increase sleep time progressively when application underrun condition clears.
     // Only increase sleep time if the mixer is ready for two consecutive times to avoid
     // that a steady state of alternating ready/not ready conditions keeps the sleep time
@@ -2721,7 +2896,7 @@
             sleepTime = idleSleepTime;
         }
     } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) {
-        memset(mMixBuffer, 0, mixBufferSize);
+        memset(mSinkBuffer, 0, mSinkBufferSize);
         sleepTime = 0;
         ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED),
                 "anticipated start");
@@ -2768,6 +2943,8 @@
         state = sq->begin();
     }
 
+    mMixerBufferValid = false;  // mMixerBuffer has no valid data until appropriate tracks found.
+
     for (size_t i=0 ; i<count ; i++) {
         const sp<Track> t = mActiveTracks[i].promote();
         if (t == 0) {
@@ -2896,7 +3073,6 @@
                     VolumeProvider *vp = track;
                     fastTrack->mBufferProvider = eabp;
                     fastTrack->mVolumeProvider = vp;
-                    fastTrack->mSampleRate = track->mSampleRate;
                     fastTrack->mChannelMask = track->mChannelMask;
                     fastTrack->mGeneration++;
                     state->mTrackMask |= 1 << j;
@@ -2949,25 +3125,20 @@
             // add frames already consumed but not yet released by the resampler
             // because mAudioTrackServerProxy->framesReady() will include these frames
             desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
+#if 0
             // the minimum track buffer size is normally twice the number of frames necessary
             // to fill one buffer and the resampler should not leave more than one buffer worth
             // of unreleased frames after each pass, but just in case...
             ALOG_ASSERT(desiredFrames <= cblk->frameCount_);
+#endif
         }
         uint32_t minFrames = 1;
         if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
                 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
             minFrames = desiredFrames;
         }
-        // It's not safe to call framesReady() for a static buffer track, so assume it's ready
-        size_t framesReady;
-        if (track->sharedBuffer() == 0) {
-            framesReady = track->framesReady();
-        } else if (track->isStopped()) {
-            framesReady = 0;
-        } else {
-            framesReady = 1;
-        }
+
+        size_t framesReady = track->framesReady();
         if ((framesReady >= minFrames) && track->isReady() &&
                 !track->isPaused() && !track->isTerminated())
         {
@@ -2975,10 +3146,11 @@
 
             mixedTracks++;
 
-            // track->mainBuffer() != mMixBuffer means there is an effect chain
-            // connected to the track
+            // track->mainBuffer() != mSinkBuffer or mMixerBuffer means
+            // there is an effect chain connected to the track
             chain.clear();
-            if (track->mainBuffer() != mMixBuffer) {
+            if (track->mainBuffer() != mSinkBuffer &&
+                    track->mainBuffer() != mMixerBuffer) {
                 chain = getEffectChain_l(track->sessionId());
                 // Delegate volume control to effect in track effect chain if needed
                 if (chain != 0) {
@@ -3080,9 +3252,9 @@
             mAudioMixer->setBufferProvider(name, track);
             mAudioMixer->enable(name);
 
-            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl);
-            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr);
-            mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va);
+            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl);
+            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr);
+            mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va);
             mAudioMixer->setParameter(
                 name,
                 AudioMixer::TRACK,
@@ -3090,7 +3262,7 @@
             mAudioMixer->setParameter(
                 name,
                 AudioMixer::TRACK,
-                AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
+                AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask());
             // limit track sample rate to 2 x output sample rate, which changes at re-configuration
             uint32_t maxSampleRate = mSampleRate * 2;
             uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate();
@@ -3103,11 +3275,41 @@
                 name,
                 AudioMixer::RESAMPLE,
                 AudioMixer::SAMPLE_RATE,
-                (void *)reqSampleRate);
-            mAudioMixer->setParameter(
-                name,
-                AudioMixer::TRACK,
-                AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
+                (void *)(uintptr_t)reqSampleRate);
+            /*
+             * Select the appropriate output buffer for the track.
+             *
+             * For tracks with effects, only mSinkBuffer can be used (at this time).
+             *
+             * Other tracks can use mMixerBuffer for higher precision
+             * channel accumulation.  If this buffer is enabled
+             * (mMixerBufferEnabled true), then selected tracks will accumulate
+             * into it.
+             *
+             */
+            if (mMixerBufferEnabled
+                    && (track->mainBuffer() == mSinkBuffer
+                            || track->mainBuffer() == mMixerBuffer)) {
+                mAudioMixer->setParameter(
+                        name,
+                        AudioMixer::TRACK,
+                        AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
+                mAudioMixer->setParameter(
+                        name,
+                        AudioMixer::TRACK,
+                        AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer);
+                // TODO: override track->mainBuffer()?
+                mMixerBufferValid = true;
+            } else {
+                mAudioMixer->setParameter(
+                        name,
+                        AudioMixer::TRACK,
+                        AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT);
+                mAudioMixer->setParameter(
+                        name,
+                        AudioMixer::TRACK,
+                        AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
+            }
             mAudioMixer->setParameter(
                 name,
                 AudioMixer::TRACK,
@@ -3221,13 +3423,25 @@
     // remove all the tracks that need to be...
     removeTracks_l(*tracksToRemove);
 
-    // mix buffer must be cleared if all tracks are connected to an
-    // effect chain as in this case the mixer will not write to
-    // mix buffer and track effects will accumulate into it
+    // sink or mix buffer must be cleared if all tracks are connected to an
+    // effect chain as in this case the mixer will not write to the sink or mix buffer
+    // and track effects will accumulate into it
     if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
             (mixedTracks == 0 && fastTracks > 0))) {
         // FIXME as a performance optimization, should remember previous zero status
-        memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));
+        if (mMixerBufferValid) {
+            memset(mMixerBuffer, 0, mMixerBufferSize);
+            // TODO: In testing, mSinkBuffer below need not be cleared because
+            // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer
+            // after mixing.
+            //
+            // To enforce this guarantee:
+            // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
+            // (mixedTracks == 0 && fastTracks > 0))
+            // must imply MIXER_TRACKS_READY.
+            // Later, we may clear buffers regardless, and skip much of this logic.
+        }
+        memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));
     }
 
     // if any fast tracks, then status is ready
@@ -3352,7 +3566,7 @@
                                                        keyValuePair.string());
             }
             if (status == NO_ERROR && reconfig) {
-                readOutputParameters();
+                readOutputParameters_l();
                 delete mAudioMixer;
                 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
                 for (size_t i = 0; i < mTracks.size() ; i++) {
@@ -3397,9 +3611,7 @@
 
     PlaybackThread::dumpInternals(fd, args);
 
-    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
-    result.append(buffer);
-    write(fd, result.string(), result.size());
+    fdprintf(fd, "  AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames());
 
     // Make a non-atomic copy of fast mixer dump state so it won't change underneath us
     const FastMixerDumpState copy(mFastMixerDumpState);
@@ -3531,6 +3743,12 @@
 
         Track* const track = t.get();
         audio_track_cblk_t* cblk = track->cblk();
+        // Only consider last track started for volume and mixer state control.
+        // In theory an older track could underrun and restart after the new one starts
+        // but as we only care about the transition phase between two tracks on a
+        // direct output, it is not a problem to ignore the underrun case.
+        sp<Track> l = mLatestActiveTrack.promote();
+        bool last = l.get() == track;
 
         // The first time a track is added we wait
         // for all its buffers to be filled before processing it
@@ -3540,11 +3758,6 @@
         } else {
             minFrames = 1;
         }
-        // Only consider last track started for volume and mixer state control.
-        // This is the last entry in mActiveTracks unless a track underruns.
-        // As we only care about the transition phase between two tracks on a
-        // direct output, it is not a problem to ignore the underrun case.
-        bool last = (i == (count - 1));
 
         if ((track->framesReady() >= minFrames) && track->isReady() &&
                 !track->isPaused() && !track->isTerminated())
@@ -3571,7 +3784,7 @@
         } else {
             // clear effect chain input buffer if the last active track started underruns
             // to avoid sending previous audio buffer again to effects
-            if (!mEffectChains.isEmpty() && (i == (count -1))) {
+            if (!mEffectChains.isEmpty() && last) {
                 mEffectChains[0]->clearInputBuffer();
             }
 
@@ -3583,7 +3796,8 @@
                 // TODO: implement behavior for compressed audio
                 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
                 size_t framesWritten = mBytesWritten / mFrameSize;
-                if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
+                if (mStandby || !last ||
+                        track->presentationComplete(framesWritten, audioHALFrames)) {
                     if (track->isStopped()) {
                         track->reset();
                     }
@@ -3596,6 +3810,9 @@
                 if (--(track->mRetryCount) <= 0) {
                     ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
                     tracksToRemove->add(track);
+                    // indicate to client process that the track was disabled because of underrun;
+                    // it will then automatically call start() when data is available
+                    android_atomic_or(CBLK_DISABLED, &cblk->mFlags);
                 } else if (last) {
                     mixerStatus = MIXER_TRACKS_ENABLED;
                 }
@@ -3612,7 +3829,7 @@
 void AudioFlinger::DirectOutputThread::threadLoop_mix()
 {
     size_t frameCount = mFrameCount;
-    int8_t *curBuf = (int8_t *)mMixBuffer;
+    int8_t *curBuf = (int8_t *)mSinkBuffer;
     // output audio to hardware
     while (frameCount) {
         AudioBufferProvider::Buffer buffer;
@@ -3627,7 +3844,7 @@
         curBuf += buffer.frameCount * mFrameSize;
         mActiveTrack->releaseBuffer(&buffer);
     }
-    mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer;
+    mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer;
     sleepTime = 0;
     standbyTime = systemTime() + standbyDelay;
     mActiveTrack.clear();
@@ -3642,20 +3859,20 @@
             sleepTime = idleSleepTime;
         }
     } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
-        memset(mMixBuffer, 0, mFrameCount * mFrameSize);
+        memset(mSinkBuffer, 0, mFrameCount * mFrameSize);
         sleepTime = 0;
     }
 }
 
 // getTrackName_l() must be called with ThreadBase::mLock held
-int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask,
-        int sessionId)
+int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused,
+        int sessionId __unused)
 {
     return 0;
 }
 
 // deleteTrackName_l() must be called with ThreadBase::mLock held
-void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
+void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused)
 {
 }
 
@@ -3691,7 +3908,7 @@
                                                        keyValuePair.string());
             }
             if (status == NO_ERROR && reconfig) {
-                readOutputParameters();
+                readOutputParameters_l();
                 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
             }
         }
@@ -3756,9 +3973,9 @@
 // ----------------------------------------------------------------------------
 
 AudioFlinger::AsyncCallbackThread::AsyncCallbackThread(
-        const sp<AudioFlinger::OffloadThread>& offloadThread)
+        const wp<AudioFlinger::PlaybackThread>& playbackThread)
     :   Thread(false /*canCallJava*/),
-        mOffloadThread(offloadThread),
+        mPlaybackThread(playbackThread),
         mWriteAckSequence(0),
         mDrainSequence(0)
 {
@@ -3781,7 +3998,12 @@
 
         {
             Mutex::Autolock _l(mLock);
-            mWaitWorkCV.wait(mLock);
+            while (!((mWriteAckSequence & 1) ||
+                     (mDrainSequence & 1) ||
+                     exitPending())) {
+                mWaitWorkCV.wait(mLock);
+            }
+
             if (exitPending()) {
                 break;
             }
@@ -3793,13 +4015,13 @@
             mDrainSequence &= ~1;
         }
         {
-            sp<AudioFlinger::OffloadThread> offloadThread = mOffloadThread.promote();
-            if (offloadThread != 0) {
+            sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote();
+            if (playbackThread != 0) {
                 if (writeAckSequence & 1) {
-                    offloadThread->resetWriteBlocked(writeAckSequence >> 1);
+                    playbackThread->resetWriteBlocked(writeAckSequence >> 1);
                 }
                 if (drainSequence & 1) {
-                    offloadThread->resetDraining(drainSequence >> 1);
+                    playbackThread->resetDraining(drainSequence >> 1);
                 }
             }
         }
@@ -3855,14 +4077,11 @@
         AudioStreamOut* output, audio_io_handle_t id, uint32_t device)
     :   DirectOutputThread(audioFlinger, output, id, device, OFFLOAD),
         mHwPaused(false),
+        mFlushPending(false),
         mPausedBytesRemaining(0)
 {
-    mCallbackThread = new AudioFlinger::AsyncCallbackThread(this);
-}
-
-AudioFlinger::OffloadThread::~OffloadThread()
-{
-    mPreviousTrack.clear();
+    //FIXME: mStandby should be set to true by ThreadBase constructor
+    mStandby = true;
 }
 
 void AudioFlinger::OffloadThread::threadLoop_exit()
@@ -3899,24 +4118,24 @@
         }
         Track* const track = t.get();
         audio_track_cblk_t* cblk = track->cblk();
-        if (mPreviousTrack != NULL) {
-            if (t != mPreviousTrack) {
-                // Flush any data still being written from last track
-                mBytesRemaining = 0;
-                if (mPausedBytesRemaining) {
-                    // Last track was paused so we also need to flush saved
-                    // mixbuffer state and invalidate track so that it will
-                    // re-submit that unwritten data when it is next resumed
-                    mPausedBytesRemaining = 0;
-                    // Invalidate is a bit drastic - would be more efficient
-                    // to have a flag to tell client that some of the
-                    // previously written data was lost
-                    mPreviousTrack->invalidate();
-                }
-            }
+        // Only consider last track started for volume and mixer state control.
+        // In theory an older track could underrun and restart after the new one starts
+        // but as we only care about the transition phase between two tracks on a
+        // direct output, it is not a problem to ignore the underrun case.
+        sp<Track> l = mLatestActiveTrack.promote();
+        bool last = l.get() == track;
+
+        if (track->isInvalid()) {
+            ALOGW("An invalidated track shouldn't be in active list");
+            tracksToRemove->add(track);
+            continue;
         }
-        mPreviousTrack = t;
-        bool last = (i == (count - 1));
+
+        if (track->mState == TrackBase::IDLE) {
+            ALOGW("An idle track shouldn't be in active list");
+            continue;
+        }
+
         if (track->isPausing()) {
             track->setPaused();
             if (last) {
@@ -3935,6 +4154,11 @@
                 mBytesRemaining = 0;    // stop writing
             }
             tracksToRemove->add(track);
+        } else if (track->isFlushPending()) {
+            track->flushAck();
+            if (last) {
+                mFlushPending = true;
+            }
         } else if (track->framesReady() && track->isReady() &&
                 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) {
             ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer);
@@ -3964,6 +4188,30 @@
             }
 
             if (last) {
+                sp<Track> previousTrack = mPreviousTrack.promote();
+                if (previousTrack != 0) {
+                    if (track != previousTrack.get()) {
+                        // Flush any data still being written from last track
+                        mBytesRemaining = 0;
+                        if (mPausedBytesRemaining) {
+                            // Last track was paused so we also need to flush saved
+                            // mixbuffer state and invalidate track so that it will
+                            // re-submit that unwritten data when it is next resumed
+                            mPausedBytesRemaining = 0;
+                            // Invalidate is a bit drastic - would be more efficient
+                            // to have a flag to tell client that some of the
+                            // previously written data was lost
+                            previousTrack->invalidate();
+                        }
+                        // flush data already sent to the DSP if changing audio session as audio
+                        // comes from a different source. Also invalidate previous track to force a
+                        // seek when resuming.
+                        if (previousTrack->sessionId() != track->sessionId()) {
+                            previousTrack->invalidate();
+                        }
+                    }
+                }
+                mPreviousTrack = track;
                 // reset retry count
                 track->mRetryCount = kMaxTrackRetriesOffload;
                 mActiveTrack = t;
@@ -3980,22 +4228,27 @@
                     // has been written
                     ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2");
                     track->mState = TrackBase::STOPPING_2; // so presentation completes after drain
-                    if (last) {
-                        sleepTime = 0;
-                        standbyTime = systemTime() + standbyDelay;
-                        mixerStatus = MIXER_DRAIN_TRACK;
-                        mDrainSequence += 2;
+                    // do not drain if no data was ever sent to HAL (mStandby == true)
+                    if (last && !mStandby) {
+                        // do not modify drain sequence if we are already draining. This happens
+                        // when resuming from pause after drain.
+                        if ((mDrainSequence & 1) == 0) {
+                            sleepTime = 0;
+                            standbyTime = systemTime() + standbyDelay;
+                            mixerStatus = MIXER_DRAIN_TRACK;
+                            mDrainSequence += 2;
+                        }
                         if (mHwPaused) {
                             // It is possible to move from PAUSED to STOPPING_1 without
                             // a resume so we must ensure hardware is running
-                            mOutput->stream->resume(mOutput->stream);
+                            doHwResume = true;
                             mHwPaused = false;
                         }
                     }
                 }
             } else if (track->isStopping_2()) {
-                // Drain has completed, signal presentation complete
-                if (!(mDrainSequence & 1) || !last) {
+                // Drain has completed or we are in standby, signal presentation complete
+                if (!(mDrainSequence & 1) || !last || mStandby) {
                     track->mState = TrackBase::STOPPED;
                     size_t audioHALFrames =
                             (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
@@ -4012,6 +4265,9 @@
                     ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list",
                           track->name());
                     tracksToRemove->add(track);
+                    // indicate to client process that the track was disabled because of underrun;
+                    // it will then automatically call start() when data is available
+                    android_atomic_or(CBLK_DISABLED, &cblk->mFlags);
                 } else if (last){
                     mixerStatus = MIXER_TRACKS_ENABLED;
                 }
@@ -4021,15 +4277,18 @@
         processVolume_l(track, last);
     }
 
-    // make sure the pause/flush/resume sequence is executed in the right order
-    if (doHwPause) {
+    // make sure the pause/flush/resume sequence is executed in the right order.
+    // If a flush is pending and a track is active but the HW is not paused, force a HW pause
+    // before flush and then resume HW. This can happen in case of pause/flush/resume
+    // if resume is received before pause is executed.
+    if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) {
         mOutput->stream->pause(mOutput->stream);
     }
     if (mFlushPending) {
         flushHw_l();
         mFlushPending = false;
     }
-    if (doHwResume) {
+    if (!mStandby && doHwResume) {
         mOutput->stream->resume(mOutput->stream);
     }
 
@@ -4039,11 +4298,6 @@
     return mixerStatus;
 }
 
-void AudioFlinger::OffloadThread::flushOutput_l()
-{
-    mFlushPending = true;
-}
-
 // must be called with thread mutex locked
 bool AudioFlinger::OffloadThread::waitingAsyncCallback_l()
 {
@@ -4058,15 +4312,15 @@
 // must be called with thread mutex locked
 bool AudioFlinger::OffloadThread::shouldStandby_l()
 {
-    bool TrackPaused = false;
+    bool trackPaused = false;
 
     // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack
     // after a timeout and we will enter standby then.
     if (mTracks.size() > 0) {
-        TrackPaused = mTracks[mTracks.size() - 1]->isPaused();
+        trackPaused = mTracks[mTracks.size() - 1]->isPaused();
     }
 
-    return !mStandby && !TrackPaused;
+    return !mStandby && !trackPaused;
 }
 
 
@@ -4084,6 +4338,8 @@
     mBytesRemaining = 0;
     mPausedWriteLength = 0;
     mPausedBytesRemaining = 0;
+    mHwPaused = false;
+
     if (mUseAsyncWrite) {
         // discard any pending drain or write ack by incrementing sequence
         mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
@@ -4094,6 +4350,18 @@
     }
 }
 
+void AudioFlinger::OffloadThread::onAddNewTrack_l()
+{
+    sp<Track> previousTrack = mPreviousTrack.promote();
+    sp<Track> latestTrack = mLatestActiveTrack.promote();
+
+    if (previousTrack != 0 && latestTrack != 0 &&
+        (previousTrack->sessionId() != latestTrack->sessionId())) {
+        mFlushPending = true;
+    }
+    PlaybackThread::onAddNewTrack_l();
+}
+
 // ----------------------------------------------------------------------------
 
 AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
@@ -4118,11 +4386,11 @@
     if (outputsReady(outputTracks)) {
         mAudioMixer->process(AudioBufferProvider::kInvalidPTS);
     } else {
-        memset(mMixBuffer, 0, mixBufferSize);
+        memset(mSinkBuffer, 0, mSinkBufferSize);
     }
     sleepTime = 0;
     writeFrames = mNormalFrameCount;
-    mCurrentWriteLength = mixBufferSize;
+    mCurrentWriteLength = mSinkBufferSize;
     standbyTime = systemTime() + standbyDelay;
 }
 
@@ -4137,7 +4405,7 @@
     } else if (mBytesWritten != 0) {
         if (mMixerStatus == MIXER_TRACKS_ENABLED) {
             writeFrames = mNormalFrameCount;
-            memset(mMixBuffer, 0, mixBufferSize);
+            memset(mSinkBuffer, 0, mSinkBufferSize);
         } else {
             // flush remaining overflow buffers in output tracks
             writeFrames = 0;
@@ -4149,9 +4417,10 @@
 ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
 {
     for (size_t i = 0; i < outputTracks.size(); i++) {
-        outputTracks[i]->write(mMixBuffer, writeFrames);
+        outputTracks[i]->write(mSinkBuffer, writeFrames);
     }
-    return (ssize_t)mixBufferSize;
+    mStandby = false;
+    return (ssize_t)mSinkBufferSize;
 }
 
 void AudioFlinger::DuplicatingThread::threadLoop_standby()
@@ -4182,7 +4451,8 @@
                                             mSampleRate,
                                             mFormat,
                                             mChannelMask,
-                                            frameCount);
+                                            frameCount,
+                                            IPCThreadState::self()->getCallingUid());
     if (outputTrack->cblk() != NULL) {
         thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
         mOutputTracks.add(outputTrack);
@@ -4261,8 +4531,6 @@
 
 AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
                                          AudioStreamIn *input,
-                                         uint32_t sampleRate,
-                                         audio_channel_mask_t channelMask,
                                          audio_io_handle_t id,
                                          audio_devices_t outDevice,
                                          audio_devices_t inDevice
@@ -4271,28 +4539,24 @@
 #endif
                                          ) :
     ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD),
-    mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
-    // mRsmpInIndex set by readInputParameters()
-    mReqChannelCount(popcount(channelMask)),
-    mReqSampleRate(sampleRate)
-    // mBytesRead is only meaningful while active, and so is cleared in start()
-    // (but might be better to also clear here for dump?)
+    mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL),
+    // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l()
+    mRsmpInRear(0)
 #ifdef TEE_SINK
     , mTeeSink(teeSink)
 #endif
 {
     snprintf(mName, kNameLength, "AudioIn_%X", id);
+    mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName);
 
-    readInputParameters();
-
+    readInputParameters_l();
 }
 
 
 AudioFlinger::RecordThread::~RecordThread()
 {
+    mAudioFlinger->unregisterWriter(mNBLogWriter);
     delete[] mRsmpInBuffer;
-    delete mResampler;
-    delete[] mRsmpOutBuffer;
 }
 
 void AudioFlinger::RecordThread::onFirstRef()
@@ -4302,232 +4566,383 @@
 
 bool AudioFlinger::RecordThread::threadLoop()
 {
-    AudioBufferProvider::Buffer buffer;
-
     nsecs_t lastWarning = 0;
 
     inputStandBy();
-    acquireWakeLock();
 
-    // used to verify we've read at least once before evaluating how many bytes were read
-    bool readOnce = false;
+reacquire_wakelock:
+    sp<RecordTrack> activeTrack;
+    int activeTracksGen;
+    {
+        Mutex::Autolock _l(mLock);
+        size_t size = mActiveTracks.size();
+        activeTracksGen = mActiveTracksGen;
+        if (size > 0) {
+            // FIXME an arbitrary choice
+            activeTrack = mActiveTracks[0];
+            acquireWakeLock_l(activeTrack->uid());
+            if (size > 1) {
+                SortedVector<int> tmp;
+                for (size_t i = 0; i < size; i++) {
+                    tmp.add(mActiveTracks[i]->uid());
+                }
+                updateWakeLockUids_l(tmp);
+            }
+        } else {
+            acquireWakeLock_l(-1);
+        }
+    }
 
     // used to request a deferred sleep, to be executed later while mutex is unlocked
-    bool doSleep = false;
+    uint32_t sleepUs = 0;
 
-    // start recording
+    // loop while there is work to do
     for (;;) {
-        sp<RecordTrack> activeTrack;
-        TrackBase::track_state activeTrackState;
         Vector< sp<EffectChain> > effectChains;
 
         // sleep with mutex unlocked
-        if (doSleep) {
-            doSleep = false;
-            usleep(kRecordThreadSleepUs);
+        if (sleepUs > 0) {
+            usleep(sleepUs);
+            sleepUs = 0;
         }
 
+        // activeTracks accumulates a copy of a subset of mActiveTracks
+        Vector< sp<RecordTrack> > activeTracks;
+
         { // scope for mLock
             Mutex::Autolock _l(mLock);
-            if (exitPending()) {
-                break;
-            }
+
             processConfigEvents_l();
             // return value 'reconfig' is currently unused
             bool reconfig = checkForNewParameters_l();
-            // make a stable copy of mActiveTrack
-            activeTrack = mActiveTrack;
-            if (activeTrack == 0) {
-                standby();
+
+            // check exitPending here because checkForNewParameters_l() and
+            // checkForNewParameters_l() can temporarily release mLock
+            if (exitPending()) {
+                break;
+            }
+
+            // if no active track(s), then standby and release wakelock
+            size_t size = mActiveTracks.size();
+            if (size == 0) {
+                standbyIfNotAlreadyInStandby();
                 // exitPending() can't become true here
                 releaseWakeLock_l();
                 ALOGV("RecordThread: loop stopping");
                 // go to sleep
                 mWaitWorkCV.wait(mLock);
                 ALOGV("RecordThread: loop starting");
-                acquireWakeLock_l();
-                continue;
+                goto reacquire_wakelock;
             }
 
-            if (activeTrack->isTerminated()) {
-                removeTrack_l(activeTrack);
-                mActiveTrack.clear();
-                continue;
+            if (mActiveTracksGen != activeTracksGen) {
+                activeTracksGen = mActiveTracksGen;
+                SortedVector<int> tmp;
+                for (size_t i = 0; i < size; i++) {
+                    tmp.add(mActiveTracks[i]->uid());
+                }
+                updateWakeLockUids_l(tmp);
             }
 
-            activeTrackState = activeTrack->mState;
-            switch (activeTrackState) {
-            case TrackBase::PAUSING:
-                standby();
-                mActiveTrack.clear();
-                mStartStopCond.broadcast();
-                doSleep = true;
-                continue;
+            bool doBroadcast = false;
+            for (size_t i = 0; i < size; ) {
 
-            case TrackBase::RESUMING:
-                mStandby = false;
-                if (mReqChannelCount != activeTrack->channelCount()) {
-                    mActiveTrack.clear();
-                    mStartStopCond.broadcast();
+                activeTrack = mActiveTracks[i];
+                if (activeTrack->isTerminated()) {
+                    removeTrack_l(activeTrack);
+                    mActiveTracks.remove(activeTrack);
+                    mActiveTracksGen++;
+                    size--;
                     continue;
                 }
-                if (readOnce) {
-                    mStartStopCond.broadcast();
-                    // record start succeeds only if first read from audio input succeeds
-                    if (mBytesRead < 0) {
-                        mActiveTrack.clear();
-                        continue;
-                    }
+
+                TrackBase::track_state activeTrackState = activeTrack->mState;
+                switch (activeTrackState) {
+
+                case TrackBase::PAUSING:
+                    mActiveTracks.remove(activeTrack);
+                    mActiveTracksGen++;
+                    doBroadcast = true;
+                    size--;
+                    continue;
+
+                case TrackBase::STARTING_1:
+                    sleepUs = 10000;
+                    i++;
+                    continue;
+
+                case TrackBase::STARTING_2:
+                    doBroadcast = true;
+                    mStandby = false;
                     activeTrack->mState = TrackBase::ACTIVE;
+                    break;
+
+                case TrackBase::ACTIVE:
+                    break;
+
+                case TrackBase::IDLE:
+                    i++;
+                    continue;
+
+                default:
+                    LOG_FATAL("Unexpected activeTrackState %d", activeTrackState);
                 }
-                break;
 
-            case TrackBase::ACTIVE:
-                break;
+                activeTracks.add(activeTrack);
+                i++;
 
-            case TrackBase::IDLE:
-                doSleep = true;
-                continue;
-
-            default:
-                LOG_FATAL("Unexpected activeTrackState %d", activeTrackState);
             }
+            if (doBroadcast) {
+                mStartStopCond.broadcast();
+            }
+
+            // sleep if there are no active tracks to process
+            if (activeTracks.size() == 0) {
+                if (sleepUs == 0) {
+                    sleepUs = kRecordThreadSleepUs;
+                }
+                continue;
+            }
+            sleepUs = 0;
 
             lockEffectChains_l(effectChains);
         }
 
-        // thread mutex is now unlocked, mActiveTrack unknown, activeTrack != 0, kept, immutable
-        // activeTrack->mState unknown, activeTrackState immutable and is ACTIVE or RESUMING
+        // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0
 
-        for (size_t i = 0; i < effectChains.size(); i ++) {
+        size_t size = effectChains.size();
+        for (size_t i = 0; i < size; i++) {
             // thread mutex is not locked, but effect chain is locked
             effectChains[i]->process_l();
         }
 
-        buffer.frameCount = mFrameCount;
-        status_t status = activeTrack->getNextBuffer(&buffer);
-        if (status == NO_ERROR) {
-            readOnce = true;
-            size_t framesOut = buffer.frameCount;
-            if (mResampler == NULL) {
-                // no resampling
-                while (framesOut) {
-                    size_t framesIn = mFrameCount - mRsmpInIndex;
-                    if (framesIn > 0) {
-                        int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
-                        int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) *
-                                activeTrack->mFrameSize;
-                        if (framesIn > framesOut) {
-                            framesIn = framesOut;
-                        }
-                        mRsmpInIndex += framesIn;
-                        framesOut -= framesIn;
-                        if (mChannelCount == mReqChannelCount) {
-                            memcpy(dst, src, framesIn * mFrameSize);
-                        } else {
-                            if (mChannelCount == 1) {
-                                upmix_to_stereo_i16_from_mono_i16((int16_t *)dst,
-                                        (int16_t *)src, framesIn);
-                            } else {
-                                downmix_to_mono_i16_from_stereo_i16((int16_t *)dst,
-                                        (int16_t *)src, framesIn);
-                            }
-                        }
-                    }
-                    if (framesOut > 0 && mFrameCount == mRsmpInIndex) {
-                        void *readInto;
-                        if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) {
-                            readInto = buffer.raw;
-                            framesOut = 0;
-                        } else {
-                            readInto = mRsmpInBuffer;
-                            mRsmpInIndex = 0;
-                        }
-                        mBytesRead = mInput->stream->read(mInput->stream, readInto,
-                                mBufferSize);
-                        if (mBytesRead <= 0) {
-                            // TODO: verify that it's benign to use a stale track state
-                            if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE))
-                            {
-                                ALOGE("Error reading audio input");
-                                // Force input into standby so that it tries to
-                                // recover at next read attempt
-                                inputStandBy();
-                                doSleep = true;
-                            }
-                            mRsmpInIndex = mFrameCount;
-                            framesOut = 0;
-                            buffer.frameCount = 0;
-                        }
-#ifdef TEE_SINK
-                        else if (mTeeSink != 0) {
-                            (void) mTeeSink->write(readInto,
-                                    mBytesRead >> Format_frameBitShift(mTeeSink->format()));
-                        }
-#endif
-                    }
-                }
-            } else {
-                // resampling
+        // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one.
+        // Only the client(s) that are too slow will overrun. But if even the fastest client is too
+        // slow, then this RecordThread will overrun by not calling HAL read often enough.
+        // If destination is non-contiguous, first read past the nominal end of buffer, then
+        // copy to the right place.  Permitted because mRsmpInBuffer was over-allocated.
 
-                // resampler accumulates, but we only have one source track
-                memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t));
-                // alter output frame count as if we were expecting stereo samples
-                if (mChannelCount == 1 && mReqChannelCount == 1) {
-                    framesOut >>= 1;
-                }
-                mResampler->resample(mRsmpOutBuffer, framesOut,
-                        this /* AudioBufferProvider* */);
-                // ditherAndClamp() works as long as all buffers returned by
-                // activeTrack->getNextBuffer() are 32 bit aligned which should be always true.
-                if (mChannelCount == 2 && mReqChannelCount == 1) {
-                    // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t
-                    ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
-                    // the resampler always outputs stereo samples:
-                    // do post stereo to mono conversion
-                    downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer,
-                            framesOut);
-                } else {
-                    ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
-                }
-                // now done with mRsmpOutBuffer
-
-            }
-            if (mFramestoDrop == 0) {
-                activeTrack->releaseBuffer(&buffer);
-            } else {
-                if (mFramestoDrop > 0) {
-                    mFramestoDrop -= buffer.frameCount;
-                    if (mFramestoDrop <= 0) {
-                        clearSyncStartEvent();
-                    }
-                } else {
-                    mFramestoDrop += buffer.frameCount;
-                    if (mFramestoDrop >= 0 || mSyncStartEvent == 0 ||
-                            mSyncStartEvent->isCancelled()) {
-                        ALOGW("Synced record %s, session %d, trigger session %d",
-                              (mFramestoDrop >= 0) ? "timed out" : "cancelled",
-                              activeTrack->sessionId(),
-                              (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0);
-                        clearSyncStartEvent();
-                    }
-                }
-            }
-            activeTrack->clearOverflow();
+        int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1);
+        ssize_t bytesRead = mInput->stream->read(mInput->stream,
+                &mRsmpInBuffer[rear * mChannelCount], mBufferSize);
+        if (bytesRead <= 0) {
+            ALOGE("read failed: bytesRead=%d < %u", bytesRead, mBufferSize);
+            // Force input into standby so that it tries to recover at next read attempt
+            inputStandBy();
+            sleepUs = kRecordThreadSleepUs;
+            continue;
         }
-        // client isn't retrieving buffers fast enough
-        else {
-            if (!activeTrack->setOverflow()) {
-                nsecs_t now = systemTime();
-                if ((now - lastWarning) > kWarningThrottleNs) {
-                    ALOGW("RecordThread: buffer overflow");
-                    lastWarning = now;
+        ALOG_ASSERT((size_t) bytesRead <= mBufferSize);
+        size_t framesRead = bytesRead / mFrameSize;
+        ALOG_ASSERT(framesRead > 0);
+        if (mTeeSink != 0) {
+            (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead);
+        }
+        // If destination is non-contiguous, we now correct for reading past end of buffer.
+        size_t part1 = mRsmpInFramesP2 - rear;
+        if (framesRead > part1) {
+            memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount],
+                    (framesRead - part1) * mFrameSize);
+        }
+        rear = mRsmpInRear += framesRead;
+
+        size = activeTracks.size();
+        // loop over each active track
+        for (size_t i = 0; i < size; i++) {
+            activeTrack = activeTracks[i];
+
+            enum {
+                OVERRUN_UNKNOWN,
+                OVERRUN_TRUE,
+                OVERRUN_FALSE
+            } overrun = OVERRUN_UNKNOWN;
+
+            // loop over getNextBuffer to handle circular sink
+            for (;;) {
+
+                activeTrack->mSink.frameCount = ~0;
+                status_t status = activeTrack->getNextBuffer(&activeTrack->mSink);
+                size_t framesOut = activeTrack->mSink.frameCount;
+                LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0));
+
+                int32_t front = activeTrack->mRsmpInFront;
+                ssize_t filled = rear - front;
+                size_t framesIn;
+
+                if (filled < 0) {
+                    // should not happen, but treat like a massive overrun and re-sync
+                    framesIn = 0;
+                    activeTrack->mRsmpInFront = rear;
+                    overrun = OVERRUN_TRUE;
+                } else if ((size_t) filled <= mRsmpInFrames) {
+                    framesIn = (size_t) filled;
+                } else {
+                    // client is not keeping up with server, but give it latest data
+                    framesIn = mRsmpInFrames;
+                    activeTrack->mRsmpInFront = front = rear - framesIn;
+                    overrun = OVERRUN_TRUE;
+                }
+
+                if (framesOut == 0 || framesIn == 0) {
+                    break;
+                }
+
+                if (activeTrack->mResampler == NULL) {
+                    // no resampling
+                    if (framesIn > framesOut) {
+                        framesIn = framesOut;
+                    } else {
+                        framesOut = framesIn;
+                    }
+                    int8_t *dst = activeTrack->mSink.i8;
+                    while (framesIn > 0) {
+                        front &= mRsmpInFramesP2 - 1;
+                        size_t part1 = mRsmpInFramesP2 - front;
+                        if (part1 > framesIn) {
+                            part1 = framesIn;
+                        }
+                        int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize);
+                        if (mChannelCount == activeTrack->mChannelCount) {
+                            memcpy(dst, src, part1 * mFrameSize);
+                        } else if (mChannelCount == 1) {
+                            upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src,
+                                    part1);
+                        } else {
+                            downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (int16_t *)src,
+                                    part1);
+                        }
+                        dst += part1 * activeTrack->mFrameSize;
+                        front += part1;
+                        framesIn -= part1;
+                    }
+                    activeTrack->mRsmpInFront += framesOut;
+
+                } else {
+                    // resampling
+                    // FIXME framesInNeeded should really be part of resampler API, and should
+                    //       depend on the SRC ratio
+                    //       to keep mRsmpInBuffer full so resampler always has sufficient input
+                    size_t framesInNeeded;
+                    // FIXME only re-calculate when it changes, and optimize for common ratios
+                    double inOverOut = (double) mSampleRate / activeTrack->mSampleRate;
+                    double outOverIn = (double) activeTrack->mSampleRate / mSampleRate;
+                    framesInNeeded = ceil(framesOut * inOverOut) + 1;
+                    ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g",
+                                framesInNeeded, framesOut, inOverOut);
+                    // Although we theoretically have framesIn in circular buffer, some of those are
+                    // unreleased frames, and thus must be discounted for purpose of budgeting.
+                    size_t unreleased = activeTrack->mRsmpInUnrel;
+                    framesIn = framesIn > unreleased ? framesIn - unreleased : 0;
+                    if (framesIn < framesInNeeded) {
+                        ALOGV("not enough to resample: have %u frames in but need %u in to "
+                                "produce %u out given in/out ratio of %.4g",
+                                framesIn, framesInNeeded, framesOut, inOverOut);
+                        size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0;
+                        LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut);
+                        if (newFramesOut == 0) {
+                            break;
+                        }
+                        framesInNeeded = ceil(newFramesOut * inOverOut) + 1;
+                        ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g",
+                                framesInNeeded, newFramesOut, outOverIn);
+                        LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded);
+                        ALOGV("success 2: have %u frames in and need %u in to produce %u out "
+                              "given in/out ratio of %.4g",
+                              framesIn, framesInNeeded, newFramesOut, inOverOut);
+                        framesOut = newFramesOut;
+                    } else {
+                        ALOGV("success 1: have %u in and need %u in to produce %u out "
+                            "given in/out ratio of %.4g",
+                            framesIn, framesInNeeded, framesOut, inOverOut);
+                    }
+
+                    // reallocate mRsmpOutBuffer as needed; we will grow but never shrink
+                    if (activeTrack->mRsmpOutFrameCount < framesOut) {
+                        // FIXME why does each track need it's own mRsmpOutBuffer? can't they share?
+                        delete[] activeTrack->mRsmpOutBuffer;
+                        // resampler always outputs stereo
+                        activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2];
+                        activeTrack->mRsmpOutFrameCount = framesOut;
+                    }
+
+                    // resampler accumulates, but we only have one source track
+                    memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t));
+                    activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut,
+                            // FIXME how about having activeTrack implement this interface itself?
+                            activeTrack->mResamplerBufferProvider
+                            /*this*/ /* AudioBufferProvider* */);
+                    // ditherAndClamp() works as long as all buffers returned by
+                    // activeTrack->getNextBuffer() are 32 bit aligned which should be always true.
+                    if (activeTrack->mChannelCount == 1) {
+                        // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t
+                        ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer,
+                                framesOut);
+                        // the resampler always outputs stereo samples:
+                        // do post stereo to mono conversion
+                        downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16,
+                                (int16_t *)activeTrack->mRsmpOutBuffer, framesOut);
+                    } else {
+                        ditherAndClamp((int32_t *)activeTrack->mSink.raw,
+                                activeTrack->mRsmpOutBuffer, framesOut);
+                    }
+                    // now done with mRsmpOutBuffer
+
+                }
+
+                if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) {
+                    overrun = OVERRUN_FALSE;
+                }
+
+                if (activeTrack->mFramesToDrop == 0) {
+                    if (framesOut > 0) {
+                        activeTrack->mSink.frameCount = framesOut;
+                        activeTrack->releaseBuffer(&activeTrack->mSink);
+                    }
+                } else {
+                    // FIXME could do a partial drop of framesOut
+                    if (activeTrack->mFramesToDrop > 0) {
+                        activeTrack->mFramesToDrop -= framesOut;
+                        if (activeTrack->mFramesToDrop <= 0) {
+                            activeTrack->clearSyncStartEvent();
+                        }
+                    } else {
+                        activeTrack->mFramesToDrop += framesOut;
+                        if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 ||
+                                activeTrack->mSyncStartEvent->isCancelled()) {
+                            ALOGW("Synced record %s, session %d, trigger session %d",
+                                  (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled",
+                                  activeTrack->sessionId(),
+                                  (activeTrack->mSyncStartEvent != 0) ?
+                                          activeTrack->mSyncStartEvent->triggerSession() : 0);
+                            activeTrack->clearSyncStartEvent();
+                        }
+                    }
+                }
+
+                if (framesOut == 0) {
+                    break;
                 }
             }
-            // Release the processor for a while before asking for a new buffer.
-            // This will give the application more chance to read from the buffer and
-            // clear the overflow.
-            doSleep = true;
+
+            switch (overrun) {
+            case OVERRUN_TRUE:
+                // client isn't retrieving buffers fast enough
+                if (!activeTrack->setOverflow()) {
+                    nsecs_t now = systemTime();
+                    // FIXME should lastWarning per track?
+                    if ((now - lastWarning) > kWarningThrottleNs) {
+                        ALOGW("RecordThread: buffer overflow");
+                        lastWarning = now;
+                    }
+                }
+                break;
+            case OVERRUN_FALSE:
+                activeTrack->clearOverflow();
+                break;
+            case OVERRUN_UNKNOWN:
+                break;
+            }
+
         }
 
         // enable changes in effect chain
@@ -4535,7 +4950,7 @@
         // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end
     }
 
-    standby();
+    standbyIfNotAlreadyInStandby();
 
     {
         Mutex::Autolock _l(mLock);
@@ -4543,7 +4958,8 @@
             sp<RecordTrack> track = mTracks[i];
             track->invalidate();
         }
-        mActiveTrack.clear();
+        mActiveTracks.clear();
+        mActiveTracksGen++;
         mStartStopCond.broadcast();
     }
 
@@ -4553,7 +4969,7 @@
     return false;
 }
 
-void AudioFlinger::RecordThread::standby()
+void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby()
 {
     if (!mStandby) {
         inputStandBy();
@@ -4571,18 +4987,20 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        size_t frameCount,
+        size_t *pFrameCount,
         int sessionId,
+        int uid,
         IAudioFlinger::track_flags_t *flags,
         pid_t tid,
         status_t *status)
 {
+    size_t frameCount = *pFrameCount;
     sp<RecordTrack> track;
     status_t lStatus;
 
     lStatus = initCheck();
     if (lStatus != NO_ERROR) {
-        ALOGE("Audio driver not initialized.");
+        ALOGE("createRecordTrack_l() audio driver not initialized");
         goto Exit;
     }
 
@@ -4593,7 +5011,7 @@
             (
                 (tid != -1) &&
                 ((frameCount == 0) ||
-                (frameCount >= (mFrameCount * kFastTrackMultiplier)))
+                (frameCount >= mFrameCount))
             ) &&
             // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format)
             // mono or stereo
@@ -4636,6 +5054,7 @@
         }
       }
     }
+    *pFrameCount = frameCount;
 
     // FIXME use flags and tid similar to createTrack_l()
 
@@ -4643,11 +5062,12 @@
         Mutex::Autolock _l(mLock);
 
         track = new RecordTrack(this, client, sampleRate,
-                      format, channelMask, frameCount, sessionId);
+                      format, channelMask, frameCount, sessionId, uid);
 
         lStatus = track->initCheck();
         if (lStatus != NO_ERROR) {
-            track.clear();
+            ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus);
+            // track must be cleared from the caller as the caller has the AF lock
             goto Exit;
         }
         mTracks.add(track);
@@ -4681,113 +5101,99 @@
     status_t status = NO_ERROR;
 
     if (event == AudioSystem::SYNC_EVENT_NONE) {
-        clearSyncStartEvent();
+        recordTrack->clearSyncStartEvent();
     } else if (event != AudioSystem::SYNC_EVENT_SAME) {
-        mSyncStartEvent = mAudioFlinger->createSyncEvent(event,
+        recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event,
                                        triggerSession,
                                        recordTrack->sessionId(),
                                        syncStartEventCallback,
-                                       this);
+                                       recordTrack);
         // Sync event can be cancelled by the trigger session if the track is not in a
         // compatible state in which case we start record immediately
-        if (mSyncStartEvent->isCancelled()) {
-            clearSyncStartEvent();
+        if (recordTrack->mSyncStartEvent->isCancelled()) {
+            recordTrack->clearSyncStartEvent();
         } else {
             // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs
-            mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000);
+            recordTrack->mFramesToDrop = -
+                    ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000);
         }
     }
 
     {
         // This section is a rendezvous between binder thread executing start() and RecordThread
         AutoMutex lock(mLock);
-        if (mActiveTrack != 0) {
-            if (recordTrack != mActiveTrack.get()) {
-                status = -EBUSY;
-            } else if (mActiveTrack->mState == TrackBase::PAUSING) {
-                mActiveTrack->mState = TrackBase::ACTIVE;
+        if (mActiveTracks.indexOf(recordTrack) >= 0) {
+            if (recordTrack->mState == TrackBase::PAUSING) {
+                ALOGV("active record track PAUSING -> ACTIVE");
+                recordTrack->mState = TrackBase::ACTIVE;
+            } else {
+                ALOGV("active record track state %d", recordTrack->mState);
             }
             return status;
         }
 
-        // FIXME why? already set in constructor, 'STARTING_1' would be more accurate
-        recordTrack->mState = TrackBase::IDLE;
-        mActiveTrack = recordTrack;
+        // TODO consider other ways of handling this, such as changing the state to :STARTING and
+        //      adding the track to mActiveTracks after returning from AudioSystem::startInput(),
+        //      or using a separate command thread
+        recordTrack->mState = TrackBase::STARTING_1;
+        mActiveTracks.add(recordTrack);
+        mActiveTracksGen++;
         mLock.unlock();
         status_t status = AudioSystem::startInput(mId);
         mLock.lock();
-        // FIXME should verify that mActiveTrack is still == recordTrack
+        // FIXME should verify that recordTrack is still in mActiveTracks
         if (status != NO_ERROR) {
-            mActiveTrack.clear();
-            clearSyncStartEvent();
+            mActiveTracks.remove(recordTrack);
+            mActiveTracksGen++;
+            recordTrack->clearSyncStartEvent();
             return status;
         }
-        mRsmpInIndex = mFrameCount;
-        mBytesRead = 0;
-        if (mResampler != NULL) {
-            mResampler->reset();
+        // Catch up with current buffer indices if thread is already running.
+        // This is what makes a new client discard all buffered data.  If the track's mRsmpInFront
+        // was initialized to some value closer to the thread's mRsmpInFront, then the track could
+        // see previously buffered data before it called start(), but with greater risk of overrun.
+
+        recordTrack->mRsmpInFront = mRsmpInRear;
+        recordTrack->mRsmpInUnrel = 0;
+        // FIXME why reset?
+        if (recordTrack->mResampler != NULL) {
+            recordTrack->mResampler->reset();
         }
-        // FIXME hijacking a playback track state name which was intended for start after pause;
-        //       here 'STARTING_2' would be more accurate
-        mActiveTrack->mState = TrackBase::RESUMING;
+        recordTrack->mState = TrackBase::STARTING_2;
         // signal thread to start
-        ALOGV("Signal record thread");
         mWaitWorkCV.broadcast();
-        // do not wait for mStartStopCond if exiting
-        if (exitPending()) {
-            mActiveTrack.clear();
-            status = INVALID_OPERATION;
-            goto startError;
-        }
-        // FIXME incorrect usage of wait: no explicit predicate or loop
-        mStartStopCond.wait(mLock);
-        if (mActiveTrack == 0) {
+        if (mActiveTracks.indexOf(recordTrack) < 0) {
             ALOGV("Record failed to start");
             status = BAD_VALUE;
             goto startError;
         }
-        ALOGV("Record started OK");
         return status;
     }
 
 startError:
     AudioSystem::stopInput(mId);
-    clearSyncStartEvent();
+    recordTrack->clearSyncStartEvent();
+    // FIXME I wonder why we do not reset the state here?
     return status;
 }
 
-void AudioFlinger::RecordThread::clearSyncStartEvent()
-{
-    if (mSyncStartEvent != 0) {
-        mSyncStartEvent->cancel();
-    }
-    mSyncStartEvent.clear();
-    mFramestoDrop = 0;
-}
-
 void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event)
 {
     sp<SyncEvent> strongEvent = event.promote();
 
     if (strongEvent != 0) {
-        RecordThread *me = (RecordThread *)strongEvent->cookie();
-        me->handleSyncStartEvent(strongEvent);
-    }
-}
-
-void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event)
-{
-    if (event == mSyncStartEvent) {
-        // TODO: use actual buffer filling status instead of 2 buffers when info is available
-        // from audio HAL
-        mFramestoDrop = mFrameCount * 2;
+        sp<RefBase> ptr = strongEvent->cookie().promote();
+        if (ptr != 0) {
+            RecordTrack *recordTrack = (RecordTrack *)ptr.get();
+            recordTrack->handleSyncStartEvent(strongEvent);
+        }
     }
 }
 
 bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
     ALOGV("RecordThread::stop");
     AutoMutex _l(mLock);
-    if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) {
+    if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) {
         return false;
     }
     // note that threadLoop may still be processing the track at this point [without lock]
@@ -4798,20 +5204,20 @@
     }
     // FIXME incorrect usage of wait: no explicit predicate or loop
     mStartStopCond.wait(mLock);
-    // if we have been restarted, recordTrack == mActiveTrack.get() here
-    if (exitPending() || recordTrack != mActiveTrack.get()) {
+    // if we have been restarted, recordTrack is in mActiveTracks here
+    if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) {
         ALOGV("Record stopped OK");
         return true;
     }
     return false;
 }
 
-bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const
+bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const
 {
     return false;
 }
 
-status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event)
+status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused)
 {
 #if 0   // This branch is currently dead code, but is preserved in case it will be needed in future
     if (!isValidSyncEvent(event)) {
@@ -4842,7 +5248,7 @@
     track->terminate();
     track->mState = TrackBase::STOPPED;
     // active tracks are removed by threadLoop()
-    if (mActiveTrack != track) {
+    if (mActiveTracks.indexOf(track) < 0) {
         removeTrack_l(track);
     }
 }
@@ -4862,105 +5268,119 @@
 
 void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args)
 {
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
+    fdprintf(fd, "\nInput thread %p:\n", this);
 
-    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
-    result.append(buffer);
-
-    if (mActiveTrack != 0) {
-        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
-        result.append(buffer);
-        snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize);
-        result.append(buffer);
-        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
-        result.append(buffer);
-        snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount);
-        result.append(buffer);
-        snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate);
-        result.append(buffer);
+    if (mActiveTracks.size() > 0) {
+        fdprintf(fd, "  Buffer size: %zu bytes\n", mBufferSize);
     } else {
-        result.append("No active record client\n");
+        fdprintf(fd, "  No active record clients\n");
     }
 
-    write(fd, result.string(), result.size());
-
     dumpBase(fd, args);
 }
 
-void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args)
+void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused)
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
     String8 result;
 
-    snprintf(buffer, SIZE, "Input thread %p tracks\n", this);
-    result.append(buffer);
-    RecordTrack::appendDumpHeader(result);
-    for (size_t i = 0; i < mTracks.size(); ++i) {
-        sp<RecordTrack> track = mTracks[i];
-        if (track != 0) {
-            track->dump(buffer, SIZE);
-            result.append(buffer);
+    size_t numtracks = mTracks.size();
+    size_t numactive = mActiveTracks.size();
+    size_t numactiveseen = 0;
+    fdprintf(fd, "  %d Tracks", numtracks);
+    if (numtracks) {
+        fdprintf(fd, " of which %d are active\n", numactive);
+        RecordTrack::appendDumpHeader(result);
+        for (size_t i = 0; i < numtracks ; ++i) {
+            sp<RecordTrack> track = mTracks[i];
+            if (track != 0) {
+                bool active = mActiveTracks.indexOf(track) >= 0;
+                if (active) {
+                    numactiveseen++;
+                }
+                track->dump(buffer, SIZE, active);
+                result.append(buffer);
+            }
         }
+    } else {
+        fdprintf(fd, "\n");
     }
 
-    if (mActiveTrack != 0) {
-        snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this);
+    if (numactiveseen != numactive) {
+        snprintf(buffer, SIZE, "  The following tracks are in the active list but"
+                " not in the track list\n");
         result.append(buffer);
         RecordTrack::appendDumpHeader(result);
-        mActiveTrack->dump(buffer, SIZE);
-        result.append(buffer);
+        for (size_t i = 0; i < numactive; ++i) {
+            sp<RecordTrack> track = mActiveTracks[i];
+            if (mTracks.indexOf(track) < 0) {
+                track->dump(buffer, SIZE, true);
+                result.append(buffer);
+            }
+        }
 
     }
     write(fd, result.string(), result.size());
 }
 
 // AudioBufferProvider interface
-status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
+status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
+        AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
 {
-    size_t framesReq = buffer->frameCount;
-    size_t framesReady = mFrameCount - mRsmpInIndex;
-    int channelCount;
-
-    if (framesReady == 0) {
-        mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize);
-        if (mBytesRead <= 0) {
-            if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) {
-                ALOGE("RecordThread::getNextBuffer() Error reading audio input");
-                // Force input into standby so that it tries to
-                // recover at next read attempt
-                inputStandBy();
-                // FIXME an awkward place to sleep, consider using doSleep when this is pulled up
-                usleep(kRecordThreadSleepUs);
-            }
-            buffer->raw = NULL;
-            buffer->frameCount = 0;
-            return NOT_ENOUGH_DATA;
-        }
-        mRsmpInIndex = 0;
-        framesReady = mFrameCount;
+    RecordTrack *activeTrack = mRecordTrack;
+    sp<ThreadBase> threadBase = activeTrack->mThread.promote();
+    if (threadBase == 0) {
+        buffer->frameCount = 0;
+        buffer->raw = NULL;
+        return NOT_ENOUGH_DATA;
+    }
+    RecordThread *recordThread = (RecordThread *) threadBase.get();
+    int32_t rear = recordThread->mRsmpInRear;
+    int32_t front = activeTrack->mRsmpInFront;
+    ssize_t filled = rear - front;
+    // FIXME should not be P2 (don't want to increase latency)
+    // FIXME if client not keeping up, discard
+    LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames));
+    // 'filled' may be non-contiguous, so return only the first contiguous chunk
+    front &= recordThread->mRsmpInFramesP2 - 1;
+    size_t part1 = recordThread->mRsmpInFramesP2 - front;
+    if (part1 > (size_t) filled) {
+        part1 = filled;
+    }
+    size_t ask = buffer->frameCount;
+    ALOG_ASSERT(ask > 0);
+    if (part1 > ask) {
+        part1 = ask;
+    }
+    if (part1 == 0) {
+        // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty
+        LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved");
+        buffer->raw = NULL;
+        buffer->frameCount = 0;
+        activeTrack->mRsmpInUnrel = 0;
+        return NOT_ENOUGH_DATA;
     }
 
-    if (framesReq > framesReady) {
-        framesReq = framesReady;
-    }
-
-    if (mChannelCount == 1 && mReqChannelCount == 2) {
-        channelCount = 1;
-    } else {
-        channelCount = 2;
-    }
-    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
-    buffer->frameCount = framesReq;
+    buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount;
+    buffer->frameCount = part1;
+    activeTrack->mRsmpInUnrel = part1;
     return NO_ERROR;
 }
 
 // AudioBufferProvider interface
-void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer(
+        AudioBufferProvider::Buffer* buffer)
 {
-    mRsmpInIndex += buffer->frameCount;
+    RecordTrack *activeTrack = mRecordTrack;
+    size_t stepCount = buffer->frameCount;
+    if (stepCount == 0) {
+        return;
+    }
+    ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel);
+    activeTrack->mRsmpInUnrel -= stepCount;
+    activeTrack->mRsmpInFront += stepCount;
+    buffer->raw = NULL;
     buffer->frameCount = 0;
 }
 
@@ -4974,11 +5394,14 @@
         AudioParameter param = AudioParameter(keyValuePair);
         int value;
         audio_format_t reqFormat = mFormat;
-        uint32_t reqSamplingRate = mReqSampleRate;
-        audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount);
+        uint32_t samplingRate = mSampleRate;
+        audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount);
 
+        // TODO Investigate when this code runs. Check with audio policy when a sample rate and
+        //      channel count change can be requested. Do we mandate the first client defines the
+        //      HAL sampling rate and channel count or do we allow changes on the fly?
         if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
-            reqSamplingRate = value;
+            samplingRate = value;
             reconfig = true;
         }
         if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
@@ -4994,7 +5417,7 @@
             if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) {
                 status = BAD_VALUE;
             } else {
-                reqChannelMask = mask;
+                channelMask = mask;
                 reconfig = true;
             }
         }
@@ -5002,7 +5425,7 @@
             // do not accept frame count changes if tracks are open as the track buffer
             // size depends on frame count and correct behavior would not be guaranteed
             // if frame count is changed after track creation
-            if (mActiveTrack != 0) {
+            if (mActiveTracks.size() > 0) {
                 status = INVALID_OPERATION;
             } else {
                 reconfig = true;
@@ -5059,15 +5482,15 @@
                     reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
                     reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
                     (mInput->stream->common.get_sample_rate(&mInput->stream->common)
-                            <= (2 * reqSamplingRate)) &&
+                            <= (2 * samplingRate)) &&
                     popcount(mInput->stream->common.get_channels(&mInput->stream->common))
                             <= FCC_2 &&
-                    (reqChannelMask == AUDIO_CHANNEL_IN_MONO ||
-                            reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) {
+                    (channelMask == AUDIO_CHANNEL_IN_MONO ||
+                            channelMask == AUDIO_CHANNEL_IN_STEREO)) {
                     status = NO_ERROR;
                 }
                 if (status == NO_ERROR) {
-                    readInputParameters();
+                    readInputParameters_l();
                     sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
                 }
             }
@@ -5097,9 +5520,9 @@
     return out_s8;
 }
 
-void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
+void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param __unused) {
     AudioSystem::OutputDescriptor desc;
-    void *param2 = NULL;
+    const void *param2 = NULL;
 
     switch (event) {
     case AudioSystem::INPUT_OPENED:
@@ -5119,52 +5542,35 @@
     mAudioFlinger->audioConfigChanged_l(event, mId, param2);
 }
 
-void AudioFlinger::RecordThread::readInputParameters()
+void AudioFlinger::RecordThread::readInputParameters_l()
 {
-    delete[] mRsmpInBuffer;
-    // mRsmpInBuffer is always assigned a new[] below
-    delete[] mRsmpOutBuffer;
-    mRsmpOutBuffer = NULL;
-    delete mResampler;
-    mResampler = NULL;
-
     mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
     mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
     mChannelCount = popcount(mChannelMask);
     mFormat = mInput->stream->common.get_format(&mInput->stream->common);
     if (mFormat != AUDIO_FORMAT_PCM_16_BIT) {
-        ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat);
+        ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat);
     }
     mFrameSize = audio_stream_frame_size(&mInput->stream->common);
     mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common);
     mFrameCount = mBufferSize / mFrameSize;
-    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
+    // This is the formula for calculating the temporary buffer size.
+    // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to
+    // 1 full output buffer, regardless of the alignment of the available input.
+    // The value is somewhat arbitrary, and could probably be even larger.
+    // A larger value should allow more old data to be read after a track calls start(),
+    // without increasing latency.
+    mRsmpInFrames = mFrameCount * 7;
+    mRsmpInFramesP2 = roundup(mRsmpInFrames);
+    delete[] mRsmpInBuffer;
+    // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer
+    mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount];
 
-    if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) {
-        int channelCount;
-        // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid
-        // stereo to mono post process as the resampler always outputs stereo.
-        if (mChannelCount == 1 && mReqChannelCount == 2) {
-            channelCount = 1;
-        } else {
-            channelCount = 2;
-        }
-        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
-        mResampler->setSampleRate(mSampleRate);
-        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
-        mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2];
-
-        // optmization: if mono to mono, alter input frame count as if we were inputing
-        // stereo samples
-        if (mChannelCount == 1 && mReqChannelCount == 1) {
-            mFrameCount >>= 1;
-        }
-
-    }
-    mRsmpInIndex = mFrameCount;
+    // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints.
+    // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks?
 }
 
-unsigned int AudioFlinger::RecordThread::getInputFramesLost()
+uint32_t AudioFlinger::RecordThread::getInputFramesLost()
 {
     Mutex::Autolock _l(mLock);
     if (initCheck() != NO_ERROR) {

services/audioflinger/Threads.h

diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index b96e1c8..96642ff 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -65,7 +65,7 @@
     class IoConfigEvent : public ConfigEvent {
     public:
         IoConfigEvent(int event, int param) :
-            ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(event) {}
+            ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(param) {}
         virtual ~IoConfigEvent() {}
 
                 int event() const { return mEvent; }
@@ -201,13 +201,13 @@
                 // effect
                 void removeEffect_l(const sp< EffectModule>& effect);
                 // detach all tracks connected to an auxiliary effect
-    virtual     void detachAuxEffect_l(int effectId) {}
+    virtual     void detachAuxEffect_l(int effectId __unused) {}
                 // returns either EFFECT_SESSION if effects on this audio session exist in one
                 // chain, or TRACK_SESSION if tracks on this audio session exist, or both
                 virtual uint32_t hasAudioSession(int sessionId) const = 0;
                 // the value returned by default implementation is not important as the
                 // strategy is only meaningful for PlaybackThread which implements this method
-                virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; }
+                virtual uint32_t getStrategyForSession_l(int sessionId __unused) { return 0; }
 
                 // suspend or restore effect according to the type of effect passed. a NULL
                 // type pointer means suspend all effects in the session
@@ -240,10 +240,13 @@
                     effect_uuid_t mType;    // effect type UUID
                 };
 
-                void        acquireWakeLock();
-                void        acquireWakeLock_l();
+                void        acquireWakeLock(int uid = -1);
+                void        acquireWakeLock_l(int uid = -1);
                 void        releaseWakeLock();
                 void        releaseWakeLock_l();
+                void        updateWakeLockUids(const SortedVector<int> &uids);
+                void        updateWakeLockUids_l(const SortedVector<int> &uids);
+                void        getPowerManager_l();
                 void setEffectSuspended_l(const effect_uuid_t *type,
                                           bool suspend,
                                           int sessionId);
@@ -254,6 +257,8 @@
                 // check if some effects must be suspended when an effect chain is added
                 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain);
 
+                String16 getWakeLockTag();
+
     virtual     void        preExit() { }
 
     friend class AudioFlinger;      // for mEffectChains
@@ -265,8 +270,8 @@
 
                 const sp<AudioFlinger>  mAudioFlinger;
 
-                // updated by PlaybackThread::readOutputParameters() or
-                // RecordThread::readInputParameters()
+                // updated by PlaybackThread::readOutputParameters_l() or
+                // RecordThread::readInputParameters_l()
                 uint32_t                mSampleRate;
                 size_t                  mFrameCount;       // output HAL, direct output, record
                 audio_channel_mask_t    mChannelMask;
@@ -302,12 +307,12 @@
                 Vector<ConfigEvent *>     mConfigEvents;
 
                 // These fields are written and read by thread itself without lock or barrier,
-                // and read by other threads without lock or barrier via standby() , outDevice()
+                // and read by other threads without lock or barrier via standby(), outDevice()
                 // and inDevice().
                 // Because of the absence of a lock or barrier, any other thread that reads
                 // these fields must use the information in isolation, or be prepared to deal
                 // with possibility that it might be inconsistent with other information.
-                bool                    mStandby;   // Whether thread is currently in standby.
+                bool                    mStandby;     // Whether thread is currently in standby.
                 audio_devices_t         mOutDevice;   // output device
                 audio_devices_t         mInDevice;    // input device
                 audio_source_t          mAudioSource; // (see audio.h, audio_source_t)
@@ -389,7 +394,7 @@
     virtual     bool        waitingAsyncCallback();
     virtual     bool        waitingAsyncCallback_l();
     virtual     bool        shouldStandby_l();
-
+    virtual     void        onAddNewTrack_l();
 
     // ThreadBase virtuals
     virtual     void        preExit();
@@ -417,11 +422,12 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                size_t frameCount,
+                                size_t *pFrameCount,
                                 const sp<IMemory>& sharedBuffer,
                                 int sessionId,
                                 IAudioFlinger::track_flags_t *flags,
                                 pid_t tid,
+                                int uid,
                                 status_t *status /*non-NULL*/);
 
                 AudioStreamOut* getOutput() const;
@@ -443,8 +449,9 @@
 
     virtual     String8     getParameters(const String8& keys);
     virtual     void        audioConfigChanged_l(int event, int param = 0);
-                status_t    getRenderPosition(size_t *halFrames, size_t *dspFrames);
-                int16_t     *mixBuffer() const { return mMixBuffer; };
+                status_t    getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
+                // TODO: rename mixBuffer() to sinkBuffer() or try to remove external use.
+                int16_t     *mixBuffer() const { return mSinkBuffer; };
 
     virtual     void detachAuxEffect_l(int effectId);
                 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track,
@@ -472,10 +479,33 @@
                 status_t         getTimestamp_l(AudioTimestamp& timestamp);
 
 protected:
-    // updated by readOutputParameters()
+    // updated by readOutputParameters_l()
     size_t                          mNormalFrameCount;  // normal mixer and effects
 
-    int16_t*                        mMixBuffer;         // frame size aligned mix buffer
+    int16_t*                        mSinkBuffer;         // frame size aligned sink buffer
+
+    // Mixer Buffer (mMixerBuffer*)
+    //
+    // In the case of floating point or multichannel data, which is not in the
+    // sink format, it is required to accumulate in a higher precision or greater channel count
+    // buffer before downmixing or data conversion to the sink buffer.
+
+    // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer.
+    bool                            mMixerBufferEnabled;
+
+    // Storage, 32 byte aligned (may make this alignment a requirement later).
+    // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames.
+    void*                           mMixerBuffer;
+
+    // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize.
+    size_t                          mMixerBufferSize;
+
+    // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only.
+    audio_format_t                  mMixerBufferFormat;
+
+    // An internal flag set to true by MixerThread::prepareTracks_l()
+    // when mMixerBuffer contains valid data after mixing.
+    bool                            mMixerBufferValid;
 
     // suspend count, > 0 means suspended.  While suspended, the thread continues to pull from
     // tracks and mix, but doesn't write to HAL.  A2DP and SCO HAL implementations can't handle
@@ -495,6 +525,9 @@
                 void        setMasterMute_l(bool muted) { mMasterMute = muted; }
 protected:
     SortedVector< wp<Track> >       mActiveTracks;  // FIXME check if this could be sp<>
+    SortedVector<int>               mWakeLockUids;
+    int                             mActiveTracksGeneration;
+    wp<Track>                       mLatestActiveTrack; // latest track added to mActiveTracks
 
     // Allocate a track name for a given channel mask.
     //   Returns name >= 0 if successful, -1 on failure.
@@ -532,7 +565,7 @@
     void        removeTrack_l(const sp<Track>& track);
     void        broadcast_l();
 
-    void        readOutputParameters();
+    void        readOutputParameters_l();
 
     virtual void dumpInternals(int fd, const Vector<String16>& args);
     void        dumpTracks(int fd, const Vector<String16>& args);
@@ -551,7 +584,7 @@
 
     // FIXME rename these former local variables of threadLoop to standard "m" names
     nsecs_t                         standbyTime;
-    size_t                          mixBufferSize;
+    size_t                          mSinkBufferSize;
 
     // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l()
     uint32_t                        activeSleepTime;
@@ -616,13 +649,12 @@
     sp<NBLog::Writer>       mFastMixerNBLogWriter;
 public:
     virtual     bool        hasFastMixer() const = 0;
-    virtual     FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const
+    virtual     FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const
                                 { FastTrackUnderruns dummy; return dummy; }
 
 protected:
                 // accessed by both binder threads and within threadLoop(), lock on mutex needed
                 unsigned    mFastTrackAvailMask;    // bit i set if fast track [i] is available
-    virtual     void        flushOutput_l();
 
 private:
     // timestamp latch:
@@ -735,17 +767,17 @@
 
     OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
                         audio_io_handle_t id, uint32_t device);
-    virtual                 ~OffloadThread();
+    virtual                 ~OffloadThread() {};
 
 protected:
     // threadLoop snippets
     virtual     mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
     virtual     void        threadLoop_exit();
-    virtual     void        flushOutput_l();
 
     virtual     bool        waitingAsyncCallback();
     virtual     bool        waitingAsyncCallback_l();
     virtual     bool        shouldStandby_l();
+    virtual     void        onAddNewTrack_l();
 
 private:
                 void        flushHw_l();
@@ -755,13 +787,13 @@
     bool        mFlushPending;
     size_t      mPausedWriteLength;     // length in bytes of write interrupted by pause
     size_t      mPausedBytesRemaining;  // bytes still waiting in mixbuffer after resume
-    sp<Track>   mPreviousTrack;         // used to detect track switch
+    wp<Track>   mPreviousTrack;         // used to detect track switch
 };
 
 class AsyncCallbackThread : public Thread {
 public:
 
-    AsyncCallbackThread(const sp<OffloadThread>& offloadThread);
+    AsyncCallbackThread(const wp<PlaybackThread>& playbackThread);
 
     virtual             ~AsyncCallbackThread();
 
@@ -778,17 +810,17 @@
             void        resetDraining();
 
 private:
-    wp<OffloadThread>   mOffloadThread;
+    const wp<PlaybackThread>   mPlaybackThread;
     // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via
     // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used
     // to indicate that the callback has been received via resetWriteBlocked()
-    uint32_t            mWriteAckSequence;
+    uint32_t                   mWriteAckSequence;
     // mDrainSequence corresponds to the last drain sequence passed by the offload thread via
     // setDraining(). The sequence is shifted one bit to the left and the lsb is used
     // to indicate that the callback has been received via resetDraining()
-    uint32_t            mDrainSequence;
-    Condition           mWaitWorkCV;
-    Mutex               mLock;
+    uint32_t                   mDrainSequence;
+    Condition                  mWaitWorkCV;
+    Mutex                      mLock;
 };
 
 class DuplicatingThread : public MixerThread {
@@ -831,17 +863,28 @@
 
 
 // record thread
-class RecordThread : public ThreadBase, public AudioBufferProvider
-                        // derives from AudioBufferProvider interface for use by resampler
+class RecordThread : public ThreadBase
 {
 public:
 
+    class RecordTrack;
+    class ResamplerBufferProvider : public AudioBufferProvider
+                        // derives from AudioBufferProvider interface for use by resampler
+    {
+    public:
+        ResamplerBufferProvider(RecordTrack* recordTrack) : mRecordTrack(recordTrack) { }
+        virtual ~ResamplerBufferProvider() { }
+        // AudioBufferProvider interface
+        virtual status_t    getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
+        virtual void        releaseBuffer(AudioBufferProvider::Buffer* buffer);
+    private:
+        RecordTrack * const mRecordTrack;
+    };
+
 #include "RecordTracks.h"
 
             RecordThread(const sp<AudioFlinger>& audioFlinger,
                     AudioStreamIn *input,
-                    uint32_t sampleRate,
-                    audio_channel_mask_t channelMask,
                     audio_io_handle_t id,
                     audio_devices_t outDevice,
                     audio_devices_t inDevice
@@ -871,8 +914,9 @@
                     uint32_t sampleRate,
                     audio_format_t format,
                     audio_channel_mask_t channelMask,
-                    size_t frameCount,
+                    size_t *pFrameCount,
                     int sessionId,
+                    int uid,
                     IAudioFlinger::track_flags_t *flags,
                     pid_t tid,
                     status_t *status /*non-NULL*/);
@@ -889,15 +933,12 @@
             AudioStreamIn* clearInput();
             virtual audio_stream_t* stream() const;
 
-    // AudioBufferProvider interface
-    virtual status_t    getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
-    virtual void        releaseBuffer(AudioBufferProvider::Buffer* buffer);
 
     virtual bool        checkForNewParameters_l();
     virtual String8     getParameters(const String8& keys);
     virtual void        audioConfigChanged_l(int event, int param = 0);
-            void        readInputParameters();
-    virtual unsigned int  getInputFramesLost();
+            void        readInputParameters_l();
+    virtual uint32_t    getInputFramesLost();
 
     virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
     virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
@@ -912,43 +953,33 @@
     virtual bool     isValidSyncEvent(const sp<SyncEvent>& event) const;
 
     static void syncStartEventCallback(const wp<SyncEvent>& event);
-           void handleSyncStartEvent(const sp<SyncEvent>& event);
 
     virtual size_t      frameCount() const { return mFrameCount; }
             bool        hasFastRecorder() const { return false; }
 
 private:
-            void    clearSyncStartEvent();
-
             // Enter standby if not already in standby, and set mStandby flag
-            void    standby();
+            void    standbyIfNotAlreadyInStandby();
 
             // Call the HAL standby method unconditionally, and don't change mStandby flag
             void    inputStandBy();
 
             AudioStreamIn                       *mInput;
             SortedVector < sp<RecordTrack> >    mTracks;
-            // mActiveTrack has dual roles:  it indicates the current active track, and
+            // mActiveTracks has dual roles:  it indicates the current active track(s), and
             // is used together with mStartStopCond to indicate start()/stop() progress
-            sp<RecordTrack>                     mActiveTrack;
+            SortedVector< sp<RecordTrack> >     mActiveTracks;
+            // generation counter for mActiveTracks
+            int                                 mActiveTracksGen;
             Condition                           mStartStopCond;
 
-            // updated by RecordThread::readInputParameters()
-            AudioResampler                      *mResampler;
-            // interleaved stereo pairs of fixed-point signed Q19.12
-            int32_t                             *mRsmpOutBuffer;
-            int16_t                             *mRsmpInBuffer; // [mFrameCount * mChannelCount]
-            size_t                              mRsmpInIndex;
-            const uint32_t                      mReqChannelCount;
-            const uint32_t                      mReqSampleRate;
-            ssize_t                             mBytesRead;
-            // sync event triggering actual audio capture. Frames read before this event will
-            // be dropped and therefore not read by the application.
-            sp<SyncEvent>                       mSyncStartEvent;
-            // number of captured frames to drop after the start sync event has been received.
-            // when < 0, maximum frames to drop before starting capture even if sync event is
-            // not received
-            ssize_t                             mFramestoDrop;
+            // resampler converts input at HAL Hz to output at AudioRecord client Hz
+            int16_t                             *mRsmpInBuffer; // see new[] for details on the size
+            size_t                              mRsmpInFrames;  // size of resampler input in frames
+            size_t                              mRsmpInFramesP2;// size rounded up to a power-of-2
+
+            // rolling index that is never cleared
+            int32_t                             mRsmpInRear;    // last filled frame + 1
 
             // For dumpsys
             const sp<NBAIO_Sink>                mTeeSink;

services/audioflinger/TrackBase.h

diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 00a91b7..58705c4 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -34,7 +34,9 @@
         RESUMING,
         ACTIVE,
         PAUSING,
-        PAUSED
+        PAUSED,
+        STARTING_1,     // for RecordTrack only
+        STARTING_2,     // for RecordTrack only
     };
 
                         TrackBase(ThreadBase *thread,
@@ -45,6 +47,7 @@
                                 size_t frameCount,
                                 const sp<IMemory>& sharedBuffer,
                                 int sessionId,
+                                int uid,
                                 bool isOut);
     virtual             ~TrackBase();
     virtual status_t    initCheck() const { return getCblk() != 0 ? NO_ERROR : NO_MEMORY; }
@@ -55,6 +58,7 @@
             sp<IMemory> getCblk() const { return mCblkMemory; }
             audio_track_cblk_t* cblk() const { return mCblk; }
             int         sessionId() const { return mSessionId; }
+            int         uid() const { return mUid; }
     virtual status_t    setSyncEvent(const sp<SyncEvent>& event);
 
 protected:
@@ -124,6 +128,7 @@
                                     // openRecord(), and then adjusted as needed
 
     const int           mSessionId;
+    int                 mUid;
     Vector < sp<SyncEvent> >mSyncEvents;
     const bool          mIsOut;
     ServerProxy*        mServerProxy;

services/audioflinger/Tracks.cpp

diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 9002f9b..92ed46a 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -68,6 +68,7 @@
             size_t frameCount,
             const sp<IMemory>& sharedBuffer,
             int sessionId,
+            int clientUid,
             bool isOut)
     :   RefBase(),
         mThread(thread),
@@ -88,6 +89,18 @@
         mId(android_atomic_inc(&nextTrackId)),
         mTerminated(false)
 {
+    // if the caller is us, trust the specified uid
+    if (IPCThreadState::self()->getCallingPid() != getpid_cached || clientUid == -1) {
+        int newclientUid = IPCThreadState::self()->getCallingUid();
+        if (clientUid != -1 && clientUid != newclientUid) {
+            ALOGW("uid %d tried to pass itself off as %d", newclientUid, clientUid);
+        }
+        clientUid = newclientUid;
+    }
+    // clientUid contains the uid of the app that is responsible for this track, so we can blame
+    // battery usage on it.
+    mUid = clientUid;
+
     // client == 0 implies sharedBuffer == 0
     ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
 
@@ -103,12 +116,11 @@
 
     if (client != 0) {
         mCblkMemory = client->heap()->allocate(size);
-        if (mCblkMemory != 0) {
-            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
-            // can't assume mCblk != NULL
-        } else {
+        if (mCblkMemory == 0 ||
+                (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) {
             ALOGE("not enough memory for AudioTrack size=%u", size);
             client->heap()->dump("AudioTrack");
+            mCblkMemory.clear();
             return;
         }
     } else {
@@ -121,7 +133,6 @@
     if (mCblk != NULL) {
         new(mCblk) audio_track_cblk_t();
         // clear all buffers
-        mCblk->frameCount_ = frameCount;
         if (sharedBuffer == 0) {
             mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
             memset(mBuffer, 0, bufferSize);
@@ -135,7 +146,7 @@
 #ifdef TEE_SINK
         if (mTeeSinkTrackEnabled) {
             NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount);
-            if (pipeFormat != Format_Invalid) {
+            if (Format_isValid(pipeFormat)) {
                 Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat);
                 size_t numCounterOffers = 0;
                 const NBAIO_Format offers[1] = {pipeFormat};
@@ -262,6 +273,11 @@
     if (!mTrack->isTimedTrack())
         return INVALID_OPERATION;
 
+    if (buffer == 0 || buffer->pointer() == NULL) {
+        ALOGE("queueTimedBuffer() buffer is 0 or has NULL pointer()");
+        return BAD_VALUE;
+    }
+
     PlaybackThread::TimedTrack* tt =
             reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
     return tt->queueTimedBuffer(buffer, pts);
@@ -288,6 +304,12 @@
     return mTrack->getTimestamp(timestamp);
 }
 
+
+void AudioFlinger::TrackHandle::signal()
+{
+    return mTrack->signal();
+}
+
 status_t AudioFlinger::TrackHandle::onTransact(
     uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
 {
@@ -307,9 +329,10 @@
             size_t frameCount,
             const sp<IMemory>& sharedBuffer,
             int sessionId,
+            int uid,
             IAudioFlinger::track_flags_t flags)
     :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer,
-            sessionId, true /*isOut*/),
+            sessionId, uid, true /*isOut*/),
     mFillingUpStatus(FS_INVALID),
     // mRetryCount initialized later when needed
     mSharedBuffer(sharedBuffer),
@@ -324,7 +347,8 @@
     mCachedVolume(1.0),
     mIsInvalid(false),
     mAudioTrackServerProxy(NULL),
-    mResumeToStopping(false)
+    mResumeToStopping(false),
+    mFlushHwPending(false)
 {
     if (mCblk != NULL) {
         if (sharedBuffer == 0) {
@@ -411,17 +435,19 @@
 
 /*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
 {
-    result.append("   Name Client Type      Fmt Chn mask Session fCount S F SRate  "
+    result.append("    Name Active Client Type      Fmt Chn mask Session fCount S F SRate  "
                   "L dB  R dB    Server Main buf  Aux Buf Flags UndFrmCnt\n");
 }
 
-void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
+void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool active)
 {
     uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
     if (isFastTrack()) {
-        sprintf(buffer, "   F %2d", mFastIndex);
+        sprintf(buffer, "    F %2d", mFastIndex);
+    } else if (mName >= AudioMixer::TRACK0) {
+        sprintf(buffer, "    %4d", mName - AudioMixer::TRACK0);
     } else {
-        sprintf(buffer, "   %4d", mName - AudioMixer::TRACK0);
+        sprintf(buffer, "    none");
     }
     track_state state = mState;
     char stateChar;
@@ -476,8 +502,9 @@
         nowInUnderrun = '?';
         break;
     }
-    snprintf(&buffer[7], size-7, " %6u %4u %08X %08X %7u %6u %1c %1d %5u %5.2g %5.2g  "
-                                 "%08X %08X %08X 0x%03X %9u%c\n",
+    snprintf(&buffer[8], size-8, " %6s %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g  "
+                                 "%08X %p %p 0x%03X %9u%c\n",
+            active ? "yes" : "no",
             (mClient == 0) ? getpid_cached : mClient->pid(),
             mStreamType,
             mFormat,
@@ -490,8 +517,8 @@
             20.0 * log10((vlr & 0xFFFF) / 4096.0),
             20.0 * log10((vlr >> 16) / 4096.0),
             mCblk->mServer,
-            (int)mMainBuffer,
-            (int)mAuxBuffer,
+            mMainBuffer,
+            mAuxBuffer,
             mCblk->mFlags,
             mAudioTrackServerProxy->getUnderrunFrames(),
             nowInUnderrun);
@@ -503,7 +530,7 @@
 
 // AudioBufferProvider interface
 status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
-        AudioBufferProvider::Buffer* buffer, int64_t pts)
+        AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
 {
     ServerProxy::Buffer buf;
     size_t desiredFrames = buffer->frameCount;
@@ -540,7 +567,7 @@
 
 // Don't call for fast tracks; the framesReady() could result in priority inversion
 bool AudioFlinger::PlaybackThread::Track::isReady() const {
-    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
+    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing() || isStopping()) {
         return true;
     }
 
@@ -553,8 +580,8 @@
     return false;
 }
 
-status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event,
-                                                    int triggerSession)
+status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused,
+                                                    int triggerSession __unused)
 {
     status_t status = NO_ERROR;
     ALOGV("start(%d), calling pid %d session %d",
@@ -603,6 +630,15 @@
         // track was already in the active list, not a problem
         if (status == ALREADY_EXISTS) {
             status = NO_ERROR;
+        } else {
+            // Acknowledge any pending flush(), so that subsequent new data isn't discarded.
+            // It is usually unsafe to access the server proxy from a binder thread.
+            // But in this case we know the mixer thread (whether normal mixer or fast mixer)
+            // isn't looking at this track yet:  we still hold the normal mixer thread lock,
+            // and for fast tracks the track is not yet in the fast mixer thread's active set.
+            ServerProxy::Buffer buffer;
+            buffer.mFrameCount = 1;
+            (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/);
         }
     } else {
         status = BAD_VALUE;
@@ -699,6 +735,7 @@
                 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload;
             }
 
+            mFlushHwPending = true;
             mResumeToStopping = false;
         } else {
             if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED &&
@@ -719,11 +756,19 @@
         // Prevent flush being lost if the track is flushed and then resumed
         // before mixer thread can run. This is important when offloading
         // because the hardware buffer could hold a large amount of audio
-        playbackThread->flushOutput_l();
         playbackThread->broadcast_l();
     }
 }
 
+// must be called with thread lock held
+void AudioFlinger::PlaybackThread::Track::flushAck()
+{
+    if (!isOffloaded())
+        return;
+
+    mFlushHwPending = false;
+}
+
 void AudioFlinger::PlaybackThread::Track::reset()
 {
     // Do not reset twice to avoid discarding data written just after a flush and before
@@ -832,6 +877,7 @@
                                         dstChain->strategy(),
                                         AUDIO_SESSION_OUTPUT_MIX,
                                         effect->id());
+            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
         }
         status = playbackThread->attachAuxEffect(this, EffectId);
     }
@@ -935,6 +981,16 @@
     mIsInvalid = true;
 }
 
+void AudioFlinger::PlaybackThread::Track::signal()
+{
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        PlaybackThread *t = (PlaybackThread *)thread.get();
+        Mutex::Autolock _l(t->mLock);
+        t->broadcast_l();
+    }
+}
+
 // ----------------------------------------------------------------------------
 
 sp<AudioFlinger::PlaybackThread::TimedTrack>
@@ -947,13 +1003,15 @@
             audio_channel_mask_t channelMask,
             size_t frameCount,
             const sp<IMemory>& sharedBuffer,
-            int sessionId) {
+            int sessionId,
+            int uid)
+{
     if (!client->reserveTimedTrack())
         return 0;
 
     return new TimedTrack(
         thread, client, streamType, sampleRate, format, channelMask, frameCount,
-        sharedBuffer, sessionId);
+        sharedBuffer, sessionId, uid);
 }
 
 AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
@@ -965,9 +1023,10 @@
             audio_channel_mask_t channelMask,
             size_t frameCount,
             const sp<IMemory>& sharedBuffer,
-            int sessionId)
+            int sessionId,
+            int uid)
     : Track(thread, client, streamType, sampleRate, format, channelMask,
-            frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED),
+            frameCount, sharedBuffer, sessionId, uid, IAudioFlinger::TRACK_TIMED),
       mQueueHeadInFlight(false),
       mTrimQueueHeadOnRelease(false),
       mFramesPendingInQueue(0),
@@ -1018,11 +1077,8 @@
     }
 
     sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
-    if (newBuffer == NULL) {
-        newBuffer = mTimedMemoryDealer->allocate(size);
-        if (newBuffer == NULL) {
-            return NO_MEMORY;
-        }
+    if (newBuffer == 0 || newBuffer->pointer() == NULL) {
+        return NO_MEMORY;
     }
 
     *buffer = newBuffer;
@@ -1121,7 +1177,7 @@
 
 void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
         const TimedBuffer& buf,
-        const char* logTag) {
+        const char* logTag __unused) {
     uint32_t bufBytes        = buf.buffer()->size();
     uint32_t consumedAlready = buf.position();
 
@@ -1462,9 +1518,10 @@
             uint32_t sampleRate,
             audio_format_t format,
             audio_channel_mask_t channelMask,
-            size_t frameCount)
+            size_t frameCount,
+            int uid)
     :   Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
-                NULL, 0, IAudioFlinger::TRACK_DEFAULT),
+                NULL, 0, uid, IAudioFlinger::TRACK_DEFAULT),
     mActive(false), mSourceThread(sourceThread), mClientProxy(NULL)
 {
 
@@ -1472,9 +1529,9 @@
         mOutBuffer.frameCount = 0;
         playbackThread->mTracks.add(this);
         ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
-                "mCblk->frameCount_ %u, mChannelMask 0x%08x",
+                "frameCount %u, mChannelMask 0x%08x",
                 mCblk, mBuffer,
-                mCblk->frameCount_, mChannelMask);
+                frameCount, mChannelMask);
         // since client and server are in the same process,
         // the buffer has the same virtual address on both sides
         mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize);
@@ -1724,25 +1781,43 @@
             audio_format_t format,
             audio_channel_mask_t channelMask,
             size_t frameCount,
-            int sessionId)
+            int sessionId,
+            int uid)
     :   TrackBase(thread, client, sampleRate, format,
-                  channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, false /*isOut*/),
-        mOverflow(false)
+                  channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, uid, false /*isOut*/),
+        mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0),
+        // See real initialization of mRsmpInFront at RecordThread::start()
+        mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL)
 {
     ALOGV("RecordTrack constructor");
     if (mCblk != NULL) {
         mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize);
     }
+
+    uint32_t channelCount = popcount(channelMask);
+    // FIXME I don't understand either of the channel count checks
+    if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 &&
+            channelCount <= FCC_2) {
+        // sink SR
+        mResampler = AudioResampler::create(16, thread->mChannelCount, sampleRate);
+        // source SR
+        mResampler->setSampleRate(thread->mSampleRate);
+        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
+        mResamplerBufferProvider = new ResamplerBufferProvider(this);
+    }
 }
 
 AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
 {
     ALOGV("%s", __func__);
+    delete mResampler;
+    delete[] mRsmpOutBuffer;
+    delete mResamplerBufferProvider;
 }
 
 // AudioBufferProvider interface
 status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
-        int64_t pts)
+        int64_t pts __unused)
 {
     ServerProxy::Buffer buf;
     buf.mFrameCount = buffer->frameCount;
@@ -1810,19 +1885,45 @@
 
 /*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
 {
-    result.append("Client Fmt Chn mask Session S   Server fCount\n");
+    result.append("    Active Client Fmt Chn mask Session S   Server fCount Resampling\n");
 }
 
-void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
+void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active)
 {
-    snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6u\n",
+    snprintf(buffer, size, "    %6s %6u %3u %08X %7u %1d %08X %6zu %10d\n",
+            active ? "yes" : "no",
             (mClient == 0) ? getpid_cached : mClient->pid(),
             mFormat,
             mChannelMask,
             mSessionId,
             mState,
             mCblk->mServer,
-            mFrameCount);
+            mFrameCount,
+            mResampler != NULL);
+
+}
+
+void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event)
+{
+    if (event == mSyncStartEvent) {
+        ssize_t framesToDrop = 0;
+        sp<ThreadBase> threadBase = mThread.promote();
+        if (threadBase != 0) {
+            // TODO: use actual buffer filling status instead of 2 buffers when info is available
+            // from audio HAL
+            framesToDrop = threadBase->mFrameCount * 2;
+        }
+        mFramesToDrop = framesToDrop;
+    }
+}
+
+void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent()
+{
+    if (mSyncStartEvent != 0) {
+        mSyncStartEvent->cancel();
+        mSyncStartEvent.clear();
+    }
+    mFramesToDrop = 0;
 }
 
 }; // namespace android

services/audioflinger/test-resample.cpp

diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
index 7a314cf..3ab3ba9 100644
--- a/services/audioflinger/test-resample.cpp
+++ b/services/audioflinger/test-resample.cpp
@@ -26,77 +26,104 @@
 #include <errno.h>
 #include <time.h>
 #include <math.h>
+#include <audio_utils/sndfile.h>
+#include <utils/Vector.h>
 
 using namespace android;
 
-struct HeaderWav {
-    HeaderWav(size_t size, int nc, int sr, int bits) {
-        strncpy(RIFF, "RIFF", 4);
-        chunkSize = size + sizeof(HeaderWav);
-        strncpy(WAVE, "WAVE", 4);
-        strncpy(fmt,  "fmt ", 4);
-        fmtSize = 16;
-        audioFormat = 1;
-        numChannels = nc;
-        samplesRate = sr;
-        byteRate = sr * numChannels * (bits/8);
-        align = nc*(bits/8);
-        bitsPerSample = bits;
-        strncpy(data, "data", 4);
-        dataSize = size;
-    }
-
-    char RIFF[4];           // RIFF
-    uint32_t chunkSize;     // File size
-    char WAVE[4];        // WAVE
-    char fmt[4];            // fmt\0
-    uint32_t fmtSize;       // fmt size
-    uint16_t audioFormat;   // 1=PCM
-    uint16_t numChannels;   // num channels
-    uint32_t samplesRate;   // sample rate in hz
-    uint32_t byteRate;      // Bps
-    uint16_t align;         // 2=16-bit mono, 4=16-bit stereo
-    uint16_t bitsPerSample; // bits per sample
-    char data[4];           // "data"
-    uint32_t dataSize;      // size
-};
+bool gVerbose = false;
 
 static int usage(const char* name) {
-    fprintf(stderr,"Usage: %s [-p] [-h] [-s] [-q {dq|lq|mq|hq|vhq}] [-i input-sample-rate] "
-                   "[-o output-sample-rate] [<input-file>] <output-file>\n", name);
+    fprintf(stderr,"Usage: %s [-p] [-h] [-v] [-s] [-q {dq|lq|mq|hq|vhq|dlq|dmq|dhq}]"
+                   " [-i input-sample-rate] [-o output-sample-rate] [-O csv] [-P csv] [<input-file>]"
+                   " <output-file>\n", name);
     fprintf(stderr,"    -p    enable profiling\n");
     fprintf(stderr,"    -h    create wav file\n");
-    fprintf(stderr,"    -s    stereo\n");
+    fprintf(stderr,"    -v    verbose : log buffer provider calls\n");
+    fprintf(stderr,"    -s    stereo (ignored if input file is specified)\n");
     fprintf(stderr,"    -q    resampler quality\n");
     fprintf(stderr,"              dq  : default quality\n");
     fprintf(stderr,"              lq  : low quality\n");
     fprintf(stderr,"              mq  : medium quality\n");
     fprintf(stderr,"              hq  : high quality\n");
     fprintf(stderr,"              vhq : very high quality\n");
-    fprintf(stderr,"    -i    input file sample rate\n");
+    fprintf(stderr,"              dlq : dynamic low quality\n");
+    fprintf(stderr,"              dmq : dynamic medium quality\n");
+    fprintf(stderr,"              dhq : dynamic high quality\n");
+    fprintf(stderr,"    -i    input file sample rate (ignored if input file is specified)\n");
     fprintf(stderr,"    -o    output file sample rate\n");
+    fprintf(stderr,"    -O    # frames output per call to resample() in CSV format\n");
+    fprintf(stderr,"    -P    # frames provided per call to resample() in CSV format\n");
     return -1;
 }
 
+// Convert a list of integers in CSV format to a Vector of those values.
+// Returns the number of elements in the list, or -1 on error.
+int parseCSV(const char *string, Vector<int>& values)
+{
+    // pass 1: count the number of values and do syntax check
+    size_t numValues = 0;
+    bool hadDigit = false;
+    for (const char *p = string; ; ) {
+        switch (*p++) {
+        case '0': case '1': case '2': case '3': case '4':
+        case '5': case '6': case '7': case '8': case '9':
+            hadDigit = true;
+            break;
+        case '\0':
+            if (hadDigit) {
+                // pass 2: allocate and initialize vector of values
+                values.resize(++numValues);
+                values.editItemAt(0) = atoi(p = optarg);
+                for (size_t i = 1; i < numValues; ) {
+                    if (*p++ == ',') {
+                        values.editItemAt(i++) = atoi(p);
+                    }
+                }
+                return numValues;
+            }
+            // fall through
+        case ',':
+            if (hadDigit) {
+                hadDigit = false;
+                numValues++;
+                break;
+            }
+            // fall through
+        default:
+            return -1;
+        }
+    }
+}
+
 int main(int argc, char* argv[]) {
 
     const char* const progname = argv[0];
-    bool profiling = false;
+    bool profileResample = false;
+    bool profileFilter = false;
     bool writeHeader = false;
     int channels = 1;
     int input_freq = 0;
     int output_freq = 0;
     AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY;
+    Vector<int> Ovalues;
+    Vector<int> Pvalues;
 
     int ch;
-    while ((ch = getopt(argc, argv, "phsq:i:o:")) != -1) {
+    while ((ch = getopt(argc, argv, "pfhvsq:i:o:O:P:")) != -1) {
         switch (ch) {
         case 'p':
-            profiling = true;
+            profileResample = true;
+            break;
+        case 'f':
+            profileFilter = true;
             break;
         case 'h':
             writeHeader = true;
             break;
+        case 'v':
+            gVerbose = true;
+            break;
         case 's':
             channels = 2;
             break;
@@ -111,6 +138,12 @@
                 quality = AudioResampler::HIGH_QUALITY;
             else if (!strcmp(optarg, "vhq"))
                 quality = AudioResampler::VERY_HIGH_QUALITY;
+            else if (!strcmp(optarg, "dlq"))
+                quality = AudioResampler::DYN_LOW_QUALITY;
+            else if (!strcmp(optarg, "dmq"))
+                quality = AudioResampler::DYN_MED_QUALITY;
+            else if (!strcmp(optarg, "dhq"))
+                quality = AudioResampler::DYN_HIGH_QUALITY;
             else {
                 usage(progname);
                 return -1;
@@ -122,6 +155,18 @@
         case 'o':
             output_freq = atoi(optarg);
             break;
+        case 'O':
+            if (parseCSV(optarg, Ovalues) < 0) {
+                fprintf(stderr, "incorrect syntax for -O option\n");
+                return -1;
+            }
+            break;
+        case 'P':
+            if (parseCSV(optarg, Pvalues) < 0) {
+                fprintf(stderr, "incorrect syntax for -P option\n");
+                return -1;
+            }
+            break;
         case '?':
         default:
             usage(progname);
@@ -148,25 +193,22 @@
     size_t input_size;
     void* input_vaddr;
     if (argc == 2) {
-        struct stat st;
-        if (stat(file_in, &st) < 0) {
-            fprintf(stderr, "stat: %s\n", strerror(errno));
-            return -1;
+        SF_INFO info;
+        info.format = 0;
+        SNDFILE *sf = sf_open(file_in, SFM_READ, &info);
+        if (sf == NULL) {
+            perror(file_in);
+            return EXIT_FAILURE;
         }
-
-        int input_fd = open(file_in, O_RDONLY);
-        if (input_fd < 0) {
-            fprintf(stderr, "open: %s\n", strerror(errno));
-            return -1;
-        }
-
-        input_size = st.st_size;
-        input_vaddr = mmap(0, input_size, PROT_READ, MAP_PRIVATE, input_fd, 0);
-        if (input_vaddr == MAP_FAILED ) {
-            fprintf(stderr, "mmap: %s\n", strerror(errno));
-            return -1;
-        }
+        input_size = info.frames * info.channels * sizeof(short);
+        input_vaddr = malloc(input_size);
+        (void) sf_readf_short(sf, (short *) input_vaddr, info.frames);
+        sf_close(sf);
+        channels = info.channels;
+        input_freq = info.samplerate;
     } else {
+        // data for testing is exactly (input sampling rate/1000)/2 seconds
+        // so 44.1khz input is 22.05 seconds
         double k = 1000; // Hz / s
         double time = (input_freq / 2) / k;
         size_t input_frames = size_t(input_freq * time);
@@ -178,7 +220,7 @@
             double y = sin(M_PI * k * t * t);
             int16_t yi = floor(y * 32767.0 + 0.5);
             for (size_t j=0 ; j<(size_t)channels ; j++) {
-                in[i*channels + j] = yi / (1+j);
+                in[i*channels + j] = yi / (1+j); // right ch. 1/2 left ch.
             }
         }
     }
@@ -186,89 +228,263 @@
     // ----------------------------------------------------------
 
     class Provider: public AudioBufferProvider {
-        int16_t* mAddr;
-        size_t mNumFrames;
+        int16_t* const  mAddr;      // base address
+        const size_t    mNumFrames; // total frames
+        const int       mChannels;
+        size_t          mNextFrame; // index of next frame to provide
+        size_t          mUnrel;     // number of frames not yet released
+        const Vector<int> mPvalues; // number of frames provided per call
+        size_t          mNextPidx;  // index of next entry in mPvalues to use
     public:
-        Provider(const void* addr, size_t size, int channels) {
-            mAddr = (int16_t*) addr;
-            mNumFrames = size / (channels*sizeof(int16_t));
+        Provider(const void* addr, size_t size, int channels, const Vector<int>& Pvalues)
+          : mAddr((int16_t*) addr),
+            mNumFrames(size / (channels*sizeof(int16_t))),
+            mChannels(channels),
+            mNextFrame(0), mUnrel(0), mPvalues(Pvalues), mNextPidx(0) {
         }
         virtual status_t getNextBuffer(Buffer* buffer,
                 int64_t pts = kInvalidPTS) {
-            buffer->frameCount = mNumFrames;
-            buffer->i16 = mAddr;
-            return NO_ERROR;
+            (void)pts; // suppress warning
+            size_t requestedFrames = buffer->frameCount;
+            if (requestedFrames > mNumFrames - mNextFrame) {
+                buffer->frameCount = mNumFrames - mNextFrame;
+            }
+            if (!mPvalues.isEmpty()) {
+                size_t provided = mPvalues[mNextPidx++];
+                printf("mPvalue[%d]=%u not %u\n", mNextPidx-1, provided, buffer->frameCount);
+                if (provided < buffer->frameCount) {
+                    buffer->frameCount = provided;
+                }
+                if (mNextPidx >= mPvalues.size()) {
+                    mNextPidx = 0;
+                }
+            }
+            if (gVerbose) {
+                printf("getNextBuffer() requested %u frames out of %u frames available,"
+                        " and returned %u frames\n",
+                        requestedFrames, mNumFrames - mNextFrame, buffer->frameCount);
+            }
+            mUnrel = buffer->frameCount;
+            if (buffer->frameCount > 0) {
+                buffer->i16 = &mAddr[mChannels * mNextFrame];
+                return NO_ERROR;
+            } else {
+                buffer->i16 = NULL;
+                return NOT_ENOUGH_DATA;
+            }
         }
         virtual void releaseBuffer(Buffer* buffer) {
+            if (buffer->frameCount > mUnrel) {
+                fprintf(stderr, "ERROR releaseBuffer() released %u frames but only %u available "
+                        "to release\n", buffer->frameCount, mUnrel);
+                mNextFrame += mUnrel;
+                mUnrel = 0;
+            } else {
+                if (gVerbose) {
+                    printf("releaseBuffer() released %u frames out of %u frames available "
+                            "to release\n", buffer->frameCount, mUnrel);
+                }
+                mNextFrame += buffer->frameCount;
+                mUnrel -= buffer->frameCount;
+            }
+            buffer->frameCount = 0;
+            buffer->i16 = NULL;
         }
-    } provider(input_vaddr, input_size, channels);
+        void reset() {
+            mNextFrame = 0;
+        }
+    } provider(input_vaddr, input_size, channels, Pvalues);
 
     size_t input_frames = input_size / (channels * sizeof(int16_t));
+    if (gVerbose) {
+        printf("%u input frames\n", input_frames);
+    }
     size_t output_size = 2 * 4 * ((int64_t) input_frames * output_freq) / input_freq;
     output_size &= ~7; // always stereo, 32-bits
 
-    void* output_vaddr = malloc(output_size);
-
-    if (profiling) {
+    if (profileFilter) {
+        // Check how fast sample rate changes are that require filter changes.
+        // The delta sample rate changes must indicate a downsampling ratio,
+        // and must be larger than 10% changes.
+        //
+        // On fast devices, filters should be generated between 0.1ms - 1ms.
+        // (single threaded).
         AudioResampler* resampler = AudioResampler::create(16, channels,
-                output_freq, quality);
-
-        size_t out_frames = output_size/8;
-        resampler->setSampleRate(input_freq);
-        resampler->setVolume(0x1000, 0x1000);
-
-        memset(output_vaddr, 0, output_size);
+                8000, quality);
+        int looplimit = 100;
         timespec start, end;
         clock_gettime(CLOCK_MONOTONIC, &start);
-        resampler->resample((int*) output_vaddr, out_frames, &provider);
-        resampler->resample((int*) output_vaddr, out_frames, &provider);
-        resampler->resample((int*) output_vaddr, out_frames, &provider);
-        resampler->resample((int*) output_vaddr, out_frames, &provider);
+        for (int i = 0; i < looplimit; ++i) {
+            resampler->setSampleRate(9000);
+            resampler->setSampleRate(12000);
+            resampler->setSampleRate(20000);
+            resampler->setSampleRate(30000);
+        }
         clock_gettime(CLOCK_MONOTONIC, &end);
         int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
         int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
-        int64_t time = (end_ns - start_ns)/4;
-        printf("%f Mspl/s\n", out_frames/(time/1e9)/1e6);
+        int64_t time = end_ns - start_ns;
+        printf("%.2f sample rate changes with filter calculation/sec\n",
+                looplimit * 4 / (time / 1e9));
 
+        // Check how fast sample rate changes are without filter changes.
+        // This should be very fast, probably 0.1us - 1us per sample rate
+        // change.
+        resampler->setSampleRate(1000);
+        looplimit = 1000;
+        clock_gettime(CLOCK_MONOTONIC, &start);
+        for (int i = 0; i < looplimit; ++i) {
+            resampler->setSampleRate(1000+i);
+        }
+        clock_gettime(CLOCK_MONOTONIC, &end);
+        start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
+        end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
+        time = end_ns - start_ns;
+        printf("%.2f sample rate changes without filter calculation/sec\n",
+                looplimit / (time / 1e9));
+        resampler->reset();
         delete resampler;
     }
 
+    void* output_vaddr = malloc(output_size);
     AudioResampler* resampler = AudioResampler::create(16, channels,
             output_freq, quality);
     size_t out_frames = output_size/8;
+
+    /* set volume precision to 12 bits, so the volume scale is 1<<12.
+     * This means the "integer" part fits in the Q19.12 precision
+     * representation of output int32_t.
+     *
+     * Generally 0 < volumePrecision <= 14 (due to the limits of
+     * int16_t values for Volume). volumePrecision cannot be 0 due
+     * to rounding and shifts.
+     */
+    const int volumePrecision = 12; // in bits
+
     resampler->setSampleRate(input_freq);
-    resampler->setVolume(0x1000, 0x1000);
+    resampler->setVolume(1 << volumePrecision, 1 << volumePrecision);
+
+    if (profileResample) {
+        /*
+         * For profiling on mobile devices, upon experimentation
+         * it is better to run a few trials with a shorter loop limit,
+         * and take the minimum time.
+         *
+         * Long tests can cause CPU temperature to build up and thermal throttling
+         * to reduce CPU frequency.
+         *
+         * For frequency checks (index=0, or 1, etc.):
+         * "cat /sys/devices/system/cpu/cpu${index}/cpufreq/scaling_*_freq"
+         *
+         * For temperature checks (index=0, or 1, etc.):
+         * "cat /sys/class/thermal/thermal_zone${index}/temp"
+         *
+         * Another way to avoid thermal throttling is to fix the CPU frequency
+         * at a lower level which prevents excessive temperatures.
+         */
+        const int trials = 4;
+        const int looplimit = 4;
+        timespec start, end;
+        int64_t time;
+
+        for (int n = 0; n < trials; ++n) {
+            clock_gettime(CLOCK_MONOTONIC, &start);
+            for (int i = 0; i < looplimit; ++i) {
+                resampler->resample((int*) output_vaddr, out_frames, &provider);
+                provider.reset(); //  during benchmarking reset only the provider
+            }
+            clock_gettime(CLOCK_MONOTONIC, &end);
+            int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
+            int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
+            int64_t diff_ns = end_ns - start_ns;
+            if (n == 0 || diff_ns < time) {
+                time = diff_ns;   // save the best out of our trials.
+            }
+        }
+        // Mfrms/s is "Millions of output frames per second".
+        printf("quality: %d  channels: %d  msec: %lld  Mfrms/s: %.2lf\n",
+                quality, channels, time/1000000, out_frames * looplimit / (time / 1e9) / 1e6);
+        resampler->reset();
+    }
 
     memset(output_vaddr, 0, output_size);
-    resampler->resample((int*) output_vaddr, out_frames, &provider);
+    if (gVerbose) {
+        printf("resample() %u output frames\n", out_frames);
+    }
+    if (Ovalues.isEmpty()) {
+        Ovalues.push(out_frames);
+    }
+    for (size_t i = 0, j = 0; i < out_frames; ) {
+        size_t thisFrames = Ovalues[j++];
+        if (j >= Ovalues.size()) {
+            j = 0;
+        }
+        if (thisFrames == 0 || thisFrames > out_frames - i) {
+            thisFrames = out_frames - i;
+        }
+        resampler->resample((int*) output_vaddr + 2*i, thisFrames, &provider);
+        i += thisFrames;
+    }
+    if (gVerbose) {
+        printf("resample() complete\n");
+    }
+    resampler->reset();
+    if (gVerbose) {
+        printf("reset() complete\n");
+    }
+    delete resampler;
+    resampler = NULL;
 
-    // down-mix (we just truncate and keep the left channel)
+    // mono takes left channel only
+    // stereo right channel is half amplitude of stereo left channel (due to input creation)
     int32_t* out = (int32_t*) output_vaddr;
     int16_t* convert = (int16_t*) malloc(out_frames * channels * sizeof(int16_t));
+
+    // round to half towards zero and saturate at int16 (non-dithered)
+    const int roundVal = (1<<(volumePrecision-1)) - 1; // volumePrecision > 0
+
     for (size_t i = 0; i < out_frames; i++) {
-        for (int j=0 ; j<channels ; j++) {
-            int32_t s = out[i * 2 + j] >> 12;
-            if (s > 32767)       s =  32767;
-            else if (s < -32768) s = -32768;
+        for (int j = 0; j < channels; j++) {
+            int32_t s = out[i * 2 + j] + roundVal; // add offset here
+            if (s < 0) {
+                s = (s + 1) >> volumePrecision; // round to 0
+                if (s < -32768) {
+                    s = -32768;
+                }
+            } else {
+                s = s >> volumePrecision;
+                if (s > 32767) {
+                    s = 32767;
+                }
+            }
             convert[i * channels + j] = int16_t(s);
         }
     }
 
     // write output to disk
-    int output_fd = open(file_out, O_WRONLY | O_CREAT | O_TRUNC,
-            S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH);
-    if (output_fd < 0) {
-        fprintf(stderr, "open: %s\n", strerror(errno));
-        return -1;
-    }
-
     if (writeHeader) {
-        HeaderWav wav(out_frames * channels * sizeof(int16_t), channels, output_freq, 16);
-        write(output_fd, &wav, sizeof(wav));
+        SF_INFO info;
+        info.frames = 0;
+        info.samplerate = output_freq;
+        info.channels = channels;
+        info.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16;
+        SNDFILE *sf = sf_open(file_out, SFM_WRITE, &info);
+        if (sf == NULL) {
+            perror(file_out);
+            return EXIT_FAILURE;
+        }
+        (void) sf_writef_short(sf, convert, out_frames);
+        sf_close(sf);
+    } else {
+        int output_fd = open(file_out, O_WRONLY | O_CREAT | O_TRUNC,
+                S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH);
+        if (output_fd < 0) {
+            perror(file_out);
+            return EXIT_FAILURE;
+        }
+        write(output_fd, convert, out_frames * channels * sizeof(int16_t));
+        close(output_fd);
     }
 
-    write(output_fd, convert, out_frames * channels * sizeof(int16_t));
-    close(output_fd);
-
-    return 0;
+    return EXIT_SUCCESS;
 }

services/camera/libcameraservice/Android.mk

diff --git a/services/camera/libcameraservice/Android.mk b/services/camera/libcameraservice/Android.mk
index d659ebb..51ba698 100644
--- a/services/camera/libcameraservice/Android.mk
+++ b/services/camera/libcameraservice/Android.mk
@@ -33,7 +33,9 @@
     device3/Camera3InputStream.cpp \
     device3/Camera3OutputStream.cpp \
     device3/Camera3ZslStream.cpp \
+    device3/StatusTracker.cpp \
     gui/RingBufferConsumer.cpp \
+    utils/CameraTraces.cpp \
 
 LOCAL_SHARED_LIBRARIES:= \
     libui \

services/camera/libcameraservice/CameraService.cpp

diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index 5e84aaf..79fbf76 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -32,6 +32,7 @@
 #include <gui/Surface.h>
 #include <hardware/hardware.h>
 #include <media/AudioSystem.h>
+#include <media/IMediaHTTPService.h>
 #include <media/mediaplayer.h>
 #include <utils/Errors.h>
 #include <utils/Log.h>
@@ -42,6 +43,7 @@
 #include "api1/Camera2Client.h"
 #include "api_pro/ProCamera2Client.h"
 #include "api2/CameraDeviceClient.h"
+#include "utils/CameraTraces.h"
 #include "CameraDeviceFactory.h"
 
 namespace android {
@@ -255,6 +257,12 @@
         return BAD_VALUE;
     }
 
+    if (getDeviceVersion(cameraId, &facing) <= CAMERA_DEVICE_API_VERSION_2_1) {
+        // Disable HAL2.x support for camera2 API for now.
+        ALOGW("%s: HAL2.x doesn't support getCameraCharacteristics for now", __FUNCTION__);
+        return BAD_VALUE;
+    }
+
     struct camera_info info;
     status_t ret = mModule->get_camera_info(cameraId, &info);
     *cameraInfo = info.static_camera_characteristics;
@@ -859,7 +867,7 @@
 
 MediaPlayer* CameraService::newMediaPlayer(const char *file) {
     MediaPlayer* mp = new MediaPlayer();
-    if (mp->setDataSource(file, NULL) == NO_ERROR) {
+    if (mp->setDataSource(NULL /* httpService */, file, NULL) == NO_ERROR) {
         mp->setAudioStreamType(AUDIO_STREAM_ENFORCED_AUDIBLE);
         mp->prepare();
     } else {
@@ -1035,13 +1043,13 @@
 // ----------------------------------------------------------------------------
 
 Mutex* CameraService::Client::getClientLockFromCookie(void* user) {
-    return gCameraService->getClientLockById((int) user);
+    return gCameraService->getClientLockById((int)(intptr_t) user);
 }
 
 // Provide client pointer for callbacks. Client lock returned from getClientLockFromCookie should
 // be acquired for this to be safe
 CameraService::Client* CameraService::Client::getClientFromCookie(void* user) {
-    BasicClient *basicClient = gCameraService->getClientByIdUnsafe((int) user);
+    BasicClient *basicClient = gCameraService->getClientByIdUnsafe((int)(intptr_t) user);
     // OK: only CameraClient calls this, and they already cast anyway.
     Client* client = static_cast<Client*>(basicClient);
 
@@ -1213,6 +1221,10 @@
 
         if (locked) mServiceLock.unlock();
 
+        // Dump camera traces if there were any
+        write(fd, "\n", 1);
+        camera3::CameraTraces::dump(fd, args);
+
         // change logging level
         int n = args.size();
         for (int i = 0; i + 1 < n; i++) {

services/camera/libcameraservice/api1/Camera2Client.cpp

diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp
index 09829ea..80b7cd4 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.cpp
+++ b/services/camera/libcameraservice/api1/Camera2Client.cpp
@@ -14,10 +14,11 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "Camera2"
+#define LOG_TAG "Camera2Client"
 #define ATRACE_TAG ATRACE_TAG_CAMERA
 //#define LOG_NDEBUG 0
 
+#include <inttypes.h>
 #include <utils/Log.h>
 #include <utils/Trace.h>
 
@@ -76,13 +77,15 @@
         return res;
     }
 
-    SharedParameters::Lock l(mParameters);
+    {
+        SharedParameters::Lock l(mParameters);
 
-    res = l.mParameters.initialize(&(mDevice->info()));
-    if (res != OK) {
-        ALOGE("%s: Camera %d: unable to build defaults: %s (%d)",
-                __FUNCTION__, mCameraId, strerror(-res), res);
-        return NO_INIT;
+        res = l.mParameters.initialize(&(mDevice->info()));
+        if (res != OK) {
+            ALOGE("%s: Camera %d: unable to build defaults: %s (%d)",
+                    __FUNCTION__, mCameraId, strerror(-res), res);
+            return NO_INIT;
+        }
     }
 
     String8 threadName;
@@ -135,6 +138,7 @@
     mCallbackProcessor->run(threadName.string());
 
     if (gLogLevel >= 1) {
+        SharedParameters::Lock l(mParameters);
         ALOGD("%s: Default parameters converted from camera %d:", __FUNCTION__,
               mCameraId);
         ALOGD("%s", l.mParameters.paramsFlattened.string());
@@ -190,7 +194,7 @@
         result.appendFormat("    GPS lat x long x alt: %f x %f x %f\n",
                 p.gpsCoordinates[0], p.gpsCoordinates[1],
                 p.gpsCoordinates[2]);
-        result.appendFormat("    GPS timestamp: %lld\n",
+        result.appendFormat("    GPS timestamp: %" PRId64 "\n",
                 p.gpsTimestamp);
         result.appendFormat("    GPS processing method: %s\n",
                 p.gpsProcessingMethod.string());
@@ -234,7 +238,7 @@
 
     result.append("    Scene mode: ");
     switch (p.sceneMode) {
-        case ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED:
+        case ANDROID_CONTROL_SCENE_MODE_DISABLED:
             result.append("AUTO\n"); break;
         CASE_APPEND_ENUM(ANDROID_CONTROL_SCENE_MODE_ACTION)
         CASE_APPEND_ENUM(ANDROID_CONTROL_SCENE_MODE_PORTRAIT)
@@ -353,6 +357,10 @@
         result.appendFormat("    meteringCropRegion\n");
         haveQuirk = true;
     }
+    if (p.quirks.partialResults) {
+        result.appendFormat("    usePartialResult\n");
+        haveQuirk = true;
+    }
     if (!haveQuirk) {
         result.appendFormat("    none\n");
     }
@@ -399,12 +407,6 @@
         l.mParameters.state = Parameters::DISCONNECTED;
     }
 
-    mStreamingProcessor->deletePreviewStream();
-    mStreamingProcessor->deleteRecordingStream();
-    mJpegProcessor->deleteStream();
-    mCallbackProcessor->deleteStream();
-    mZslProcessor->deleteStream();
-
     mStreamingProcessor->requestExit();
     mFrameProcessor->requestExit();
     mCaptureSequencer->requestExit();
@@ -421,6 +423,14 @@
     mZslProcessorThread->join();
     mCallbackProcessor->join();
 
+    ALOGV("Camera %d: Deleting streams", mCameraId);
+
+    mStreamingProcessor->deletePreviewStream();
+    mStreamingProcessor->deleteRecordingStream();
+    mJpegProcessor->deleteStream();
+    mCallbackProcessor->deleteStream();
+    mZslProcessor->deleteStream();
+
     ALOGV("Camera %d: Disconnecting device", mCameraId);
 
     mDevice->disconnect();
@@ -724,6 +734,7 @@
         return OK;
     }
     params.state = Parameters::STOPPED;
+    int lastPreviewStreamId = mStreamingProcessor->getPreviewStreamId();
 
     res = mStreamingProcessor->updatePreviewStream(params);
     if (res != OK) {
@@ -732,6 +743,8 @@
         return res;
     }
 
+    bool previewStreamChanged = mStreamingProcessor->getPreviewStreamId() != lastPreviewStreamId;
+
     // We could wait to create the JPEG output stream until first actual use
     // (first takePicture call). However, this would substantially increase the
     // first capture latency on HAL3 devices, and potentially on some HAL2
@@ -781,6 +794,19 @@
             return res;
         }
         outputStreams.push(getCallbackStreamId());
+    } else if (previewStreamChanged && mCallbackProcessor->getStreamId() != NO_STREAM) {
+        /**
+         * Delete the unused callback stream when preview stream is changed and
+         * preview is not enabled. Don't need stop preview stream as preview is in
+         * STOPPED state now.
+         */
+        ALOGV("%s: Camera %d: Delete unused preview callback stream.",  __FUNCTION__, mCameraId);
+        res = mCallbackProcessor->deleteStream();
+        if (res != OK) {
+            ALOGE("%s: Camera %d: Unable to delete callback stream %s (%d)",
+                    __FUNCTION__, mCameraId, strerror(-res), res);
+            return res;
+        }
     }
     if (params.zslMode && !params.recordingHint) {
         res = updateProcessorStream(mZslProcessor, params);
@@ -790,6 +816,8 @@
             return res;
         }
         outputStreams.push(getZslStreamId());
+    } else {
+        mZslProcessor->deleteStream();
     }
 
     outputStreams.push(getPreviewStreamId());
@@ -1136,7 +1164,7 @@
          * Handle quirk mode for AF in scene modes
          */
         if (l.mParameters.quirks.triggerAfWithAuto &&
-                l.mParameters.sceneMode != ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED &&
+                l.mParameters.sceneMode != ANDROID_CONTROL_SCENE_MODE_DISABLED &&
                 l.mParameters.focusMode != Parameters::FOCUS_MODE_AUTO &&
                 !l.mParameters.focusingAreas[0].isEmpty()) {
             ALOGV("%s: Quirk: Switching from focusMode %d to AUTO",
@@ -1149,6 +1177,8 @@
         l.mParameters.currentAfTriggerId = ++l.mParameters.afTriggerCounter;
         triggerId = l.mParameters.currentAfTriggerId;
     }
+    ATRACE_ASYNC_BEGIN(kAutofocusLabel, triggerId);
+
     syncWithDevice();
 
     mDevice->triggerAutofocus(triggerId);
@@ -1171,6 +1201,12 @@
                 l.mParameters.focusMode == Parameters::FOCUS_MODE_INFINITY) {
             return OK;
         }
+
+        // An active AF trigger is canceled
+        if (l.mParameters.afTriggerCounter == l.mParameters.currentAfTriggerId) {
+            ATRACE_ASYNC_END(kAutofocusLabel, l.mParameters.currentAfTriggerId);
+        }
+
         triggerId = ++l.mParameters.afTriggerCounter;
 
         // When using triggerAfWithAuto quirk, may need to reset focus mode to
@@ -1199,6 +1235,7 @@
     status_t res;
     if ( (res = checkPid(__FUNCTION__) ) != OK) return res;
 
+    int takePictureCounter;
     {
         SharedParameters::Lock l(mParameters);
         switch (l.mParameters.state) {
@@ -1237,8 +1274,11 @@
                     __FUNCTION__, mCameraId, strerror(-res), res);
             return res;
         }
+        takePictureCounter = ++l.mParameters.takePictureCounter;
     }
 
+    ATRACE_ASYNC_BEGIN(kTakepictureLabel, takePictureCounter);
+
     // Need HAL to have correct settings before (possibly) triggering precapture
     syncWithDevice();
 
@@ -1466,7 +1506,24 @@
     bool afInMotion = false;
     {
         SharedParameters::Lock l(mParameters);
+        // Trace end of AF state
+        char tmp[32];
+        if (l.mParameters.afStateCounter > 0) {
+            camera_metadata_enum_snprint(
+                ANDROID_CONTROL_AF_STATE, l.mParameters.focusState, tmp, sizeof(tmp));
+            ATRACE_ASYNC_END(tmp, l.mParameters.afStateCounter);
+        }
+
+        // Update state
         l.mParameters.focusState = newState;
+        l.mParameters.afStateCounter++;
+
+        // Trace start of AF state
+
+        camera_metadata_enum_snprint(
+            ANDROID_CONTROL_AF_STATE, l.mParameters.focusState, tmp, sizeof(tmp));
+        ATRACE_ASYNC_BEGIN(tmp, l.mParameters.afStateCounter);
+
         switch (l.mParameters.focusMode) {
             case Parameters::FOCUS_MODE_AUTO:
             case Parameters::FOCUS_MODE_MACRO:
@@ -1560,6 +1617,7 @@
         }
     }
     if (sendCompletedMessage) {
+        ATRACE_ASYNC_END(kAutofocusLabel, triggerId);
         SharedCameraCallbacks::Lock l(mSharedCameraCallbacks);
         if (l.mRemoteCallback != 0) {
             l.mRemoteCallback->notifyCallback(CAMERA_MSG_FOCUS,
@@ -1769,4 +1827,7 @@
     return res;
 }
 
+const char* Camera2Client::kAutofocusLabel = "autofocus";
+const char* Camera2Client::kTakepictureLabel = "take_picture";
+
 } // namespace android

services/camera/libcameraservice/api1/Camera2Client.h

diff --git a/services/camera/libcameraservice/api1/Camera2Client.h b/services/camera/libcameraservice/api1/Camera2Client.h
index 53629a1..fe0bf74 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.h
+++ b/services/camera/libcameraservice/api1/Camera2Client.h
@@ -136,6 +136,10 @@
     static const int32_t kCaptureRequestIdStart = 30000000;
     static const int32_t kCaptureRequestIdEnd   = 40000000;
 
+    // Constant strings for ATRACE logging
+    static const char* kAutofocusLabel;
+    static const char* kTakepictureLabel;
+
 private:
     /** ICamera interface-related private members */
     typedef camera2::Parameters Parameters;

services/camera/libcameraservice/api1/CameraClient.cpp

diff --git a/services/camera/libcameraservice/api1/CameraClient.cpp b/services/camera/libcameraservice/api1/CameraClient.cpp
index bd6805d..30b7bb8 100644
--- a/services/camera/libcameraservice/api1/CameraClient.cpp
+++ b/services/camera/libcameraservice/api1/CameraClient.cpp
@@ -85,7 +85,7 @@
     mHardware->setCallbacks(notifyCallback,
             dataCallback,
             dataCallbackTimestamp,
-            (void *)mCameraId);
+            (void *)(uintptr_t)mCameraId);
 
     // Enable zoom, error, focus, and metadata messages by default
     enableMsgType(CAMERA_MSG_ERROR | CAMERA_MSG_ZOOM | CAMERA_MSG_FOCUS |

services/camera/libcameraservice/api1/client2/CallbackProcessor.cpp

diff --git a/services/camera/libcameraservice/api1/client2/CallbackProcessor.cpp b/services/camera/libcameraservice/api1/client2/CallbackProcessor.cpp
index 9d8c4a1..d2ac79c 100644
--- a/services/camera/libcameraservice/api1/client2/CallbackProcessor.cpp
+++ b/services/camera/libcameraservice/api1/client2/CallbackProcessor.cpp
@@ -312,6 +312,16 @@
             return OK;
         }
 
+        if (imgBuffer.width != static_cast<uint32_t>(l.mParameters.previewWidth) ||
+                imgBuffer.height != static_cast<uint32_t>(l.mParameters.previewHeight)) {
+            ALOGW("%s: The preview size has changed to %d x %d from %d x %d, this buffer is"
+                    " no longer valid, dropping",__FUNCTION__,
+                    l.mParameters.previewWidth, l.mParameters.previewHeight,
+                    imgBuffer.width, imgBuffer.height);
+            mCallbackConsumer->unlockBuffer(imgBuffer);
+            return OK;
+        }
+
         previewFormat = l.mParameters.previewFormat;
         useFlexibleYuv = l.mParameters.fastInfo.useFlexibleYuv &&
                 (previewFormat == HAL_PIXEL_FORMAT_YCrCb_420_SP ||

services/camera/libcameraservice/api1/client2/CaptureSequencer.cpp

diff --git a/services/camera/libcameraservice/api1/client2/CaptureSequencer.cpp b/services/camera/libcameraservice/api1/client2/CaptureSequencer.cpp
index ca3198f..8a4ce4e 100644
--- a/services/camera/libcameraservice/api1/client2/CaptureSequencer.cpp
+++ b/services/camera/libcameraservice/api1/client2/CaptureSequencer.cpp
@@ -43,6 +43,7 @@
         mShutterNotified(false),
         mClient(client),
         mCaptureState(IDLE),
+        mStateTransitionCount(0),
         mTriggerId(0),
         mTimeoutCount(0),
         mCaptureId(Camera2Client::kCaptureRequestIdStart),
@@ -103,12 +104,12 @@
     }
 }
 
-void CaptureSequencer::onFrameAvailable(int32_t frameId,
+void CaptureSequencer::onFrameAvailable(int32_t requestId,
         const CameraMetadata &frame) {
     ALOGV("%s: Listener found new frame", __FUNCTION__);
     ATRACE_CALL();
     Mutex::Autolock l(mInputMutex);
-    mNewFrameId = frameId;
+    mNewFrameId = requestId;
     mNewFrame = frame;
     if (!mNewFrameReceived) {
         mNewFrameReceived = true;
@@ -198,8 +199,14 @@
 
     Mutex::Autolock l(mStateMutex);
     if (currentState != mCaptureState) {
+        if (mCaptureState != IDLE) {
+            ATRACE_ASYNC_END(kStateNames[mCaptureState], mStateTransitionCount);
+        }
         mCaptureState = currentState;
-        ATRACE_INT("cam2_capt_state", mCaptureState);
+        mStateTransitionCount++;
+        if (mCaptureState != IDLE) {
+            ATRACE_ASYNC_BEGIN(kStateNames[mCaptureState], mStateTransitionCount);
+        }
         ALOGV("Camera %d: New capture state %s",
                 client->getCameraId(), kStateNames[mCaptureState]);
         mStateChanged.signal();
@@ -243,6 +250,7 @@
         mBusy = false;
     }
 
+    int takePictureCounter = 0;
     {
         SharedParameters::Lock l(client->getParameters());
         switch (l.mParameters.state) {
@@ -270,6 +278,7 @@
                         Parameters::getStateName(l.mParameters.state));
                 res = INVALID_OPERATION;
         }
+        takePictureCounter = l.mParameters.takePictureCounter;
     }
     sp<ZslProcessorInterface> processor = mZslProcessor.promote();
     if (processor != 0) {
@@ -282,6 +291,8 @@
      * Fire the jpegCallback in Camera#takePicture(..., jpegCallback)
      */
     if (mCaptureBuffer != 0 && res == OK) {
+        ATRACE_ASYNC_END(Camera2Client::kTakepictureLabel, takePictureCounter);
+
         Camera2Client::SharedCameraCallbacks::Lock
             l(client->mSharedCameraCallbacks);
         ALOGV("%s: Sending still image to client", __FUNCTION__);
@@ -379,11 +390,23 @@
         sp<Camera2Client> &client) {
     ATRACE_CALL();
 
+    bool isAeConverged = false;
     // Get the onFrameAvailable callback when the requestID == mCaptureId
     client->registerFrameListener(mCaptureId, mCaptureId + 1,
             this);
+
+    {
+        Mutex::Autolock l(mInputMutex);
+        isAeConverged = (mAEState == ANDROID_CONTROL_AE_STATE_CONVERGED);
+    }
+
     {
         SharedParameters::Lock l(client->getParameters());
+        // Skip AE precapture when it is already converged and not in force flash mode.
+        if (l.mParameters.flashMode != Parameters::FLASH_MODE_ON && isAeConverged) {
+            return STANDARD_CAPTURE;
+        }
+
         mTriggerId = l.mParameters.precaptureTriggerCounter++;
     }
     client->getCameraDevice()->triggerPrecaptureMetering(mTriggerId);

services/camera/libcameraservice/api1/client2/CaptureSequencer.h

diff --git a/services/camera/libcameraservice/api1/client2/CaptureSequencer.h b/services/camera/libcameraservice/api1/client2/CaptureSequencer.h
index 7ad461a..9fb4ee7 100644
--- a/services/camera/libcameraservice/api1/client2/CaptureSequencer.h
+++ b/services/camera/libcameraservice/api1/client2/CaptureSequencer.h
@@ -62,7 +62,7 @@
     void notifyAutoExposure(uint8_t newState, int triggerId);
 
     // Notifications from the frame processor
-    virtual void onFrameAvailable(int32_t frameId, const CameraMetadata &frame);
+    virtual void onFrameAvailable(int32_t requestId, const CameraMetadata &frame);
 
     // Notifications from the JPEG processor
     void onCaptureAvailable(nsecs_t timestamp, sp<MemoryBase> captureBuffer);
@@ -125,6 +125,7 @@
         NUM_CAPTURE_STATES
     } mCaptureState;
     static const char* kStateNames[];
+    int mStateTransitionCount;
     Mutex mStateMutex; // Guards mCaptureState
     Condition mStateChanged;
 

services/camera/libcameraservice/api1/client2/FrameProcessor.cpp

diff --git a/services/camera/libcameraservice/api1/client2/FrameProcessor.cpp b/services/camera/libcameraservice/api1/client2/FrameProcessor.cpp
index c34cb12..19acae4 100644
--- a/services/camera/libcameraservice/api1/client2/FrameProcessor.cpp
+++ b/services/camera/libcameraservice/api1/client2/FrameProcessor.cpp
@@ -29,13 +29,27 @@
 namespace camera2 {
 
 FrameProcessor::FrameProcessor(wp<CameraDeviceBase> device,
-                               wp<Camera2Client> client) :
+                               sp<Camera2Client> client) :
     FrameProcessorBase(device),
     mClient(client),
-    mLastFrameNumberOfFaces(0) {
+    mLastFrameNumberOfFaces(0),
+    mLast3AFrameNumber(-1) {
 
     sp<CameraDeviceBase> d = device.promote();
     mSynthesize3ANotify = !(d->willNotify3A());
+
+    {
+        SharedParameters::Lock l(client->getParameters());
+        mUsePartialQuirk = l.mParameters.quirks.partialResults;
+
+        // Initialize starting 3A state
+        m3aState.afTriggerId = l.mParameters.afTriggerCounter;
+        m3aState.aeTriggerId = l.mParameters.precaptureTriggerCounter;
+        // Check if lens is fixed-focus
+        if (l.mParameters.focusMode == Parameters::FOCUS_MODE_FIXED) {
+            m3aState.afMode = ANDROID_CONTROL_AF_MODE_OFF;
+        }
+    }
 }
 
 FrameProcessor::~FrameProcessor() {
@@ -49,20 +63,25 @@
         return false;
     }
 
-    if (processFaceDetect(frame, client) != OK) {
+    bool partialResult = false;
+    if (mUsePartialQuirk) {
+        camera_metadata_entry_t entry;
+        entry = frame.find(ANDROID_QUIRKS_PARTIAL_RESULT);
+        if (entry.count > 0 &&
+                entry.data.u8[0] == ANDROID_QUIRKS_PARTIAL_RESULT_PARTIAL) {
+            partialResult = true;
+        }
+    }
+
+    if (!partialResult && processFaceDetect(frame, client) != OK) {
         return false;
     }
 
     if (mSynthesize3ANotify) {
-        // Ignoring missing fields for now
         process3aState(frame, client);
     }
 
-    if (!FrameProcessorBase::processSingleFrame(frame, device)) {
-        return false;
-    }
-
-    return true;
+    return FrameProcessorBase::processSingleFrame(frame, device);
 }
 
 status_t FrameProcessor::processFaceDetect(const CameraMetadata &frame,
@@ -198,86 +217,75 @@
 
     ATRACE_CALL();
     camera_metadata_ro_entry_t entry;
-    int mId = client->getCameraId();
+    int cameraId = client->getCameraId();
 
     entry = frame.find(ANDROID_REQUEST_FRAME_COUNT);
     int32_t frameNumber = entry.data.i32[0];
 
+    // Don't send 3A notifications for the same frame number twice
+    if (frameNumber <= mLast3AFrameNumber) {
+        ALOGV("%s: Already sent 3A for frame number %d, skipping",
+                __FUNCTION__, frameNumber);
+        return OK;
+    }
+
+    mLast3AFrameNumber = frameNumber;
+
     // Get 3A states from result metadata
     bool gotAllStates = true;
 
     AlgState new3aState;
 
-    entry = frame.find(ANDROID_CONTROL_AE_STATE);
-    if (entry.count == 0) {
-        ALOGE("%s: Camera %d: No AE state provided by HAL for frame %d!",
-                __FUNCTION__, mId, frameNumber);
-        gotAllStates = false;
-    } else {
-        new3aState.aeState =
-                static_cast<camera_metadata_enum_android_control_ae_state>(
-                    entry.data.u8[0]);
-    }
+    // TODO: Also use AE mode, AE trigger ID
 
-    entry = frame.find(ANDROID_CONTROL_AF_STATE);
-    if (entry.count == 0) {
-        ALOGE("%s: Camera %d: No AF state provided by HAL for frame %d!",
-                __FUNCTION__, mId, frameNumber);
-        gotAllStates = false;
-    } else {
-        new3aState.afState =
-                static_cast<camera_metadata_enum_android_control_af_state>(
-                    entry.data.u8[0]);
-    }
+    gotAllStates &= get3aResult<uint8_t>(frame, ANDROID_CONTROL_AF_MODE,
+            &new3aState.afMode, frameNumber, cameraId);
 
-    entry = frame.find(ANDROID_CONTROL_AWB_STATE);
-    if (entry.count == 0) {
-        ALOGE("%s: Camera %d: No AWB state provided by HAL for frame %d!",
-                __FUNCTION__, mId, frameNumber);
-        gotAllStates = false;
-    } else {
-        new3aState.awbState =
-                static_cast<camera_metadata_enum_android_control_awb_state>(
-                    entry.data.u8[0]);
-    }
+    gotAllStates &= get3aResult<uint8_t>(frame, ANDROID_CONTROL_AWB_MODE,
+            &new3aState.awbMode, frameNumber, cameraId);
 
-    int32_t afTriggerId = 0;
-    entry = frame.find(ANDROID_CONTROL_AF_TRIGGER_ID);
-    if (entry.count == 0) {
-        ALOGE("%s: Camera %d: No AF trigger ID provided by HAL for frame %d!",
-                __FUNCTION__, mId, frameNumber);
-        gotAllStates = false;
-    } else {
-        afTriggerId = entry.data.i32[0];
-    }
+    gotAllStates &= get3aResult<uint8_t>(frame, ANDROID_CONTROL_AE_STATE,
+            &new3aState.aeState, frameNumber, cameraId);
 
-    int32_t aeTriggerId = 0;
-    entry = frame.find(ANDROID_CONTROL_AE_PRECAPTURE_ID);
-    if (entry.count == 0) {
-        ALOGE("%s: Camera %d: No AE precapture trigger ID provided by HAL"
-                " for frame %d!",
-                __FUNCTION__, mId, frameNumber);
-        gotAllStates = false;
-    } else {
-        aeTriggerId = entry.data.i32[0];
-    }
+    gotAllStates &= get3aResult<uint8_t>(frame, ANDROID_CONTROL_AF_STATE,
+            &new3aState.afState, frameNumber, cameraId);
+
+    gotAllStates &= get3aResult<uint8_t>(frame, ANDROID_CONTROL_AWB_STATE,
+            &new3aState.awbState, frameNumber, cameraId);
+
+    gotAllStates &= get3aResult<int32_t>(frame, ANDROID_CONTROL_AF_TRIGGER_ID,
+            &new3aState.afTriggerId, frameNumber, cameraId);
+
+    gotAllStates &= get3aResult<int32_t>(frame, ANDROID_CONTROL_AE_PRECAPTURE_ID,
+            &new3aState.aeTriggerId, frameNumber, cameraId);
 
     if (!gotAllStates) return BAD_VALUE;
 
     if (new3aState.aeState != m3aState.aeState) {
-        ALOGV("%s: AE state changed from 0x%x to 0x%x",
-                __FUNCTION__, m3aState.aeState, new3aState.aeState);
-        client->notifyAutoExposure(new3aState.aeState, aeTriggerId);
+        ALOGV("%s: Camera %d: AE state %d->%d",
+                __FUNCTION__, cameraId,
+                m3aState.aeState, new3aState.aeState);
+        client->notifyAutoExposure(new3aState.aeState, new3aState.aeTriggerId);
     }
-    if (new3aState.afState != m3aState.afState) {
-        ALOGV("%s: AF state changed from 0x%x to 0x%x",
-                __FUNCTION__, m3aState.afState, new3aState.afState);
-        client->notifyAutoFocus(new3aState.afState, afTriggerId);
+
+    if (new3aState.afState != m3aState.afState ||
+        new3aState.afMode != m3aState.afMode ||
+        new3aState.afTriggerId != m3aState.afTriggerId) {
+        ALOGV("%s: Camera %d: AF state %d->%d. AF mode %d->%d. Trigger %d->%d",
+                __FUNCTION__, cameraId,
+                m3aState.afState, new3aState.afState,
+                m3aState.afMode, new3aState.afMode,
+                m3aState.afTriggerId, new3aState.afTriggerId);
+        client->notifyAutoFocus(new3aState.afState, new3aState.afTriggerId);
     }
-    if (new3aState.awbState != m3aState.awbState) {
-        ALOGV("%s: AWB state changed from 0x%x to 0x%x",
-                __FUNCTION__, m3aState.awbState, new3aState.awbState);
-        client->notifyAutoWhitebalance(new3aState.awbState, aeTriggerId);
+    if (new3aState.awbState != m3aState.awbState ||
+        new3aState.awbMode != m3aState.awbMode) {
+        ALOGV("%s: Camera %d: AWB state %d->%d. AWB mode %d->%d",
+                __FUNCTION__, cameraId,
+                m3aState.awbState, new3aState.awbState,
+                m3aState.awbMode, new3aState.awbMode);
+        client->notifyAutoWhitebalance(new3aState.awbState,
+                new3aState.aeTriggerId);
     }
 
     m3aState = new3aState;
@@ -285,6 +293,39 @@
     return OK;
 }
 
+template<typename Src, typename T>
+bool FrameProcessor::get3aResult(const CameraMetadata& result, int32_t tag,
+        T* value, int32_t frameNumber, int cameraId) {
+    camera_metadata_ro_entry_t entry;
+    if (value == NULL) {
+        ALOGE("%s: Camera %d: Value to write to is NULL",
+                __FUNCTION__, cameraId);
+        return false;
+    }
+
+    entry = result.find(tag);
+    if (entry.count == 0) {
+        ALOGE("%s: Camera %d: No %s provided by HAL for frame %d!",
+                __FUNCTION__, cameraId,
+                get_camera_metadata_tag_name(tag), frameNumber);
+        return false;
+    } else {
+        switch(sizeof(Src)){
+            case sizeof(uint8_t):
+                *value = static_cast<T>(entry.data.u8[0]);
+                break;
+            case sizeof(int32_t):
+                *value = static_cast<T>(entry.data.i32[0]);
+                break;
+            default:
+                ALOGE("%s: Camera %d: Unsupported source",
+                        __FUNCTION__, cameraId);
+                return false;
+        }
+    }
+    return true;
+}
+
 
 void FrameProcessor::callbackFaceDetection(sp<Camera2Client> client,
                                      const camera_frame_metadata &metadata) {

services/camera/libcameraservice/api1/client2/FrameProcessor.h

diff --git a/services/camera/libcameraservice/api1/client2/FrameProcessor.h b/services/camera/libcameraservice/api1/client2/FrameProcessor.h
index 2a17d45..856ad32 100644
--- a/services/camera/libcameraservice/api1/client2/FrameProcessor.h
+++ b/services/camera/libcameraservice/api1/client2/FrameProcessor.h
@@ -39,7 +39,7 @@
  */
 class FrameProcessor : public FrameProcessorBase {
   public:
-    FrameProcessor(wp<CameraDeviceBase> device, wp<Camera2Client> client);
+    FrameProcessor(wp<CameraDeviceBase> device, sp<Camera2Client> client);
     ~FrameProcessor();
 
   private:
@@ -61,18 +61,44 @@
     status_t process3aState(const CameraMetadata &frame,
             const sp<Camera2Client> &client);
 
+    // Helper for process3aState
+    template<typename Src, typename T>
+    bool get3aResult(const CameraMetadata& result, int32_t tag, T* value,
+            int32_t frameNumber, int cameraId);
+
+
     struct AlgState {
+        // TODO: also track AE mode
+        camera_metadata_enum_android_control_af_mode   afMode;
+        camera_metadata_enum_android_control_awb_mode  awbMode;
+
         camera_metadata_enum_android_control_ae_state  aeState;
         camera_metadata_enum_android_control_af_state  afState;
         camera_metadata_enum_android_control_awb_state awbState;
 
+        int32_t                                        afTriggerId;
+        int32_t                                        aeTriggerId;
+
+        // These defaults need to match those in Parameters.cpp
         AlgState() :
+                afMode(ANDROID_CONTROL_AF_MODE_AUTO),
+                awbMode(ANDROID_CONTROL_AWB_MODE_AUTO),
                 aeState(ANDROID_CONTROL_AE_STATE_INACTIVE),
                 afState(ANDROID_CONTROL_AF_STATE_INACTIVE),
-                awbState(ANDROID_CONTROL_AWB_STATE_INACTIVE) {
+                awbState(ANDROID_CONTROL_AWB_STATE_INACTIVE),
+                afTriggerId(0),
+                aeTriggerId(0) {
         }
     } m3aState;
 
+    // Whether the partial result quirk is enabled for this device
+    bool mUsePartialQuirk;
+
+    // Track most recent frame number for which 3A notifications were sent for.
+    // Used to filter against sending 3A notifications for the same frame
+    // several times.
+    int32_t mLast3AFrameNumber;
+
     // Emit FaceDetection event to java if faces changed
     void callbackFaceDetection(sp<Camera2Client> client,
                                const camera_frame_metadata &metadata);

services/camera/libcameraservice/api1/client2/JpegProcessor.cpp

diff --git a/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp b/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp
index 77d5c8a..ec81456 100644
--- a/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp
+++ b/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp
@@ -200,51 +200,60 @@
     ATRACE_CALL();
     status_t res;
     sp<Camera2Heap> captureHeap;
+    sp<MemoryBase> captureBuffer;
 
     CpuConsumer::LockedBuffer imgBuffer;
 
-    res = mCaptureConsumer->lockNextBuffer(&imgBuffer);
-    if (res != OK) {
-        if (res != BAD_VALUE) {
-            ALOGE("%s: Camera %d: Error receiving still image buffer: "
-                    "%s (%d)", __FUNCTION__,
-                    mId, strerror(-res), res);
+    {
+        Mutex::Autolock l(mInputMutex);
+        if (mCaptureStreamId == NO_STREAM) {
+            ALOGW("%s: Camera %d: No stream is available", __FUNCTION__, mId);
+            return INVALID_OPERATION;
         }
-        return res;
-    }
 
-    ALOGV("%s: Camera %d: Still capture available", __FUNCTION__,
-            mId);
+        res = mCaptureConsumer->lockNextBuffer(&imgBuffer);
+        if (res != OK) {
+            if (res != BAD_VALUE) {
+                ALOGE("%s: Camera %d: Error receiving still image buffer: "
+                        "%s (%d)", __FUNCTION__,
+                        mId, strerror(-res), res);
+            }
+            return res;
+        }
 
-    if (imgBuffer.format != HAL_PIXEL_FORMAT_BLOB) {
-        ALOGE("%s: Camera %d: Unexpected format for still image: "
-                "%x, expected %x", __FUNCTION__, mId,
-                imgBuffer.format,
-                HAL_PIXEL_FORMAT_BLOB);
+        ALOGV("%s: Camera %d: Still capture available", __FUNCTION__,
+                mId);
+
+        if (imgBuffer.format != HAL_PIXEL_FORMAT_BLOB) {
+            ALOGE("%s: Camera %d: Unexpected format for still image: "
+                    "%x, expected %x", __FUNCTION__, mId,
+                    imgBuffer.format,
+                    HAL_PIXEL_FORMAT_BLOB);
+            mCaptureConsumer->unlockBuffer(imgBuffer);
+            return OK;
+        }
+
+        // Find size of JPEG image
+        size_t jpegSize = findJpegSize(imgBuffer.data, imgBuffer.width);
+        if (jpegSize == 0) { // failed to find size, default to whole buffer
+            jpegSize = imgBuffer.width;
+        }
+        size_t heapSize = mCaptureHeap->getSize();
+        if (jpegSize > heapSize) {
+            ALOGW("%s: JPEG image is larger than expected, truncating "
+                    "(got %d, expected at most %d bytes)",
+                    __FUNCTION__, jpegSize, heapSize);
+            jpegSize = heapSize;
+        }
+
+        // TODO: Optimize this to avoid memcopy
+        captureBuffer = new MemoryBase(mCaptureHeap, 0, jpegSize);
+        void* captureMemory = mCaptureHeap->getBase();
+        memcpy(captureMemory, imgBuffer.data, jpegSize);
+
         mCaptureConsumer->unlockBuffer(imgBuffer);
-        return OK;
     }
 
-    // Find size of JPEG image
-    size_t jpegSize = findJpegSize(imgBuffer.data, imgBuffer.width);
-    if (jpegSize == 0) { // failed to find size, default to whole buffer
-        jpegSize = imgBuffer.width;
-    }
-    size_t heapSize = mCaptureHeap->getSize();
-    if (jpegSize > heapSize) {
-        ALOGW("%s: JPEG image is larger than expected, truncating "
-                "(got %d, expected at most %d bytes)",
-                __FUNCTION__, jpegSize, heapSize);
-        jpegSize = heapSize;
-    }
-
-    // TODO: Optimize this to avoid memcopy
-    sp<MemoryBase> captureBuffer = new MemoryBase(mCaptureHeap, 0, jpegSize);
-    void* captureMemory = mCaptureHeap->getBase();
-    memcpy(captureMemory, imgBuffer.data, jpegSize);
-
-    mCaptureConsumer->unlockBuffer(imgBuffer);
-
     sp<CaptureSequencer> sequencer = mSequencer.promote();
     if (sequencer != 0) {
         sequencer->onCaptureAvailable(imgBuffer.timestamp, captureBuffer);

services/camera/libcameraservice/api1/client2/Parameters.cpp

diff --git a/services/camera/libcameraservice/api1/client2/Parameters.cpp b/services/camera/libcameraservice/api1/client2/Parameters.cpp
index a6c1083..2cf0d29 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.cpp
+++ b/services/camera/libcameraservice/api1/client2/Parameters.cpp
@@ -183,8 +183,7 @@
     // still have to do something sane for them
 
     // NOTE: Not scaled like FPS range values are.
-    previewFps = fpsFromRange(previewFpsRange[0], previewFpsRange[1]);
-    lastSetPreviewFps = previewFps;
+    int previewFps = fpsFromRange(previewFpsRange[0], previewFpsRange[1]);
     params.set(CameraParameters::KEY_PREVIEW_FRAME_RATE,
             previewFps);
 
@@ -250,9 +249,17 @@
         staticInfo(ANDROID_JPEG_AVAILABLE_THUMBNAIL_SIZES, 4);
     if (!availableJpegThumbnailSizes.count) return NO_INIT;
 
-    // TODO: Pick default thumbnail size sensibly
-    jpegThumbSize[0] = availableJpegThumbnailSizes.data.i32[0];
-    jpegThumbSize[1] = availableJpegThumbnailSizes.data.i32[1];
+    // Pick the largest thumbnail size that matches still image aspect ratio.
+    ALOG_ASSERT(pictureWidth > 0 && pictureHeight > 0,
+            "Invalid picture size, %d x %d", pictureWidth, pictureHeight);
+    float picAspectRatio = static_cast<float>(pictureWidth) / pictureHeight;
+    Size thumbnailSize =
+            getMaxSizeForRatio(
+                    picAspectRatio,
+                    &availableJpegThumbnailSizes.data.i32[0],
+                    availableJpegThumbnailSizes.count);
+    jpegThumbSize[0] = thumbnailSize.width;
+    jpegThumbSize[1] = thumbnailSize.height;
 
     params.set(CameraParameters::KEY_JPEG_THUMBNAIL_WIDTH,
             jpegThumbSize[0]);
@@ -459,7 +466,7 @@
                 supportedAntibanding);
     }
 
-    sceneMode = ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED;
+    sceneMode = ANDROID_CONTROL_SCENE_MODE_DISABLED;
     params.set(CameraParameters::KEY_SCENE_MODE,
             CameraParameters::SCENE_MODE_AUTO);
 
@@ -475,7 +482,7 @@
             if (addComma) supportedSceneModes += ",";
             addComma = true;
             switch (availableSceneModes.data.u8[i]) {
-                case ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED:
+                case ANDROID_CONTROL_SCENE_MODE_DISABLED:
                     noSceneModes = true;
                     break;
                 case ANDROID_CONTROL_SCENE_MODE_FACE_PRIORITY:
@@ -657,13 +664,13 @@
     focusState = ANDROID_CONTROL_AF_STATE_INACTIVE;
     shadowFocusMode = FOCUS_MODE_INVALID;
 
-    camera_metadata_ro_entry_t max3aRegions =
-        staticInfo(ANDROID_CONTROL_MAX_REGIONS, 1, 1);
-    if (!max3aRegions.count) return NO_INIT;
+    camera_metadata_ro_entry_t max3aRegions = staticInfo(ANDROID_CONTROL_MAX_REGIONS,
+            Parameters::NUM_REGION, Parameters::NUM_REGION);
+    if (max3aRegions.count != Parameters::NUM_REGION) return NO_INIT;
 
     int32_t maxNumFocusAreas = 0;
     if (focusMode != Parameters::FOCUS_MODE_FIXED) {
-        maxNumFocusAreas = max3aRegions.data.i32[0];
+        maxNumFocusAreas = max3aRegions.data.i32[Parameters::REGION_AF];
     }
     params.set(CameraParameters::KEY_MAX_NUM_FOCUS_AREAS, maxNumFocusAreas);
     params.set(CameraParameters::KEY_FOCUS_AREAS,
@@ -723,7 +730,7 @@
 
     meteringAreas.add(Parameters::Area(0, 0, 0, 0, 0));
     params.set(CameraParameters::KEY_MAX_NUM_METERING_AREAS,
-            max3aRegions.data.i32[0]);
+            max3aRegions.data.i32[Parameters::REGION_AE]);
     params.set(CameraParameters::KEY_METERING_AREAS,
             "(0,0,0,0,0)");
 
@@ -797,30 +804,25 @@
 
     enableFocusMoveMessages = false;
     afTriggerCounter = 1;
+    afStateCounter = 0;
     currentAfTriggerId = -1;
     afInMotion = false;
 
     precaptureTriggerCounter = 1;
 
+    takePictureCounter = 0;
+
     previewCallbackFlags = 0;
     previewCallbackOneShot = false;
     previewCallbackSurface = false;
 
-    camera_metadata_ro_entry_t supportedHardwareLevel =
-        staticInfo(ANDROID_INFO_SUPPORTED_HARDWARE_LEVEL, 0, 0, false);
-    if (!supportedHardwareLevel.count || (supportedHardwareLevel.data.u8[0] ==
-            ANDROID_INFO_SUPPORTED_HARDWARE_LEVEL_LIMITED)) {
-        ALOGI("Camera %d: ZSL mode disabled for limited mode HALs", cameraId);
+    char value[PROPERTY_VALUE_MAX];
+    property_get("camera.disable_zsl_mode", value, "0");
+    if (!strcmp(value,"1")) {
+        ALOGI("Camera %d: Disabling ZSL mode", cameraId);
         zslMode = false;
     } else {
-        char value[PROPERTY_VALUE_MAX];
-        property_get("camera.disable_zsl_mode", value, "0");
-        if (!strcmp(value,"1")) {
-            ALOGI("Camera %d: Disabling ZSL mode", cameraId);
-            zslMode = false;
-        } else {
-            zslMode = true;
-        }
+        zslMode = true;
     }
 
     lightFx = LIGHTFX_NONE;
@@ -1044,6 +1046,11 @@
     ALOGV_IF(quirks.meteringCropRegion, "Camera %d: Quirk meteringCropRegion"
                 " enabled", cameraId);
 
+    entry = info->find(ANDROID_QUIRKS_USE_PARTIAL_RESULT);
+    quirks.partialResults = (entry.count != 0 && entry.data.u8[0] == 1);
+    ALOGV_IF(quirks.partialResults, "Camera %d: Quirk usePartialResult"
+                " enabled", cameraId);
+
     return OK;
 }
 
@@ -1126,13 +1133,22 @@
 
     // PREVIEW_FPS_RANGE
     bool fpsRangeChanged = false;
+    int32_t lastSetFpsRange[2];
+
+    params.getPreviewFpsRange(&lastSetFpsRange[0], &lastSetFpsRange[1]);
+    lastSetFpsRange[0] /= kFpsToApiScale;
+    lastSetFpsRange[1] /= kFpsToApiScale;
+
     newParams.getPreviewFpsRange(&validatedParams.previewFpsRange[0],
             &validatedParams.previewFpsRange[1]);
     validatedParams.previewFpsRange[0] /= kFpsToApiScale;
     validatedParams.previewFpsRange[1] /= kFpsToApiScale;
 
-    if (validatedParams.previewFpsRange[0] != previewFpsRange[0] ||
-            validatedParams.previewFpsRange[1] != previewFpsRange[1]) {
+    // Compare the FPS range value from the last set() to the current set()
+    // to determine if the client has changed it
+    if (validatedParams.previewFpsRange[0] != lastSetFpsRange[0] ||
+            validatedParams.previewFpsRange[1] != lastSetFpsRange[1]) {
+
         fpsRangeChanged = true;
         camera_metadata_ro_entry_t availablePreviewFpsRanges =
             staticInfo(ANDROID_CONTROL_AE_AVAILABLE_TARGET_FPS_RANGES, 2);
@@ -1150,16 +1166,6 @@
                     validatedParams.previewFpsRange[1]);
             return BAD_VALUE;
         }
-        validatedParams.previewFps =
-            fpsFromRange(validatedParams.previewFpsRange[0],
-                         validatedParams.previewFpsRange[1]);
-
-        // Update our last-seen single preview FPS, needed for disambiguating
-        // when the application is intending to use the deprecated single-FPS
-        // setting vs. the range FPS setting
-        validatedParams.lastSetPreviewFps = newParams.getPreviewFrameRate();
-
-        newParams.setPreviewFrameRate(validatedParams.previewFps);
     }
 
     // PREVIEW_FORMAT
@@ -1197,12 +1203,11 @@
     // PREVIEW_FRAME_RATE Deprecated, only use if the preview fps range is
     // unchanged this time.  The single-value FPS is the same as the minimum of
     // the range.  To detect whether the application has changed the value of
-    // previewFps, compare against their last-set preview FPS instead of the
-    // single FPS we may have synthesized from a range FPS set.
+    // previewFps, compare against their last-set preview FPS.
     if (!fpsRangeChanged) {
-        validatedParams.previewFps = newParams.getPreviewFrameRate();
-        if (validatedParams.previewFps != lastSetPreviewFps ||
-                recordingHintChanged) {
+        int previewFps = newParams.getPreviewFrameRate();
+        int lastSetPreviewFps = params.getPreviewFrameRate();
+        if (previewFps != lastSetPreviewFps || recordingHintChanged) {
             camera_metadata_ro_entry_t availableFrameRates =
                 staticInfo(ANDROID_CONTROL_AE_AVAILABLE_TARGET_FPS_RANGES);
             /**
@@ -1215,8 +1220,8 @@
               * Either way, in case of multiple ranges, break the tie by
               * selecting the smaller range.
               */
-            int targetFps = validatedParams.previewFps;
-            // all ranges which have targetFps
+
+            // all ranges which have previewFps
             Vector<Range> candidateRanges;
             for (i = 0; i < availableFrameRates.count; i+=2) {
                 Range r = {
@@ -1224,13 +1229,13 @@
                             availableFrameRates.data.i32[i+1]
                 };
 
-                if (r.min <= targetFps && targetFps <= r.max) {
+                if (r.min <= previewFps && previewFps <= r.max) {
                     candidateRanges.push(r);
                 }
             }
             if (candidateRanges.isEmpty()) {
                 ALOGE("%s: Requested preview frame rate %d is not supported",
-                        __FUNCTION__, validatedParams.previewFps);
+                        __FUNCTION__, previewFps);
                 return BAD_VALUE;
             }
             // most applicable range with targetFps
@@ -1269,14 +1274,6 @@
                 validatedParams.previewFpsRange[1],
                 validatedParams.recordingHint);
         }
-        newParams.set(CameraParameters::KEY_PREVIEW_FPS_RANGE,
-                String8::format("%d,%d",
-                        validatedParams.previewFpsRange[0] * kFpsToApiScale,
-                        validatedParams.previewFpsRange[1] * kFpsToApiScale));
-        // Update our last-seen single preview FPS, needed for disambiguating
-        // when the application is intending to use the deprecated single-FPS
-        // setting vs. the range FPS setting
-        validatedParams.lastSetPreviewFps = validatedParams.previewFps;
     }
 
     // PICTURE_SIZE
@@ -1449,7 +1446,7 @@
         newParams.get(CameraParameters::KEY_SCENE_MODE) );
     if (validatedParams.sceneMode != sceneMode &&
             validatedParams.sceneMode !=
-            ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED) {
+            ANDROID_CONTROL_SCENE_MODE_DISABLED) {
         camera_metadata_ro_entry_t availableSceneModes =
             staticInfo(ANDROID_CONTROL_AVAILABLE_SCENE_MODES);
         for (i = 0; i < availableSceneModes.count; i++) {
@@ -1464,7 +1461,7 @@
         }
     }
     bool sceneModeSet =
-            validatedParams.sceneMode != ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED;
+            validatedParams.sceneMode != ANDROID_CONTROL_SCENE_MODE_DISABLED;
 
     // FLASH_MODE
     if (sceneModeSet) {
@@ -1594,10 +1591,11 @@
     // FOCUS_AREAS
     res = parseAreas(newParams.get(CameraParameters::KEY_FOCUS_AREAS),
             &validatedParams.focusingAreas);
-    size_t max3aRegions =
-        (size_t)staticInfo(ANDROID_CONTROL_MAX_REGIONS, 1, 1).data.i32[0];
+    size_t maxAfRegions = (size_t)staticInfo(ANDROID_CONTROL_MAX_REGIONS,
+              Parameters::NUM_REGION, Parameters::NUM_REGION).
+              data.i32[Parameters::REGION_AF];
     if (res == OK) res = validateAreas(validatedParams.focusingAreas,
-            max3aRegions, AREA_KIND_FOCUS);
+            maxAfRegions, AREA_KIND_FOCUS);
     if (res != OK) {
         ALOGE("%s: Requested focus areas are malformed: %s",
                 __FUNCTION__, newParams.get(CameraParameters::KEY_FOCUS_AREAS));
@@ -1627,10 +1625,13 @@
         newParams.get(CameraParameters::KEY_AUTO_WHITEBALANCE_LOCK));
 
     // METERING_AREAS
+    size_t maxAeRegions = (size_t)staticInfo(ANDROID_CONTROL_MAX_REGIONS,
+            Parameters::NUM_REGION, Parameters::NUM_REGION).
+            data.i32[Parameters::REGION_AE];
     res = parseAreas(newParams.get(CameraParameters::KEY_METERING_AREAS),
             &validatedParams.meteringAreas);
     if (res == OK) {
-        res = validateAreas(validatedParams.meteringAreas, max3aRegions,
+        res = validateAreas(validatedParams.meteringAreas, maxAeRegions,
                             AREA_KIND_METERING);
     }
     if (res != OK) {
@@ -1779,7 +1780,7 @@
     // (face detection statistics and face priority scene mode). Map from other
     // to the other.
     bool sceneModeActive =
-            sceneMode != (uint8_t)ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED;
+            sceneMode != (uint8_t)ANDROID_CONTROL_SCENE_MODE_DISABLED;
     uint8_t reqControlMode = ANDROID_CONTROL_MODE_AUTO;
     if (enableFaceDetect || sceneModeActive) {
         reqControlMode = ANDROID_CONTROL_MODE_USE_SCENE_MODE;
@@ -1791,7 +1792,7 @@
     uint8_t reqSceneMode =
             sceneModeActive ? sceneMode :
             enableFaceDetect ? (uint8_t)ANDROID_CONTROL_SCENE_MODE_FACE_PRIORITY :
-            (uint8_t)ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED;
+            (uint8_t)ANDROID_CONTROL_SCENE_MODE_DISABLED;
     res = request->update(ANDROID_CONTROL_SCENE_MODE,
             &reqSceneMode, 1);
     if (res != OK) return res;
@@ -1861,23 +1862,23 @@
 
     size_t reqFocusingAreasSize = focusingAreas.size() * 5;
     int32_t *reqFocusingAreas = new int32_t[reqFocusingAreasSize];
-    for (size_t i = 0; i < reqFocusingAreasSize; i += 5) {
-        if (focusingAreas[i].weight != 0) {
+    for (size_t i = 0, j = 0; i < reqFocusingAreasSize; i += 5, j++) {
+        if (focusingAreas[j].weight != 0) {
             reqFocusingAreas[i + 0] =
-                    normalizedXToArray(focusingAreas[i].left);
+                    normalizedXToArray(focusingAreas[j].left);
             reqFocusingAreas[i + 1] =
-                    normalizedYToArray(focusingAreas[i].top);
+                    normalizedYToArray(focusingAreas[j].top);
             reqFocusingAreas[i + 2] =
-                    normalizedXToArray(focusingAreas[i].right);
+                    normalizedXToArray(focusingAreas[j].right);
             reqFocusingAreas[i + 3] =
-                    normalizedYToArray(focusingAreas[i].bottom);
+                    normalizedYToArray(focusingAreas[j].bottom);
         } else {
             reqFocusingAreas[i + 0] = 0;
             reqFocusingAreas[i + 1] = 0;
             reqFocusingAreas[i + 2] = 0;
             reqFocusingAreas[i + 3] = 0;
         }
-        reqFocusingAreas[i + 4] = focusingAreas[i].weight;
+        reqFocusingAreas[i + 4] = focusingAreas[j].weight;
     }
     res = request->update(ANDROID_CONTROL_AF_REGIONS,
             reqFocusingAreas, reqFocusingAreasSize);
@@ -1890,28 +1891,45 @@
 
     size_t reqMeteringAreasSize = meteringAreas.size() * 5;
     int32_t *reqMeteringAreas = new int32_t[reqMeteringAreasSize];
-    for (size_t i = 0; i < reqMeteringAreasSize; i += 5) {
-        if (meteringAreas[i].weight != 0) {
+    for (size_t i = 0, j = 0; i < reqMeteringAreasSize; i += 5, j++) {
+        if (meteringAreas[j].weight != 0) {
             reqMeteringAreas[i + 0] =
-                normalizedXToArray(meteringAreas[i].left);
+                normalizedXToArray(meteringAreas[j].left);
             reqMeteringAreas[i + 1] =
-                normalizedYToArray(meteringAreas[i].top);
+                normalizedYToArray(meteringAreas[j].top);
             reqMeteringAreas[i + 2] =
-                normalizedXToArray(meteringAreas[i].right);
+                normalizedXToArray(meteringAreas[j].right);
             reqMeteringAreas[i + 3] =
-                normalizedYToArray(meteringAreas[i].bottom);
+                normalizedYToArray(meteringAreas[j].bottom);
         } else {
             reqMeteringAreas[i + 0] = 0;
             reqMeteringAreas[i + 1] = 0;
             reqMeteringAreas[i + 2] = 0;
             reqMeteringAreas[i + 3] = 0;
         }
-        reqMeteringAreas[i + 4] = meteringAreas[i].weight;
+        reqMeteringAreas[i + 4] = meteringAreas[j].weight;
     }
     res = request->update(ANDROID_CONTROL_AE_REGIONS,
             reqMeteringAreas, reqMeteringAreasSize);
     if (res != OK) return res;
 
+    // Set awb regions to be the same as the metering regions if allowed
+    size_t maxAwbRegions = (size_t)staticInfo(ANDROID_CONTROL_MAX_REGIONS,
+            Parameters::NUM_REGION, Parameters::NUM_REGION).
+            data.i32[Parameters::REGION_AWB];
+    if (maxAwbRegions > 0) {
+        if (maxAwbRegions >= meteringAreas.size()) {
+            res = request->update(ANDROID_CONTROL_AWB_REGIONS,
+                    reqMeteringAreas, reqMeteringAreasSize);
+        } else {
+            // Ensure the awb regions are zeroed if the region count is too high.
+            int32_t zeroedAwbAreas[5] = {0, 0, 0, 0, 0};
+            res = request->update(ANDROID_CONTROL_AWB_REGIONS,
+                    zeroedAwbAreas, sizeof(zeroedAwbAreas)/sizeof(int32_t));
+        }
+        if (res != OK) return res;
+    }
+
     delete[] reqMeteringAreas;
 
     /* don't include jpeg thumbnail size - it's valid for
@@ -2152,9 +2170,9 @@
 int Parameters::sceneModeStringToEnum(const char *sceneMode) {
     return
         !sceneMode ?
-            ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED :
+            ANDROID_CONTROL_SCENE_MODE_DISABLED :
         !strcmp(sceneMode, CameraParameters::SCENE_MODE_AUTO) ?
-            ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED :
+            ANDROID_CONTROL_SCENE_MODE_DISABLED :
         !strcmp(sceneMode, CameraParameters::SCENE_MODE_ACTION) ?
             ANDROID_CONTROL_SCENE_MODE_ACTION :
         !strcmp(sceneMode, CameraParameters::SCENE_MODE_PORTRAIT) ?
@@ -2525,6 +2543,32 @@
     return OK;
 }
 
+Parameters::Size Parameters::getMaxSizeForRatio(
+        float ratio, const int32_t* sizeArray, size_t count) {
+    ALOG_ASSERT(sizeArray != NULL, "size array shouldn't be NULL");
+    ALOG_ASSERT(count >= 2 && count % 2 == 0, "count must be a positive even number");
+
+    Size maxSize = {0, 0};
+    for (size_t i = 0; i < count; i += 2) {
+        if (sizeArray[i] > 0 && sizeArray[i+1] > 0) {
+            float curRatio = static_cast<float>(sizeArray[i]) / sizeArray[i+1];
+            if (fabs(curRatio - ratio) < ASPECT_RATIO_TOLERANCE && maxSize.width < sizeArray[i]) {
+                maxSize.width = sizeArray[i];
+                maxSize.height = sizeArray[i+1];
+            }
+        }
+    }
+
+    if (maxSize.width == 0 || maxSize.height == 0) {
+        maxSize.width = sizeArray[0];
+        maxSize.height = sizeArray[1];
+        ALOGW("Unable to find the size to match the given aspect ratio %f."
+                "Fall back to %d x %d", ratio, maxSize.width, maxSize.height);
+    }
+
+    return maxSize;
+}
+
 Parameters::CropRegion Parameters::calculateCropRegion(
                             Parameters::CropRegion::Outputs outputs) const {
 

services/camera/libcameraservice/api1/client2/Parameters.h

diff --git a/services/camera/libcameraservice/api1/client2/Parameters.h b/services/camera/libcameraservice/api1/client2/Parameters.h
index 0505b0e..60c4687 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.h
+++ b/services/camera/libcameraservice/api1/client2/Parameters.h
@@ -46,8 +46,6 @@
 
     int previewWidth, previewHeight;
     int32_t previewFpsRange[2];
-    int lastSetPreviewFps; // the last single FPS value seen in a set call
-    int previewFps; // deprecated, here only for tracking changes
     int previewFormat;
 
     int previewTransform; // set by CAMERA_CMD_SET_DISPLAY_ORIENTATION
@@ -115,6 +113,14 @@
     bool autoExposureLock;
     bool autoWhiteBalanceLock;
 
+    // 3A region types, for use with ANDROID_CONTROL_MAX_REGIONS
+    enum region_t {
+        REGION_AE = 0,
+        REGION_AWB,
+        REGION_AF,
+        NUM_REGION // Number of region types
+    } region;
+
     Vector<Area> meteringAreas;
 
     int zoom;
@@ -141,11 +147,14 @@
 
     bool enableFocusMoveMessages;
     int afTriggerCounter;
+    int afStateCounter;
     int currentAfTriggerId;
     bool afInMotion;
 
     int precaptureTriggerCounter;
 
+    int takePictureCounter;
+
     uint32_t previewCallbackFlags;
     bool previewCallbackOneShot;
     bool previewCallbackSurface;
@@ -168,6 +177,8 @@
     // Max preview size allowed
     static const unsigned int MAX_PREVIEW_WIDTH = 1920;
     static const unsigned int MAX_PREVIEW_HEIGHT = 1080;
+    // Aspect ratio tolerance
+    static const float ASPECT_RATIO_TOLERANCE = 0.001;
 
     // Full static camera info, object owned by someone else, such as
     // Camera2Device.
@@ -204,6 +215,7 @@
         bool triggerAfWithAuto;
         bool useZslFormat;
         bool meteringCropRegion;
+        bool partialResults;
     } quirks;
 
     /**
@@ -331,6 +343,8 @@
     Vector<Size> availablePreviewSizes;
     // Get size list (that are no larger than limit) from static metadata.
     status_t getFilteredPreviewSizes(Size limit, Vector<Size> *sizes);
+    // Get max size (from the size array) that matches the given aspect ratio.
+    Size getMaxSizeForRatio(float ratio, const int32_t* sizeArray, size_t count);
 };
 
 // This class encapsulates the Parameters class so that it can only be accessed

services/camera/libcameraservice/api1/client2/ZslProcessor.cpp

diff --git a/services/camera/libcameraservice/api1/client2/ZslProcessor.cpp b/services/camera/libcameraservice/api1/client2/ZslProcessor.cpp
index 08ab357..453d54c 100644
--- a/services/camera/libcameraservice/api1/client2/ZslProcessor.cpp
+++ b/services/camera/libcameraservice/api1/client2/ZslProcessor.cpp
@@ -71,7 +71,7 @@
     }
 }
 
-void ZslProcessor::onFrameAvailable(int32_t /*frameId*/,
+void ZslProcessor::onFrameAvailable(int32_t /*requestId*/,
         const CameraMetadata &frame) {
     Mutex::Autolock l(mInputMutex);
     camera_metadata_ro_entry_t entry;
@@ -540,7 +540,7 @@
             if (entry.count > 0) frameAeState = entry.data.u8[0];
         }
         String8 result =
-                String8::format("   %d: b: %lld\tf: %lld, AE state: %d", i,
+                String8::format("   %zu: b: %lld\tf: %lld, AE state: %d", i,
                         bufferTimestamp, frameTimestamp, frameAeState);
         ALOGV("%s", result.string());
         if (fd != -1) {

services/camera/libcameraservice/api1/client2/ZslProcessor.h

diff --git a/services/camera/libcameraservice/api1/client2/ZslProcessor.h b/services/camera/libcameraservice/api1/client2/ZslProcessor.h
index 5fb178f..6d3cb85 100644
--- a/services/camera/libcameraservice/api1/client2/ZslProcessor.h
+++ b/services/camera/libcameraservice/api1/client2/ZslProcessor.h
@@ -54,7 +54,7 @@
     // From mZslConsumer
     virtual void onFrameAvailable();
     // From FrameProcessor
-    virtual void onFrameAvailable(int32_t frameId, const CameraMetadata &frame);
+    virtual void onFrameAvailable(int32_t requestId, const CameraMetadata &frame);
 
     virtual void onBufferReleased(buffer_handle_t *handle);
 

services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp

diff --git a/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp b/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp
index 3e05091..c1d0496 100644
--- a/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp
+++ b/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp
@@ -61,7 +61,7 @@
     deleteStream();
 }
 
-void ZslProcessor3::onFrameAvailable(int32_t /*frameId*/,
+void ZslProcessor3::onFrameAvailable(int32_t /*requestId*/,
                                      const CameraMetadata &frame) {
     Mutex::Autolock l(mInputMutex);
     camera_metadata_ro_entry_t entry;
@@ -273,6 +273,15 @@
             return INVALID_OPERATION;
         }
 
+        // Flush device to clear out all in-flight requests pending in HAL.
+        res = client->getCameraDevice()->flush();
+        if (res != OK) {
+            ALOGE("%s: Camera %d: Failed to flush device: "
+                "%s (%d)",
+                __FUNCTION__, client->getCameraId(), strerror(-res), res);
+            return res;
+        }
+
         // Update JPEG settings
         {
             SharedParameters::Lock l(client->getParameters());
@@ -355,7 +364,7 @@
             if (entry.count > 0) frameAeState = entry.data.u8[0];
         }
         String8 result =
-                String8::format("   %d: b: %lld\tf: %lld, AE state: %d", i,
+                String8::format("   %zu: b: %lld\tf: %lld, AE state: %d", i,
                         bufferTimestamp, frameTimestamp, frameAeState);
         ALOGV("%s", result.string());
         if (fd != -1) {

services/camera/libcameraservice/api1/client2/ZslProcessor3.h

diff --git a/services/camera/libcameraservice/api1/client2/ZslProcessor3.h b/services/camera/libcameraservice/api1/client2/ZslProcessor3.h
index 35b85f5..d2f8322 100644
--- a/services/camera/libcameraservice/api1/client2/ZslProcessor3.h
+++ b/services/camera/libcameraservice/api1/client2/ZslProcessor3.h
@@ -51,7 +51,7 @@
     ~ZslProcessor3();
 
     // From FrameProcessor
-    virtual void onFrameAvailable(int32_t frameId, const CameraMetadata &frame);
+    virtual void onFrameAvailable(int32_t requestId, const CameraMetadata &frame);
 
     /**
      ****************************************

services/camera/libcameraservice/api2/CameraDeviceClient.cpp

diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
index 83466cb..187220e 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
@@ -45,14 +45,6 @@
                 cameraId, cameraFacing, clientPid, clientUid, servicePid),
     mRemoteCallback(remoteCallback) {
 }
-void CameraDeviceClientBase::notifyError() {
-    // Thread safe. Don't bother locking.
-    sp<ICameraDeviceCallbacks> remoteCb = mRemoteCallback;
-
-    if (remoteCb != 0) {
-        remoteCb->notifyCallback(CAMERA_MSG_ERROR, CAMERA_ERROR_RELEASED, 0);
-    }
-}
 
 // Interface used by CameraService
 
@@ -89,7 +81,8 @@
 
     mFrameProcessor->registerListener(FRAME_PROCESSOR_LISTENER_MIN_ID,
                                       FRAME_PROCESSOR_LISTENER_MAX_ID,
-                                      /*listener*/this);
+                                      /*listener*/this,
+                                      /*quirkSendPartials*/true);
 
     return OK;
 }
@@ -164,7 +157,6 @@
     metadata.update(ANDROID_REQUEST_OUTPUT_STREAMS, &outputStreamIds[0],
                     outputStreamIds.size());
 
-    // TODO: @hide ANDROID_REQUEST_ID, or use another request token
     int32_t requestId = mRequestIdCounter++;
     metadata.update(ANDROID_REQUEST_ID, &requestId, /*size*/1);
     ALOGV("%s: Camera %d: Submitting request with ID %d",
@@ -298,11 +290,28 @@
         }
     }
 
+    // HACK b/10949105
+    // Query consumer usage bits to set async operation mode for
+    // GLConsumer using controlledByApp parameter.
+    bool useAsync = false;
+    int32_t consumerUsage;
+    if ((res = bufferProducer->query(NATIVE_WINDOW_CONSUMER_USAGE_BITS,
+            &consumerUsage)) != OK) {
+        ALOGE("%s: Camera %d: Failed to query consumer usage", __FUNCTION__,
+              mCameraId);
+        return res;
+    }
+    if (consumerUsage & GraphicBuffer::USAGE_HW_TEXTURE) {
+        ALOGW("%s: Camera %d: Forcing asynchronous mode for stream",
+                __FUNCTION__, mCameraId);
+        useAsync = true;
+    }
+
     sp<IBinder> binder;
     sp<ANativeWindow> anw;
     if (bufferProducer != 0) {
         binder = bufferProducer->asBinder();
-        anw = new Surface(bufferProducer);
+        anw = new Surface(bufferProducer, useAsync);
     }
 
     // TODO: remove w,h,f since we are ignoring them
@@ -484,6 +493,34 @@
     return dumpDevice(fd, args);
 }
 
+
+void CameraDeviceClient::notifyError() {
+    // Thread safe. Don't bother locking.
+    sp<ICameraDeviceCallbacks> remoteCb = getRemoteCallback();
+
+    if (remoteCb != 0) {
+        remoteCb->onDeviceError(ICameraDeviceCallbacks::ERROR_CAMERA_DEVICE);
+    }
+}
+
+void CameraDeviceClient::notifyIdle() {
+    // Thread safe. Don't bother locking.
+    sp<ICameraDeviceCallbacks> remoteCb = getRemoteCallback();
+
+    if (remoteCb != 0) {
+        remoteCb->onDeviceIdle();
+    }
+}
+
+void CameraDeviceClient::notifyShutter(int requestId,
+        nsecs_t timestamp) {
+    // Thread safe. Don't bother locking.
+    sp<ICameraDeviceCallbacks> remoteCb = getRemoteCallback();
+    if (remoteCb != 0) {
+        remoteCb->onCaptureStarted(requestId, timestamp);
+    }
+}
+
 // TODO: refactor the code below this with IProCameraUser.
 // it's 100% copy-pasted, so lets not change it right now to make it easier.
 
@@ -515,8 +552,8 @@
 }
 
 /** Device-related methods */
-void CameraDeviceClient::onFrameAvailable(int32_t frameId,
-                                        const CameraMetadata& frame) {
+void CameraDeviceClient::onFrameAvailable(int32_t requestId,
+        const CameraMetadata& frame) {
     ATRACE_CALL();
     ALOGV("%s", __FUNCTION__);
 
@@ -524,7 +561,7 @@
     sp<ICameraDeviceCallbacks> remoteCb = mRemoteCallback;
     if (remoteCb != NULL) {
         ALOGV("%s: frame = %p ", __FUNCTION__, &frame);
-        remoteCb->onResultReceived(frameId, frame);
+        remoteCb->onResultReceived(requestId, frame);
     }
 }
 
@@ -598,26 +635,56 @@
         return INVALID_OPERATION;
     }
 
+    camera_metadata_ro_entry_t entryFacing = staticInfo.find(ANDROID_LENS_FACING);
+    if (entry.count == 0) {
+        ALOGE("%s: Camera %d: Can't find android.lens.facing in "
+                "static metadata!", __FUNCTION__, mCameraId);
+        return INVALID_OPERATION;
+    }
+
     int32_t& flags = *transform;
 
+    bool mirror = (entryFacing.data.u8[0] == ANDROID_LENS_FACING_FRONT);
     int orientation = entry.data.i32[0];
-    switch (orientation) {
-        case 0:
-            flags = 0;
-            break;
-        case 90:
-            flags = NATIVE_WINDOW_TRANSFORM_ROT_90;
-            break;
-        case 180:
-            flags = NATIVE_WINDOW_TRANSFORM_ROT_180;
-            break;
-        case 270:
-            flags = NATIVE_WINDOW_TRANSFORM_ROT_270;
-            break;
-        default:
-            ALOGE("%s: Invalid HAL android.sensor.orientation value: %d",
-                  __FUNCTION__, orientation);
-            return INVALID_OPERATION;
+    if (!mirror) {
+        switch (orientation) {
+            case 0:
+                flags = 0;
+                break;
+            case 90:
+                flags = NATIVE_WINDOW_TRANSFORM_ROT_90;
+                break;
+            case 180:
+                flags = NATIVE_WINDOW_TRANSFORM_ROT_180;
+                break;
+            case 270:
+                flags = NATIVE_WINDOW_TRANSFORM_ROT_270;
+                break;
+            default:
+                ALOGE("%s: Invalid HAL android.sensor.orientation value: %d",
+                      __FUNCTION__, orientation);
+                return INVALID_OPERATION;
+        }
+    } else {
+        switch (orientation) {
+            case 0:
+                flags = HAL_TRANSFORM_FLIP_H;
+                break;
+            case 90:
+                flags = HAL_TRANSFORM_FLIP_H | HAL_TRANSFORM_ROT_90;
+                break;
+            case 180:
+                flags = HAL_TRANSFORM_FLIP_V;
+                break;
+            case 270:
+                flags = HAL_TRANSFORM_FLIP_V | HAL_TRANSFORM_ROT_90;
+                break;
+            default:
+                ALOGE("%s: Invalid HAL android.sensor.orientation value: %d",
+                      __FUNCTION__, orientation);
+                return INVALID_OPERATION;
+        }
+
     }
 
     /**

services/camera/libcameraservice/api2/CameraDeviceClient.h

diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.h b/services/camera/libcameraservice/api2/CameraDeviceClient.h
index b490924..b9c16aa 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.h
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.h
@@ -45,8 +45,6 @@
             uid_t clientUid,
             int servicePid);
 
-    virtual void notifyError();
-
     sp<ICameraDeviceCallbacks> mRemoteCallback;
 };
 
@@ -112,11 +110,19 @@
     virtual status_t      dump(int fd, const Vector<String16>& args);
 
     /**
+     * Device listener interface
+     */
+
+    virtual void notifyIdle();
+    virtual void notifyError();
+    virtual void notifyShutter(int requestId, nsecs_t timestamp);
+
+    /**
      * Interface used by independent components of CameraDeviceClient.
      */
 protected:
     /** FilteredListener implementation **/
-    virtual void          onFrameAvailable(int32_t frameId,
+    virtual void          onFrameAvailable(int32_t requestId,
                                            const CameraMetadata& frame);
     virtual void          detachDevice();
 

services/camera/libcameraservice/api_pro/ProCamera2Client.cpp

diff --git a/services/camera/libcameraservice/api_pro/ProCamera2Client.cpp b/services/camera/libcameraservice/api_pro/ProCamera2Client.cpp
index 2b583e5..1a7a7a7 100644
--- a/services/camera/libcameraservice/api_pro/ProCamera2Client.cpp
+++ b/services/camera/libcameraservice/api_pro/ProCamera2Client.cpp
@@ -374,7 +374,7 @@
 }
 
 /** Device-related methods */
-void ProCamera2Client::onFrameAvailable(int32_t frameId,
+void ProCamera2Client::onFrameAvailable(int32_t requestId,
                                         const CameraMetadata& frame) {
     ATRACE_CALL();
     ALOGV("%s", __FUNCTION__);
@@ -386,7 +386,7 @@
         CameraMetadata tmp(frame);
         camera_metadata_t* meta = tmp.release();
         ALOGV("%s: meta = %p ", __FUNCTION__, meta);
-        mRemoteCallback->onResultReceived(frameId, meta);
+        mRemoteCallback->onResultReceived(requestId, meta);
         tmp.acquire(meta);
     }
 

services/camera/libcameraservice/api_pro/ProCamera2Client.h

diff --git a/services/camera/libcameraservice/api_pro/ProCamera2Client.h b/services/camera/libcameraservice/api_pro/ProCamera2Client.h
index 0bf6784..8a0f547 100644
--- a/services/camera/libcameraservice/api_pro/ProCamera2Client.h
+++ b/services/camera/libcameraservice/api_pro/ProCamera2Client.h
@@ -97,7 +97,7 @@
 
 protected:
     /** FilteredListener implementation **/
-    virtual void          onFrameAvailable(int32_t frameId,
+    virtual void          onFrameAvailable(int32_t requestId,
                                            const CameraMetadata& frame);
     virtual void          detachDevice();
 

services/camera/libcameraservice/common/Camera2ClientBase.cpp

diff --git a/services/camera/libcameraservice/common/Camera2ClientBase.cpp b/services/camera/libcameraservice/common/Camera2ClientBase.cpp
index e808bf3..2d1253f 100644
--- a/services/camera/libcameraservice/common/Camera2ClientBase.cpp
+++ b/services/camera/libcameraservice/common/Camera2ClientBase.cpp
@@ -226,13 +226,18 @@
 }
 
 template <typename TClientBase>
-void Camera2ClientBase<TClientBase>::notifyShutter(int frameNumber,
+void Camera2ClientBase<TClientBase>::notifyIdle() {
+    ALOGV("Camera device is now idle");
+}
+
+template <typename TClientBase>
+void Camera2ClientBase<TClientBase>::notifyShutter(int requestId,
                                                    nsecs_t timestamp) {
-    (void)frameNumber;
+    (void)requestId;
     (void)timestamp;
 
-    ALOGV("%s: Shutter notification for frame %d at time %lld", __FUNCTION__,
-          frameNumber, timestamp);
+    ALOGV("%s: Shutter notification for request id %d at time %lld",
+            __FUNCTION__, requestId, timestamp);
 }
 
 template <typename TClientBase>
@@ -244,13 +249,6 @@
     ALOGV("%s: Autofocus state now %d, last trigger %d",
           __FUNCTION__, newState, triggerId);
 
-    typename SharedCameraCallbacks::Lock l(mSharedCameraCallbacks);
-    if (l.mRemoteCallback != 0) {
-        l.mRemoteCallback->notifyCallback(CAMERA_MSG_FOCUS_MOVE, 1, 0);
-    }
-    if (l.mRemoteCallback != 0) {
-        l.mRemoteCallback->notifyCallback(CAMERA_MSG_FOCUS, 1, 0);
-    }
 }
 
 template <typename TClientBase>

services/camera/libcameraservice/common/Camera2ClientBase.h

diff --git a/services/camera/libcameraservice/common/Camera2ClientBase.h b/services/camera/libcameraservice/common/Camera2ClientBase.h
index d23197c..61e44f0 100644
--- a/services/camera/libcameraservice/common/Camera2ClientBase.h
+++ b/services/camera/libcameraservice/common/Camera2ClientBase.h
@@ -62,7 +62,8 @@
      */
 
     virtual void          notifyError(int errorCode, int arg1, int arg2);
-    virtual void          notifyShutter(int frameNumber, nsecs_t timestamp);
+    virtual void          notifyIdle();
+    virtual void          notifyShutter(int requestId, nsecs_t timestamp);
     virtual void          notifyAutoFocus(uint8_t newState, int triggerId);
     virtual void          notifyAutoExposure(uint8_t newState, int triggerId);
     virtual void          notifyAutoWhitebalance(uint8_t newState,

services/camera/libcameraservice/common/CameraDeviceBase.h

diff --git a/services/camera/libcameraservice/common/CameraDeviceBase.h b/services/camera/libcameraservice/common/CameraDeviceBase.h
index ebbd4ea..e80abf1 100644
--- a/services/camera/libcameraservice/common/CameraDeviceBase.h
+++ b/services/camera/libcameraservice/common/CameraDeviceBase.h
@@ -138,9 +138,18 @@
      */
     class NotificationListener {
       public:
-        // Refer to the Camera2 HAL definition for notification definitions
+        // The set of notifications is a merge of the notifications required for
+        // API1 and API2.
+
+        // Required for API 1 and 2
         virtual void notifyError(int errorCode, int arg1, int arg2) = 0;
-        virtual void notifyShutter(int frameNumber, nsecs_t timestamp) = 0;
+
+        // Required only for API2
+        virtual void notifyIdle() = 0;
+        virtual void notifyShutter(int requestId,
+                nsecs_t timestamp) = 0;
+
+        // Required only for API1
         virtual void notifyAutoFocus(uint8_t newState, int triggerId) = 0;
         virtual void notifyAutoExposure(uint8_t newState, int triggerId) = 0;
         virtual void notifyAutoWhitebalance(uint8_t newState,
@@ -165,12 +174,14 @@
     /**
      * Wait for a new frame to be produced, with timeout in nanoseconds.
      * Returns TIMED_OUT when no frame produced within the specified duration
+     * May be called concurrently to most methods, except for getNextFrame
      */
     virtual status_t waitForNextFrame(nsecs_t timeout) = 0;
 
     /**
      * Get next metadata frame from the frame queue. Returns NULL if the queue
      * is empty; caller takes ownership of the metadata buffer.
+     * May be called concurrently to most methods, except for waitForNextFrame
      */
     virtual status_t getNextFrame(CameraMetadata *frame) = 0;
 

services/camera/libcameraservice/common/FrameProcessorBase.cpp

diff --git a/services/camera/libcameraservice/common/FrameProcessorBase.cpp b/services/camera/libcameraservice/common/FrameProcessorBase.cpp
index e7b440a..f2064fb 100644
--- a/services/camera/libcameraservice/common/FrameProcessorBase.cpp
+++ b/services/camera/libcameraservice/common/FrameProcessorBase.cpp
@@ -37,11 +37,11 @@
 }
 
 status_t FrameProcessorBase::registerListener(int32_t minId,
-        int32_t maxId, wp<FilteredListener> listener) {
+        int32_t maxId, wp<FilteredListener> listener, bool quirkSendPartials) {
     Mutex::Autolock l(mInputMutex);
     ALOGV("%s: Registering listener for frame id range %d - %d",
             __FUNCTION__, minId, maxId);
-    RangeListener rListener = { minId, maxId, listener };
+    RangeListener rListener = { minId, maxId, listener, quirkSendPartials };
     mRangeListeners.push_back(rListener);
     return OK;
 }
@@ -66,7 +66,14 @@
 void FrameProcessorBase::dump(int fd, const Vector<String16>& /*args*/) {
     String8 result("    Latest received frame:\n");
     write(fd, result.string(), result.size());
-    mLastFrame.dump(fd, 2, 6);
+
+    CameraMetadata lastFrame;
+    {
+        // Don't race while dumping metadata
+        Mutex::Autolock al(mLastFrameMutex);
+        lastFrame = CameraMetadata(mLastFrame);
+    }
+    lastFrame.dump(fd, 2, 6);
 }
 
 bool FrameProcessorBase::threadLoop() {
@@ -113,6 +120,7 @@
         }
 
         if (!frame.isEmpty()) {
+            Mutex::Autolock al(mLastFrameMutex);
             mLastFrame.acquire(frame);
         }
     }
@@ -137,6 +145,16 @@
     ATRACE_CALL();
     camera_metadata_ro_entry_t entry;
 
+    // Quirks: Don't deliver partial results to listeners that don't want them
+    bool quirkIsPartial = false;
+    entry = frame.find(ANDROID_QUIRKS_PARTIAL_RESULT);
+    if (entry.count != 0 &&
+            entry.data.u8[0] == ANDROID_QUIRKS_PARTIAL_RESULT_PARTIAL) {
+        ALOGV("%s: Camera %d: Not forwarding partial result to listeners",
+                __FUNCTION__, device->getId());
+        quirkIsPartial = true;
+    }
+
     entry = frame.find(ANDROID_REQUEST_ID);
     if (entry.count == 0) {
         ALOGE("%s: Camera %d: Error reading frame id",
@@ -152,7 +170,8 @@
         List<RangeListener>::iterator item = mRangeListeners.begin();
         while (item != mRangeListeners.end()) {
             if (requestId >= item->minId &&
-                    requestId < item->maxId) {
+                    requestId < item->maxId &&
+                    (!quirkIsPartial || item->quirkSendPartials) ) {
                 sp<FilteredListener> listener = item->listener.promote();
                 if (listener == 0) {
                     item = mRangeListeners.erase(item);

services/camera/libcameraservice/common/FrameProcessorBase.h

diff --git a/services/camera/libcameraservice/common/FrameProcessorBase.h b/services/camera/libcameraservice/common/FrameProcessorBase.h
index 1e46beb..89b608a 100644
--- a/services/camera/libcameraservice/common/FrameProcessorBase.h
+++ b/services/camera/libcameraservice/common/FrameProcessorBase.h
@@ -39,14 +39,16 @@
     virtual ~FrameProcessorBase();
 
     struct FilteredListener: virtual public RefBase {
-        virtual void onFrameAvailable(int32_t frameId,
+        virtual void onFrameAvailable(int32_t requestId,
                                       const CameraMetadata &frame) = 0;
     };
 
     // Register a listener for a range of IDs [minId, maxId). Multiple listeners
-    // can be listening to the same range
+    // can be listening to the same range.
+    // QUIRK: sendPartials controls whether partial results will be sent.
     status_t registerListener(int32_t minId, int32_t maxId,
-                              wp<FilteredListener> listener);
+                              wp<FilteredListener> listener,
+                              bool quirkSendPartials = true);
     status_t removeListener(int32_t minId, int32_t maxId,
                             wp<FilteredListener> listener);
 
@@ -58,11 +60,13 @@
     virtual bool threadLoop();
 
     Mutex mInputMutex;
+    Mutex mLastFrameMutex;
 
     struct RangeListener {
         int32_t minId;
         int32_t maxId;
         wp<FilteredListener> listener;
+        bool quirkSendPartials;
     };
     List<RangeListener> mRangeListeners;
 

services/camera/libcameraservice/device2/Camera2Device.cpp

diff --git a/services/camera/libcameraservice/device2/Camera2Device.cpp b/services/camera/libcameraservice/device2/Camera2Device.cpp
index fe2cd77..dc97c47 100644
--- a/services/camera/libcameraservice/device2/Camera2Device.cpp
+++ b/services/camera/libcameraservice/device2/Camera2Device.cpp
@@ -25,6 +25,7 @@
 #define ALOGVV(...) ((void)0)
 #endif
 
+#include <inttypes.h>
 #include <utils/Log.h>
 #include <utils/Trace.h>
 #include <utils/Timers.h>
@@ -464,8 +465,10 @@
                 listener->notifyError(ext1, ext2, ext3);
                 break;
             case CAMERA2_MSG_SHUTTER: {
-                nsecs_t timestamp = (nsecs_t)ext2 | ((nsecs_t)(ext3) << 32 );
-                listener->notifyShutter(ext1, timestamp);
+                // TODO: Only needed for camera2 API, which is unsupported
+                // by HAL2 directly.
+                // nsecs_t timestamp = (nsecs_t)ext2 | ((nsecs_t)(ext3) << 32 );
+                // listener->notifyShutter(requestId, timestamp);
                 break;
             }
             case CAMERA2_MSG_AUTOFOCUS:
@@ -820,7 +823,7 @@
         result.append("      Stream slot: Empty\n");
         write(fd, result.string(), result.size());
     } else {
-        result.appendFormat("      Stream slot: %d entries\n",
+        result.appendFormat("      Stream slot: %zu entries\n",
                 mStreamSlot.size());
         int i = 0;
         for (List<camera_metadata_t*>::iterator r = mStreamSlot.begin();
@@ -835,7 +838,7 @@
         result = "      Main queue is empty\n";
         write(fd, result.string(), result.size());
     } else {
-        result = String8::format("      Main queue has %d entries:\n",
+        result = String8::format("      Main queue has %zu entries:\n",
                 mEntries.size());
         int i = 0;
         for (List<camera_metadata_t*>::iterator r = mEntries.begin();
@@ -1212,11 +1215,11 @@
     ATRACE_CALL();
     String8 result = String8::format("      Stream %d: %d x %d, format 0x%x\n",
             mId, mWidth, mHeight, mFormat);
-    result.appendFormat("        size %d, usage 0x%x, requested format 0x%x\n",
+    result.appendFormat("        size %zu, usage 0x%x, requested format 0x%x\n",
             mSize, mUsage, mFormatRequested);
     result.appendFormat("        total buffers: %d, dequeued buffers: %d\n",
             mTotalBuffers, mActiveBuffers);
-    result.appendFormat("        frame count: %d, last timestamp %lld\n",
+    result.appendFormat("        frame count: %d, last timestamp %" PRId64 "\n",
             mFrameCount, mLastTimestamp);
     write(fd, result.string(), result.size());
     return OK;

services/camera/libcameraservice/device2/Camera2Device.h

diff --git a/services/camera/libcameraservice/device2/Camera2Device.h b/services/camera/libcameraservice/device2/Camera2Device.h
index 2aa22a2..1f53c56 100644
--- a/services/camera/libcameraservice/device2/Camera2Device.h
+++ b/services/camera/libcameraservice/device2/Camera2Device.h
@@ -28,6 +28,10 @@
 
 /**
  * CameraDevice for HAL devices with version CAMERA_DEVICE_API_VERSION_2_0
+ *
+ * TODO for camera2 API implementation:
+ * Does not produce notifyShutter / notifyIdle callbacks to NotificationListener
+ * Use waitUntilDrained for idle.
  */
 class Camera2Device: public CameraDeviceBase {
   public:

services/camera/libcameraservice/device3/Camera3Device.cpp

diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index b468eb3..da3e121 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -41,6 +41,7 @@
 #include <utils/Trace.h>
 #include <utils/Timers.h>
 
+#include "utils/CameraTraces.h"
 #include "device3/Camera3Device.h"
 #include "device3/Camera3OutputStream.h"
 #include "device3/Camera3InputStream.h"
@@ -54,6 +55,7 @@
         mId(id),
         mHal3Device(NULL),
         mStatus(STATUS_UNINITIALIZED),
+        mUsePartialResultQuirk(false),
         mNextResultFrameNumber(0),
         mNextShutterFrameNumber(0),
         mListener(NULL)
@@ -82,6 +84,7 @@
 status_t Camera3Device::initialize(camera_module_t *module)
 {
     ATRACE_CALL();
+    Mutex::Autolock il(mInterfaceLock);
     Mutex::Autolock l(mLock);
 
     ALOGV("%s: Initializing device for camera %d", __FUNCTION__, mId);
@@ -97,8 +100,10 @@
 
     camera3_device_t *device;
 
+    ATRACE_BEGIN("camera3->open");
     res = module->common.methods->open(&module->common, deviceName.string(),
             reinterpret_cast<hw_device_t**>(&device));
+    ATRACE_END();
 
     if (res != OK) {
         SET_ERR_L("Could not open camera: %s (%d)", strerror(-res), res);
@@ -159,9 +164,20 @@
         }
     }
 
+    /** Start up status tracker thread */
+    mStatusTracker = new StatusTracker(this);
+    res = mStatusTracker->run(String8::format("C3Dev-%d-Status", mId).string());
+    if (res != OK) {
+        SET_ERR_L("Unable to start status tracking thread: %s (%d)",
+                strerror(-res), res);
+        device->common.close(&device->common);
+        mStatusTracker.clear();
+        return res;
+    }
+
     /** Start up request queue thread */
 
-    mRequestThread = new RequestThread(this, device);
+    mRequestThread = new RequestThread(this, mStatusTracker, device);
     res = mRequestThread->run(String8::format("C3Dev-%d-ReqQueue", mId).string());
     if (res != OK) {
         SET_ERR_L("Unable to start request queue thread: %s (%d)",
@@ -175,81 +191,141 @@
 
     mDeviceInfo = info.static_camera_characteristics;
     mHal3Device = device;
-    mStatus = STATUS_IDLE;
+    mStatus = STATUS_UNCONFIGURED;
     mNextStreamId = 0;
     mNeedConfig = true;
+    mPauseStateNotify = false;
+
+    /** Check for quirks */
+
+    // Will the HAL be sending in early partial result metadata?
+    camera_metadata_entry partialResultsQuirk =
+            mDeviceInfo.find(ANDROID_QUIRKS_USE_PARTIAL_RESULT);
+    if (partialResultsQuirk.count > 0 && partialResultsQuirk.data.u8[0] == 1) {
+        mUsePartialResultQuirk = true;
+    }
 
     return OK;
 }
 
 status_t Camera3Device::disconnect() {
     ATRACE_CALL();
-    Mutex::Autolock l(mLock);
+    Mutex::Autolock il(mInterfaceLock);
 
     ALOGV("%s: E", __FUNCTION__);
 
     status_t res = OK;
-    if (mStatus == STATUS_UNINITIALIZED) return res;
 
-    if (mStatus == STATUS_ACTIVE ||
-            (mStatus == STATUS_ERROR && mRequestThread != NULL)) {
-        res = mRequestThread->clearRepeatingRequests();
-        if (res != OK) {
-            SET_ERR_L("Can't stop streaming");
-            // Continue to close device even in case of error
-        } else {
-            res = waitUntilDrainedLocked();
+    {
+        Mutex::Autolock l(mLock);
+        if (mStatus == STATUS_UNINITIALIZED) return res;
+
+        if (mStatus == STATUS_ACTIVE ||
+                (mStatus == STATUS_ERROR && mRequestThread != NULL)) {
+            res = mRequestThread->clearRepeatingRequests();
             if (res != OK) {
-                SET_ERR_L("Timeout waiting for HAL to drain");
+                SET_ERR_L("Can't stop streaming");
                 // Continue to close device even in case of error
+            } else {
+                res = waitUntilStateThenRelock(/*active*/ false, kShutdownTimeout);
+                if (res != OK) {
+                    SET_ERR_L("Timeout waiting for HAL to drain");
+                    // Continue to close device even in case of error
+                }
             }
         }
-    }
-    assert(mStatus == STATUS_IDLE || mStatus == STATUS_ERROR);
 
-    if (mStatus == STATUS_ERROR) {
-        CLOGE("Shutting down in an error state");
-    }
-
-    if (mRequestThread != NULL) {
-        mRequestThread->requestExit();
-    }
-
-    mOutputStreams.clear();
-    mInputStream.clear();
-
-    if (mRequestThread != NULL) {
-        if (mStatus != STATUS_ERROR) {
-            // HAL may be in a bad state, so waiting for request thread
-            // (which may be stuck in the HAL processCaptureRequest call)
-            // could be dangerous.
-            mRequestThread->join();
+        if (mStatus == STATUS_ERROR) {
+            CLOGE("Shutting down in an error state");
         }
+
+        if (mStatusTracker != NULL) {
+            mStatusTracker->requestExit();
+        }
+
+        if (mRequestThread != NULL) {
+            mRequestThread->requestExit();
+        }
+
+        mOutputStreams.clear();
+        mInputStream.clear();
+    }
+
+    // Joining done without holding mLock, otherwise deadlocks may ensue
+    // as the threads try to access parent state
+    if (mRequestThread != NULL && mStatus != STATUS_ERROR) {
+        // HAL may be in a bad state, so waiting for request thread
+        // (which may be stuck in the HAL processCaptureRequest call)
+        // could be dangerous.
+        mRequestThread->join();
+    }
+
+    if (mStatusTracker != NULL) {
+        mStatusTracker->join();
+    }
+
+    {
+        Mutex::Autolock l(mLock);
+
         mRequestThread.clear();
-    }
+        mStatusTracker.clear();
 
-    if (mHal3Device != NULL) {
-        mHal3Device->common.close(&mHal3Device->common);
-        mHal3Device = NULL;
-    }
+        if (mHal3Device != NULL) {
+            ATRACE_BEGIN("camera3->close");
+            mHal3Device->common.close(&mHal3Device->common);
+            ATRACE_END();
+            mHal3Device = NULL;
+        }
 
-    mStatus = STATUS_UNINITIALIZED;
+        mStatus = STATUS_UNINITIALIZED;
+    }
 
     ALOGV("%s: X", __FUNCTION__);
     return res;
 }
 
+// For dumping/debugging only -
+// try to acquire a lock a few times, eventually give up to proceed with
+// debug/dump operations
+bool Camera3Device::tryLockSpinRightRound(Mutex& lock) {
+    bool gotLock = false;
+    for (size_t i = 0; i < kDumpLockAttempts; ++i) {
+        if (lock.tryLock() == NO_ERROR) {
+            gotLock = true;
+            break;
+        } else {
+            usleep(kDumpSleepDuration);
+        }
+    }
+    return gotLock;
+}
+
 status_t Camera3Device::dump(int fd, const Vector<String16> &args) {
     ATRACE_CALL();
     (void)args;
+
+    // Try to lock, but continue in case of failure (to avoid blocking in
+    // deadlocks)
+    bool gotInterfaceLock = tryLockSpinRightRound(mInterfaceLock);
+    bool gotLock = tryLockSpinRightRound(mLock);
+
+    ALOGW_IF(!gotInterfaceLock,
+            "Camera %d: %s: Unable to lock interface lock, proceeding anyway",
+            mId, __FUNCTION__);
+    ALOGW_IF(!gotLock,
+            "Camera %d: %s: Unable to lock main lock, proceeding anyway",
+            mId, __FUNCTION__);
+
     String8 lines;
 
     const char *status =
             mStatus == STATUS_ERROR         ? "ERROR" :
             mStatus == STATUS_UNINITIALIZED ? "UNINITIALIZED" :
-            mStatus == STATUS_IDLE          ? "IDLE" :
+            mStatus == STATUS_UNCONFIGURED  ? "UNCONFIGURED" :
+            mStatus == STATUS_CONFIGURED    ? "CONFIGURED" :
             mStatus == STATUS_ACTIVE        ? "ACTIVE" :
             "Unknown";
+
     lines.appendFormat("    Device status: %s\n", status);
     if (mStatus == STATUS_ERROR) {
         lines.appendFormat("    Error cause: %s\n", mErrorCause.string());
@@ -285,7 +361,7 @@
         lines = String8("    Last request sent:\n");
         write(fd, lines.string(), lines.size());
 
-        CameraMetadata lastRequest = getLatestRequest();
+        CameraMetadata lastRequest = getLatestRequestLocked();
         lastRequest.dump(fd, /*verbosity*/2, /*indentation*/6);
     }
 
@@ -295,6 +371,9 @@
         mHal3Device->ops->dump(mHal3Device, fd);
     }
 
+    if (gotLock) mLock.unlock();
+    if (gotInterfaceLock) mInterfaceLock.unlock();
+
     return OK;
 }
 
@@ -311,6 +390,8 @@
 
 status_t Camera3Device::capture(CameraMetadata &request) {
     ATRACE_CALL();
+    status_t res;
+    Mutex::Autolock il(mInterfaceLock);
     Mutex::Autolock l(mLock);
 
     // TODO: take ownership of the request
@@ -322,7 +403,9 @@
         case STATUS_UNINITIALIZED:
             CLOGE("Device not initialized");
             return INVALID_OPERATION;
-        case STATUS_IDLE:
+        case STATUS_UNCONFIGURED:
+            // May be lazily configuring streams, will check during setup
+        case STATUS_CONFIGURED:
         case STATUS_ACTIVE:
             // OK
             break;
@@ -337,12 +420,23 @@
         return BAD_VALUE;
     }
 
-    return mRequestThread->queueRequest(newRequest);
+    res = mRequestThread->queueRequest(newRequest);
+    if (res == OK) {
+        waitUntilStateThenRelock(/*active*/ true, kActiveTimeout);
+        if (res != OK) {
+            SET_ERR_L("Can't transition to active in %f seconds!",
+                    kActiveTimeout/1e9);
+        }
+        ALOGV("Camera %d: Capture request enqueued", mId);
+    }
+    return res;
 }
 
 
 status_t Camera3Device::setStreamingRequest(const CameraMetadata &request) {
     ATRACE_CALL();
+    status_t res;
+    Mutex::Autolock il(mInterfaceLock);
     Mutex::Autolock l(mLock);
 
     switch (mStatus) {
@@ -352,7 +446,9 @@
         case STATUS_UNINITIALIZED:
             CLOGE("Device not initialized");
             return INVALID_OPERATION;
-        case STATUS_IDLE:
+        case STATUS_UNCONFIGURED:
+            // May be lazily configuring streams, will check during setup
+        case STATUS_CONFIGURED:
         case STATUS_ACTIVE:
             // OK
             break;
@@ -370,7 +466,16 @@
     RequestList newRepeatingRequests;
     newRepeatingRequests.push_back(newRepeatingRequest);
 
-    return mRequestThread->setRepeatingRequests(newRepeatingRequests);
+    res = mRequestThread->setRepeatingRequests(newRepeatingRequests);
+    if (res == OK) {
+        waitUntilStateThenRelock(/*active*/ true, kActiveTimeout);
+        if (res != OK) {
+            SET_ERR_L("Can't transition to active in %f seconds!",
+                    kActiveTimeout/1e9);
+        }
+        ALOGV("Camera %d: Repeating request set", mId);
+    }
+    return res;
 }
 
 
@@ -378,12 +483,16 @@
         const CameraMetadata &request) {
     status_t res;
 
-    if (mStatus == STATUS_IDLE) {
+    if (mStatus == STATUS_UNCONFIGURED || mNeedConfig) {
         res = configureStreamsLocked();
         if (res != OK) {
             SET_ERR_L("Can't set up streams: %s (%d)", strerror(-res), res);
             return NULL;
         }
+        if (mStatus == STATUS_UNCONFIGURED) {
+            CLOGE("No streams configured");
+            return NULL;
+        }
     }
 
     sp<CaptureRequest> newRequest = createCaptureRequest(request);
@@ -392,6 +501,7 @@
 
 status_t Camera3Device::clearStreamingRequest() {
     ATRACE_CALL();
+    Mutex::Autolock il(mInterfaceLock);
     Mutex::Autolock l(mLock);
 
     switch (mStatus) {
@@ -401,7 +511,8 @@
         case STATUS_UNINITIALIZED:
             CLOGE("Device not initialized");
             return INVALID_OPERATION;
-        case STATUS_IDLE:
+        case STATUS_UNCONFIGURED:
+        case STATUS_CONFIGURED:
         case STATUS_ACTIVE:
             // OK
             break;
@@ -409,12 +520,13 @@
             SET_ERR_L("Unexpected status: %d", mStatus);
             return INVALID_OPERATION;
     }
-
+    ALOGV("Camera %d: Clearing repeating request", mId);
     return mRequestThread->clearRepeatingRequests();
 }
 
 status_t Camera3Device::waitUntilRequestReceived(int32_t requestId, nsecs_t timeout) {
     ATRACE_CALL();
+    Mutex::Autolock il(mInterfaceLock);
 
     return mRequestThread->waitUntilRequestProcessed(requestId, timeout);
 }
@@ -422,7 +534,10 @@
 status_t Camera3Device::createInputStream(
         uint32_t width, uint32_t height, int format, int *id) {
     ATRACE_CALL();
+    Mutex::Autolock il(mInterfaceLock);
     Mutex::Autolock l(mLock);
+    ALOGV("Camera %d: Creating new input stream %d: %d x %d, format %d",
+            mId, mNextStreamId, width, height, format);
 
     status_t res;
     bool wasActive = false;
@@ -434,26 +549,24 @@
         case STATUS_UNINITIALIZED:
             ALOGE("%s: Device not initialized", __FUNCTION__);
             return INVALID_OPERATION;
-        case STATUS_IDLE:
+        case STATUS_UNCONFIGURED:
+        case STATUS_CONFIGURED:
             // OK
             break;
         case STATUS_ACTIVE:
             ALOGV("%s: Stopping activity to reconfigure streams", __FUNCTION__);
-            mRequestThread->setPaused(true);
-            res = waitUntilDrainedLocked();
+            res = internalPauseAndWaitLocked();
             if (res != OK) {
-                ALOGE("%s: Can't pause captures to reconfigure streams!",
-                        __FUNCTION__);
-                mStatus = STATUS_ERROR;
+                SET_ERR_L("Can't pause captures to reconfigure streams!");
                 return res;
             }
             wasActive = true;
             break;
         default:
-            ALOGE("%s: Unexpected status: %d", __FUNCTION__, mStatus);
+            SET_ERR_L("%s: Unexpected status: %d", mStatus);
             return INVALID_OPERATION;
     }
-    assert(mStatus == STATUS_IDLE);
+    assert(mStatus != STATUS_ACTIVE);
 
     if (mInputStream != 0) {
         ALOGE("%s: Cannot create more than 1 input stream", __FUNCTION__);
@@ -462,6 +575,7 @@
 
     sp<Camera3InputStream> newStream = new Camera3InputStream(mNextStreamId,
                 width, height, format);
+    newStream->setStatusTracker(mStatusTracker);
 
     mInputStream = newStream;
 
@@ -476,9 +590,10 @@
                     __FUNCTION__, mNextStreamId, strerror(-res), res);
             return res;
         }
-        mRequestThread->setPaused(false);
+        internalResumeLocked();
     }
 
+    ALOGV("Camera %d: Created input stream", mId);
     return OK;
 }
 
@@ -490,7 +605,10 @@
             int *id,
             sp<Camera3ZslStream>* zslStream) {
     ATRACE_CALL();
+    Mutex::Autolock il(mInterfaceLock);
     Mutex::Autolock l(mLock);
+    ALOGV("Camera %d: Creating ZSL stream %d: %d x %d, depth %d",
+            mId, mNextStreamId, width, height, depth);
 
     status_t res;
     bool wasActive = false;
@@ -502,26 +620,24 @@
         case STATUS_UNINITIALIZED:
             ALOGE("%s: Device not initialized", __FUNCTION__);
             return INVALID_OPERATION;
-        case STATUS_IDLE:
+        case STATUS_UNCONFIGURED:
+        case STATUS_CONFIGURED:
             // OK
             break;
         case STATUS_ACTIVE:
             ALOGV("%s: Stopping activity to reconfigure streams", __FUNCTION__);
-            mRequestThread->setPaused(true);
-            res = waitUntilDrainedLocked();
+            res = internalPauseAndWaitLocked();
             if (res != OK) {
-                ALOGE("%s: Can't pause captures to reconfigure streams!",
-                        __FUNCTION__);
-                mStatus = STATUS_ERROR;
+                SET_ERR_L("Can't pause captures to reconfigure streams!");
                 return res;
             }
             wasActive = true;
             break;
         default:
-            ALOGE("%s: Unexpected status: %d", __FUNCTION__, mStatus);
+            SET_ERR_L("Unexpected status: %d", mStatus);
             return INVALID_OPERATION;
     }
-    assert(mStatus == STATUS_IDLE);
+    assert(mStatus != STATUS_ACTIVE);
 
     if (mInputStream != 0) {
         ALOGE("%s: Cannot create more than 1 input stream", __FUNCTION__);
@@ -530,6 +646,7 @@
 
     sp<Camera3ZslStream> newStream = new Camera3ZslStream(mNextStreamId,
                 width, height, depth);
+    newStream->setStatusTracker(mStatusTracker);
 
     res = mOutputStreams.add(mNextStreamId, newStream);
     if (res < 0) {
@@ -551,16 +668,20 @@
                     __FUNCTION__, mNextStreamId, strerror(-res), res);
             return res;
         }
-        mRequestThread->setPaused(false);
+        internalResumeLocked();
     }
 
+    ALOGV("Camera %d: Created ZSL stream", mId);
     return OK;
 }
 
 status_t Camera3Device::createStream(sp<ANativeWindow> consumer,
         uint32_t width, uint32_t height, int format, size_t size, int *id) {
     ATRACE_CALL();
+    Mutex::Autolock il(mInterfaceLock);
     Mutex::Autolock l(mLock);
+    ALOGV("Camera %d: Creating new stream %d: %d x %d, format %d, size %d",
+            mId, mNextStreamId, width, height, format, size);
 
     status_t res;
     bool wasActive = false;
@@ -572,16 +693,15 @@
         case STATUS_UNINITIALIZED:
             CLOGE("Device not initialized");
             return INVALID_OPERATION;
-        case STATUS_IDLE:
+        case STATUS_UNCONFIGURED:
+        case STATUS_CONFIGURED:
             // OK
             break;
         case STATUS_ACTIVE:
             ALOGV("%s: Stopping activity to reconfigure streams", __FUNCTION__);
-            mRequestThread->setPaused(true);
-            res = waitUntilDrainedLocked();
+            res = internalPauseAndWaitLocked();
             if (res != OK) {
-                ALOGE("%s: Can't pause captures to reconfigure streams!",
-                        __FUNCTION__);
+                SET_ERR_L("Can't pause captures to reconfigure streams!");
                 return res;
             }
             wasActive = true;
@@ -590,7 +710,7 @@
             SET_ERR_L("Unexpected status: %d", mStatus);
             return INVALID_OPERATION;
     }
-    assert(mStatus == STATUS_IDLE);
+    assert(mStatus != STATUS_ACTIVE);
 
     sp<Camera3OutputStream> newStream;
     if (format == HAL_PIXEL_FORMAT_BLOB) {
@@ -600,6 +720,7 @@
         newStream = new Camera3OutputStream(mNextStreamId, consumer,
                 width, height, format);
     }
+    newStream->setStatusTracker(mStatusTracker);
 
     res = mOutputStreams.add(mNextStreamId, newStream);
     if (res < 0) {
@@ -619,9 +740,9 @@
                     mNextStreamId, strerror(-res), res);
             return res;
         }
-        mRequestThread->setPaused(false);
+        internalResumeLocked();
     }
-
+    ALOGV("Camera %d: Created new stream", mId);
     return OK;
 }
 
@@ -637,6 +758,7 @@
 status_t Camera3Device::getStreamInfo(int id,
         uint32_t *width, uint32_t *height, uint32_t *format) {
     ATRACE_CALL();
+    Mutex::Autolock il(mInterfaceLock);
     Mutex::Autolock l(mLock);
 
     switch (mStatus) {
@@ -646,7 +768,8 @@
         case STATUS_UNINITIALIZED:
             CLOGE("Device not initialized!");
             return INVALID_OPERATION;
-        case STATUS_IDLE:
+        case STATUS_UNCONFIGURED:
+        case STATUS_CONFIGURED:
         case STATUS_ACTIVE:
             // OK
             break;
@@ -671,6 +794,7 @@
 status_t Camera3Device::setStreamTransform(int id,
         int transform) {
     ATRACE_CALL();
+    Mutex::Autolock il(mInterfaceLock);
     Mutex::Autolock l(mLock);
 
     switch (mStatus) {
@@ -680,7 +804,8 @@
         case STATUS_UNINITIALIZED:
             CLOGE("Device not initialized");
             return INVALID_OPERATION;
-        case STATUS_IDLE:
+        case STATUS_UNCONFIGURED:
+        case STATUS_CONFIGURED:
         case STATUS_ACTIVE:
             // OK
             break;
@@ -701,6 +826,7 @@
 
 status_t Camera3Device::deleteStream(int id) {
     ATRACE_CALL();
+    Mutex::Autolock il(mInterfaceLock);
     Mutex::Autolock l(mLock);
     status_t res;
 
@@ -708,22 +834,26 @@
 
     // CameraDevice semantics require device to already be idle before
     // deleteStream is called, unlike for createStream.
-    if (mStatus != STATUS_IDLE) {
+    if (mStatus == STATUS_ACTIVE) {
         ALOGV("%s: Camera %d: Device not idle", __FUNCTION__, mId);
         return -EBUSY;
     }
 
     sp<Camera3StreamInterface> deletedStream;
+    ssize_t outputStreamIdx = mOutputStreams.indexOfKey(id);
     if (mInputStream != NULL && id == mInputStream->getId()) {
         deletedStream = mInputStream;
         mInputStream.clear();
     } else {
-        ssize_t idx = mOutputStreams.indexOfKey(id);
-        if (idx == NAME_NOT_FOUND) {
+        if (outputStreamIdx == NAME_NOT_FOUND) {
             CLOGE("Stream %d does not exist", id);
             return BAD_VALUE;
         }
-        deletedStream = mOutputStreams.editValueAt(idx);
+    }
+
+    // Delete output stream or the output part of a bi-directional stream.
+    if (outputStreamIdx != NAME_NOT_FOUND) {
+        deletedStream = mOutputStreams.editValueAt(outputStreamIdx);
         mOutputStreams.removeItem(id);
     }
 
@@ -752,6 +882,7 @@
         CameraMetadata *request) {
     ATRACE_CALL();
     ALOGV("%s: for template %d", __FUNCTION__, templateId);
+    Mutex::Autolock il(mInterfaceLock);
     Mutex::Autolock l(mLock);
 
     switch (mStatus) {
@@ -761,7 +892,8 @@
         case STATUS_UNINITIALIZED:
             CLOGE("Device is not initialized!");
             return INVALID_OPERATION;
-        case STATUS_IDLE:
+        case STATUS_UNCONFIGURED:
+        case STATUS_CONFIGURED:
         case STATUS_ACTIVE:
             // OK
             break;
@@ -787,61 +919,88 @@
 
 status_t Camera3Device::waitUntilDrained() {
     ATRACE_CALL();
+    Mutex::Autolock il(mInterfaceLock);
     Mutex::Autolock l(mLock);
 
-    return waitUntilDrainedLocked();
-}
-
-status_t Camera3Device::waitUntilDrainedLocked() {
-    ATRACE_CALL();
-    status_t res;
-
     switch (mStatus) {
         case STATUS_UNINITIALIZED:
-        case STATUS_IDLE:
+        case STATUS_UNCONFIGURED:
             ALOGV("%s: Already idle", __FUNCTION__);
             return OK;
+        case STATUS_CONFIGURED:
+            // To avoid race conditions, check with tracker to be sure
         case STATUS_ERROR:
         case STATUS_ACTIVE:
-            // Need to shut down
+            // Need to verify shut down
             break;
         default:
             SET_ERR_L("Unexpected status: %d",mStatus);
             return INVALID_OPERATION;
     }
 
-    if (mRequestThread != NULL) {
-        res = mRequestThread->waitUntilPaused(kShutdownTimeout);
-        if (res != OK) {
-            SET_ERR_L("Can't stop request thread in %f seconds!",
-                    kShutdownTimeout/1e9);
-            return res;
-        }
-    }
-    if (mInputStream != NULL) {
-        res = mInputStream->waitUntilIdle(kShutdownTimeout);
-        if (res != OK) {
-            SET_ERR_L("Can't idle input stream %d in %f seconds!",
-                    mInputStream->getId(), kShutdownTimeout/1e9);
-            return res;
-        }
-    }
-    for (size_t i = 0; i < mOutputStreams.size(); i++) {
-        res = mOutputStreams.editValueAt(i)->waitUntilIdle(kShutdownTimeout);
-        if (res != OK) {
-            SET_ERR_L("Can't idle output stream %d in %f seconds!",
-                    mOutputStreams.keyAt(i), kShutdownTimeout/1e9);
-            return res;
-        }
+    ALOGV("%s: Camera %d: Waiting until idle", __FUNCTION__, mId);
+    status_t res = waitUntilStateThenRelock(/*active*/ false, kShutdownTimeout);
+    return res;
+}
+
+// Pause to reconfigure
+status_t Camera3Device::internalPauseAndWaitLocked() {
+    mRequestThread->setPaused(true);
+    mPauseStateNotify = true;
+
+    ALOGV("%s: Camera %d: Internal wait until idle", __FUNCTION__, mId);
+    status_t res = waitUntilStateThenRelock(/*active*/ false, kShutdownTimeout);
+    if (res != OK) {
+        SET_ERR_L("Can't idle device in %f seconds!",
+                kShutdownTimeout/1e9);
     }
 
-    if (mStatus != STATUS_ERROR) {
-        mStatus = STATUS_IDLE;
-    }
+    return res;
+}
 
+// Resume after internalPauseAndWaitLocked
+status_t Camera3Device::internalResumeLocked() {
+    status_t res;
+
+    mRequestThread->setPaused(false);
+
+    res = waitUntilStateThenRelock(/*active*/ true, kActiveTimeout);
+    if (res != OK) {
+        SET_ERR_L("Can't transition to active in %f seconds!",
+                kActiveTimeout/1e9);
+    }
+    mPauseStateNotify = false;
     return OK;
 }
 
+status_t Camera3Device::waitUntilStateThenRelock(bool active,
+        nsecs_t timeout) {
+    status_t res = OK;
+    if (active == (mStatus == STATUS_ACTIVE)) {
+        // Desired state already reached
+        return res;
+    }
+
+    bool stateSeen = false;
+    do {
+        mRecentStatusUpdates.clear();
+
+        res = mStatusChanged.waitRelative(mLock, timeout);
+        if (res != OK) break;
+
+        // Check state change history during wait
+        for (size_t i = 0; i < mRecentStatusUpdates.size(); i++) {
+            if (active == (mRecentStatusUpdates[i] == STATUS_ACTIVE) ) {
+                stateSeen = true;
+                break;
+            }
+        }
+    } while (!stateSeen);
+
+    return res;
+}
+
+
 status_t Camera3Device::setNotifyCallback(NotificationListener *listener) {
     ATRACE_CALL();
     Mutex::Autolock l(mOutputLock);
@@ -859,7 +1018,6 @@
 }
 
 status_t Camera3Device::waitForNextFrame(nsecs_t timeout) {
-    ATRACE_CALL();
     status_t res;
     Mutex::Autolock l(mOutputLock);
 
@@ -893,6 +1051,7 @@
 
 status_t Camera3Device::triggerAutofocus(uint32_t id) {
     ATRACE_CALL();
+    Mutex::Autolock il(mInterfaceLock);
 
     ALOGV("%s: Triggering autofocus, id %d", __FUNCTION__, id);
     // Mix-in this trigger into the next request and only the next request.
@@ -913,6 +1072,7 @@
 
 status_t Camera3Device::triggerCancelAutofocus(uint32_t id) {
     ATRACE_CALL();
+    Mutex::Autolock il(mInterfaceLock);
 
     ALOGV("%s: Triggering cancel autofocus, id %d", __FUNCTION__, id);
     // Mix-in this trigger into the next request and only the next request.
@@ -933,6 +1093,7 @@
 
 status_t Camera3Device::triggerPrecaptureMetering(uint32_t id) {
     ATRACE_CALL();
+    Mutex::Autolock il(mInterfaceLock);
 
     ALOGV("%s: Triggering precapture metering, id %d", __FUNCTION__, id);
     // Mix-in this trigger into the next request and only the next request.
@@ -963,11 +1124,53 @@
 status_t Camera3Device::flush() {
     ATRACE_CALL();
     ALOGV("%s: Camera %d: Flushing all requests", __FUNCTION__, mId);
-
+    Mutex::Autolock il(mInterfaceLock);
     Mutex::Autolock l(mLock);
 
     mRequestThread->clear();
-    return mHal3Device->ops->flush(mHal3Device);
+    status_t res;
+    if (mHal3Device->common.version >= CAMERA_DEVICE_API_VERSION_3_1) {
+        res = mHal3Device->ops->flush(mHal3Device);
+    } else {
+        res = waitUntilDrained();
+    }
+
+    return res;
+}
+
+/**
+ * Methods called by subclasses
+ */
+
+void Camera3Device::notifyStatus(bool idle) {
+    {
+        // Need mLock to safely update state and synchronize to current
+        // state of methods in flight.
+        Mutex::Autolock l(mLock);
+        // We can get various system-idle notices from the status tracker
+        // while starting up. Only care about them if we've actually sent
+        // in some requests recently.
+        if (mStatus != STATUS_ACTIVE && mStatus != STATUS_CONFIGURED) {
+            return;
+        }
+        ALOGV("%s: Camera %d: Now %s", __FUNCTION__, mId,
+                idle ? "idle" : "active");
+        mStatus = idle ? STATUS_CONFIGURED : STATUS_ACTIVE;
+        mRecentStatusUpdates.add(mStatus);
+        mStatusChanged.signal();
+
+        // Skip notifying listener if we're doing some user-transparent
+        // state changes
+        if (mPauseStateNotify) return;
+    }
+    NotificationListener *listener;
+    {
+        Mutex::Autolock l(mOutputLock);
+        listener = mListener;
+    }
+    if (idle && listener != NULL) {
+        listener->notifyIdle();
+    }
 }
 
 /**
@@ -1046,18 +1249,18 @@
     ATRACE_CALL();
     status_t res;
 
-    if (mStatus != STATUS_IDLE) {
+    if (mStatus != STATUS_UNCONFIGURED && mStatus != STATUS_CONFIGURED) {
         CLOGE("Not idle");
         return INVALID_OPERATION;
     }
 
     if (!mNeedConfig) {
         ALOGV("%s: Skipping config, no stream changes", __FUNCTION__);
-        mStatus = STATUS_ACTIVE;
         return OK;
     }
 
     // Start configuring the streams
+    ALOGV("%s: Camera %d: Starting stream configuration", __FUNCTION__, mId);
 
     camera3_stream_configuration config;
 
@@ -1139,11 +1342,18 @@
     // across configure_streams() calls
     mRequestThread->configurationComplete();
 
-    // Finish configuring the streams lazily on first reference
+    // Update device state
 
-    mStatus = STATUS_ACTIVE;
     mNeedConfig = false;
 
+    if (config.num_streams > 0) {
+        mStatus = STATUS_CONFIGURED;
+    } else {
+        mStatus = STATUS_UNCONFIGURED;
+    }
+
+    ALOGV("%s: Camera %d: Stream configuration complete", __FUNCTION__, mId);
+
     return OK;
 }
 
@@ -1179,6 +1389,10 @@
     // But only do error state transition steps for the first error
     if (mStatus == STATUS_ERROR || mStatus == STATUS_UNINITIALIZED) return;
 
+    // Save stack trace. View by dumping it later.
+    CameraTraces::saveTrace();
+    // TODO: consider adding errorCause and client pid/procname
+
     mErrorCause = errorCause;
 
     mRequestThread->setPaused(true);
@@ -1190,18 +1404,187 @@
  */
 
 status_t Camera3Device::registerInFlight(int32_t frameNumber,
-        int32_t numBuffers) {
+        int32_t requestId, int32_t numBuffers) {
     ATRACE_CALL();
     Mutex::Autolock l(mInFlightLock);
 
     ssize_t res;
-    res = mInFlightMap.add(frameNumber, InFlightRequest(numBuffers));
+    res = mInFlightMap.add(frameNumber, InFlightRequest(requestId, numBuffers));
     if (res < 0) return res;
 
     return OK;
 }
 
 /**
+ * QUIRK(partial results)
+ * Check if all 3A fields are ready, and send off a partial 3A-only result
+ * to the output frame queue
+ */
+bool Camera3Device::processPartial3AQuirk(
+        int32_t frameNumber, int32_t requestId,
+        const CameraMetadata& partial) {
+
+    // Check if all 3A states are present
+    // The full list of fields is
+    //   android.control.afMode
+    //   android.control.awbMode
+    //   android.control.aeState
+    //   android.control.awbState
+    //   android.control.afState
+    //   android.control.afTriggerID
+    //   android.control.aePrecaptureID
+    // TODO: Add android.control.aeMode
+
+    bool gotAllStates = true;
+
+    uint8_t afMode;
+    uint8_t awbMode;
+    uint8_t aeState;
+    uint8_t afState;
+    uint8_t awbState;
+    int32_t afTriggerId;
+    int32_t aeTriggerId;
+
+    gotAllStates &= get3AResult(partial, ANDROID_CONTROL_AF_MODE,
+        &afMode, frameNumber);
+
+    gotAllStates &= get3AResult(partial, ANDROID_CONTROL_AWB_MODE,
+        &awbMode, frameNumber);
+
+    gotAllStates &= get3AResult(partial, ANDROID_CONTROL_AE_STATE,
+        &aeState, frameNumber);
+
+    gotAllStates &= get3AResult(partial, ANDROID_CONTROL_AF_STATE,
+        &afState, frameNumber);
+
+    gotAllStates &= get3AResult(partial, ANDROID_CONTROL_AWB_STATE,
+        &awbState, frameNumber);
+
+    gotAllStates &= get3AResult(partial, ANDROID_CONTROL_AF_TRIGGER_ID,
+        &afTriggerId, frameNumber);
+
+    gotAllStates &= get3AResult(partial, ANDROID_CONTROL_AE_PRECAPTURE_ID,
+        &aeTriggerId, frameNumber);
+
+    if (!gotAllStates) return false;
+
+    ALOGVV("%s: Camera %d: Frame %d, Request ID %d: AF mode %d, AWB mode %d, "
+        "AF state %d, AE state %d, AWB state %d, "
+        "AF trigger %d, AE precapture trigger %d",
+        __FUNCTION__, mId, frameNumber, requestId,
+        afMode, awbMode,
+        afState, aeState, awbState,
+        afTriggerId, aeTriggerId);
+
+    // Got all states, so construct a minimal result to send
+    // In addition to the above fields, this means adding in
+    //   android.request.frameCount
+    //   android.request.requestId
+    //   android.quirks.partialResult
+
+    const size_t kMinimal3AResultEntries = 10;
+
+    Mutex::Autolock l(mOutputLock);
+
+    CameraMetadata& min3AResult =
+            *mResultQueue.insert(
+                mResultQueue.end(),
+                CameraMetadata(kMinimal3AResultEntries, /*dataCapacity*/ 0));
+
+    if (!insert3AResult(min3AResult, ANDROID_REQUEST_FRAME_COUNT,
+            &frameNumber, frameNumber)) {
+        return false;
+    }
+
+    if (!insert3AResult(min3AResult, ANDROID_REQUEST_ID,
+            &requestId, frameNumber)) {
+        return false;
+    }
+
+    static const uint8_t partialResult = ANDROID_QUIRKS_PARTIAL_RESULT_PARTIAL;
+    if (!insert3AResult(min3AResult, ANDROID_QUIRKS_PARTIAL_RESULT,
+            &partialResult, frameNumber)) {
+        return false;
+    }
+
+    if (!insert3AResult(min3AResult, ANDROID_CONTROL_AF_MODE,
+            &afMode, frameNumber)) {
+        return false;
+    }
+
+    if (!insert3AResult(min3AResult, ANDROID_CONTROL_AWB_MODE,
+            &awbMode, frameNumber)) {
+        return false;
+    }
+
+    if (!insert3AResult(min3AResult, ANDROID_CONTROL_AE_STATE,
+            &aeState, frameNumber)) {
+        return false;
+    }
+
+    if (!insert3AResult(min3AResult, ANDROID_CONTROL_AF_STATE,
+            &afState, frameNumber)) {
+        return false;
+    }
+
+    if (!insert3AResult(min3AResult, ANDROID_CONTROL_AWB_STATE,
+            &awbState, frameNumber)) {
+        return false;
+    }
+
+    if (!insert3AResult(min3AResult, ANDROID_CONTROL_AF_TRIGGER_ID,
+            &afTriggerId, frameNumber)) {
+        return false;
+    }
+
+    if (!insert3AResult(min3AResult, ANDROID_CONTROL_AE_PRECAPTURE_ID,
+            &aeTriggerId, frameNumber)) {
+        return false;
+    }
+
+    mResultSignal.signal();
+
+    return true;
+}
+
+template<typename T>
+bool Camera3Device::get3AResult(const CameraMetadata& result, int32_t tag,
+        T* value, int32_t frameNumber) {
+    (void) frameNumber;
+
+    camera_metadata_ro_entry_t entry;
+
+    entry = result.find(tag);
+    if (entry.count == 0) {
+        ALOGVV("%s: Camera %d: Frame %d: No %s provided by HAL!", __FUNCTION__,
+            mId, frameNumber, get_camera_metadata_tag_name(tag));
+        return false;
+    }
+
+    if (sizeof(T) == sizeof(uint8_t)) {
+        *value = entry.data.u8[0];
+    } else if (sizeof(T) == sizeof(int32_t)) {
+        *value = entry.data.i32[0];
+    } else {
+        ALOGE("%s: Unexpected type", __FUNCTION__);
+        return false;
+    }
+    return true;
+}
+
+template<typename T>
+bool Camera3Device::insert3AResult(CameraMetadata& result, int32_t tag,
+        const T* value, int32_t frameNumber) {
+    if (result.update(tag, value, 1) != NO_ERROR) {
+        mResultQueue.erase(--mResultQueue.end(), mResultQueue.end());
+        SET_ERR("Frame %d: Failed to set %s in partial metadata",
+                frameNumber, get_camera_metadata_tag_name(tag));
+        return false;
+    }
+    return true;
+}
+
+/**
  * Camera HAL device callback methods
  */
 
@@ -1216,6 +1599,8 @@
                 frameNumber);
         return;
     }
+    bool partialResultQuirk = false;
+    CameraMetadata collectedQuirkResult;
 
     // Get capture timestamp from list of in-flight requests, where it was added
     // by the shutter notification for this frame. Then update the in-flight
@@ -1231,19 +1616,58 @@
             return;
         }
         InFlightRequest &request = mInFlightMap.editValueAt(idx);
+
+        // Check if this result carries only partial metadata
+        if (mUsePartialResultQuirk && result->result != NULL) {
+            camera_metadata_ro_entry_t partialResultEntry;
+            res = find_camera_metadata_ro_entry(result->result,
+                    ANDROID_QUIRKS_PARTIAL_RESULT, &partialResultEntry);
+            if (res != NAME_NOT_FOUND &&
+                    partialResultEntry.count > 0 &&
+                    partialResultEntry.data.u8[0] ==
+                    ANDROID_QUIRKS_PARTIAL_RESULT_PARTIAL) {
+                // A partial result. Flag this as such, and collect this
+                // set of metadata into the in-flight entry.
+                partialResultQuirk = true;
+                request.partialResultQuirk.collectedResult.append(
+                    result->result);
+                request.partialResultQuirk.collectedResult.erase(
+                    ANDROID_QUIRKS_PARTIAL_RESULT);
+                // Fire off a 3A-only result if possible
+                if (!request.partialResultQuirk.haveSent3A) {
+                    request.partialResultQuirk.haveSent3A =
+                            processPartial3AQuirk(frameNumber,
+                                    request.requestId,
+                                    request.partialResultQuirk.collectedResult);
+                }
+            }
+        }
+
         timestamp = request.captureTimestamp;
-        if (timestamp == 0) {
+        /**
+         * One of the following must happen before it's legal to call process_capture_result,
+         * unless partial metadata is being provided:
+         * - CAMERA3_MSG_SHUTTER (expected during normal operation)
+         * - CAMERA3_MSG_ERROR (expected during flush)
+         */
+        if (request.requestStatus == OK && timestamp == 0 && !partialResultQuirk) {
             SET_ERR("Called before shutter notify for frame %d",
                     frameNumber);
             return;
         }
 
-        if (result->result != NULL) {
+        // Did we get the (final) result metadata for this capture?
+        if (result->result != NULL && !partialResultQuirk) {
             if (request.haveResultMetadata) {
                 SET_ERR("Called multiple times with metadata for frame %d",
                         frameNumber);
                 return;
             }
+            if (mUsePartialResultQuirk &&
+                    !request.partialResultQuirk.collectedResult.isEmpty()) {
+                collectedQuirkResult.acquire(
+                    request.partialResultQuirk.collectedResult);
+            }
             request.haveResultMetadata = true;
         }
 
@@ -1255,7 +1679,10 @@
             return;
         }
 
-        if (request.haveResultMetadata && request.numBuffersLeft == 0) {
+        // Check if everything has arrived for this result (buffers and metadata), remove it from
+        // InFlightMap if both arrived or HAL reports error for this request (i.e. during flush).
+        if ((request.requestStatus != OK) ||
+                (request.haveResultMetadata && request.numBuffersLeft == 0)) {
             ATRACE_ASYNC_END("frame capture", frameNumber);
             mInFlightMap.removeItemsAt(idx, 1);
         }
@@ -1269,9 +1696,12 @@
     }
 
     // Process the result metadata, if provided
-    if (result->result != NULL) {
+    bool gotResult = false;
+    if (result->result != NULL && !partialResultQuirk) {
         Mutex::Autolock l(mOutputLock);
 
+        gotResult = true;
+
         if (frameNumber != mNextResultFrameNumber) {
             SET_ERR("Out-of-order capture result metadata submitted! "
                     "(got frame number %d, expecting %d)",
@@ -1280,19 +1710,26 @@
         }
         mNextResultFrameNumber++;
 
-        CameraMetadata &captureResult =
-                *mResultQueue.insert(mResultQueue.end(), CameraMetadata());
-
+        CameraMetadata captureResult;
         captureResult = result->result;
+
         if (captureResult.update(ANDROID_REQUEST_FRAME_COUNT,
                         (int32_t*)&frameNumber, 1) != OK) {
             SET_ERR("Failed to set frame# in metadata (%d)",
                     frameNumber);
+            gotResult = false;
         } else {
             ALOGVV("%s: Camera %d: Set frame# in metadata (%d)",
                     __FUNCTION__, mId, frameNumber);
         }
 
+        // Append any previous partials to form a complete result
+        if (mUsePartialResultQuirk && !collectedQuirkResult.isEmpty()) {
+            captureResult.append(collectedQuirkResult);
+        }
+
+        captureResult.sort();
+
         // Check that there's a timestamp in the result metadata
 
         camera_metadata_entry entry =
@@ -1300,10 +1737,19 @@
         if (entry.count == 0) {
             SET_ERR("No timestamp provided by HAL for frame %d!",
                     frameNumber);
+            gotResult = false;
         } else if (timestamp != entry.data.i64[0]) {
             SET_ERR("Timestamp mismatch between shutter notify and result"
                     " metadata for frame %d (%lld vs %lld respectively)",
                     frameNumber, timestamp, entry.data.i64[0]);
+            gotResult = false;
+        }
+
+        if (gotResult) {
+            // Valid result, insert into queue
+            CameraMetadata& queuedResult =
+                *mResultQueue.insert(mResultQueue.end(), CameraMetadata());
+            queuedResult.swap(captureResult);
         }
     } // scope for mOutputLock
 
@@ -1316,14 +1762,14 @@
         // Note: stream may be deallocated at this point, if this buffer was the
         // last reference to it.
         if (res != OK) {
-            SET_ERR("Can't return buffer %d for frame %d to its stream: "
+            ALOGE("Can't return buffer %d for frame %d to its stream: "
                     " %s (%d)", i, frameNumber, strerror(-res), res);
         }
     }
 
     // Finally, signal any waiters for new frames
 
-    if (result->result != NULL) {
+    if (gotResult) {
         mResultSignal.signal();
     }
 
@@ -1356,6 +1802,16 @@
             ALOGV("Camera %d: %s: HAL error, frame %d, stream %d: %d",
                     mId, __FUNCTION__, msg->message.error.frame_number,
                     streamId, msg->message.error.error_code);
+
+            // Set request error status for the request in the in-flight tracking
+            {
+                Mutex::Autolock l(mInFlightLock);
+                ssize_t idx = mInFlightMap.indexOfKey(msg->message.error.frame_number);
+                if (idx >= 0) {
+                    mInFlightMap.editValueAt(idx).requestStatus = msg->message.error.error_code;
+                }
+            }
+
             if (listener != NULL) {
                 listener->notifyError(msg->message.error.error_code,
                         msg->message.error.frame_number, streamId);
@@ -1378,12 +1834,17 @@
                 mNextShutterFrameNumber++;
             }
 
+            int32_t requestId = -1;
+
             // Set timestamp for the request in the in-flight tracking
+            // and get the request ID to send upstream
             {
                 Mutex::Autolock l(mInFlightLock);
                 idx = mInFlightMap.indexOfKey(frameNumber);
                 if (idx >= 0) {
-                    mInFlightMap.editValueAt(idx).captureTimestamp = timestamp;
+                    InFlightRequest &r = mInFlightMap.editValueAt(idx);
+                    r.captureTimestamp = timestamp;
+                    requestId = r.requestId;
                 }
             }
             if (idx < 0) {
@@ -1391,11 +1852,11 @@
                         frameNumber);
                 break;
             }
-            ALOGVV("Camera %d: %s: Shutter fired for frame %d at %lld",
-                    mId, __FUNCTION__, frameNumber, timestamp);
+            ALOGVV("Camera %d: %s: Shutter fired for frame %d (id %d) at %lld",
+                    mId, __FUNCTION__, frameNumber, requestId, timestamp);
             // Call listener, if any
             if (listener != NULL) {
-                listener->notifyShutter(frameNumber, timestamp);
+                listener->notifyShutter(requestId, timestamp);
             }
             break;
         }
@@ -1405,40 +1866,15 @@
     }
 }
 
-CameraMetadata Camera3Device::getLatestRequest() {
+CameraMetadata Camera3Device::getLatestRequestLocked() {
     ALOGV("%s", __FUNCTION__);
 
-    bool locked = false;
-
-    /**
-     * Why trylock instead of autolock?
-     *
-     * We want to be able to call this function from
-     * dumpsys, which often happens during deadlocks.
-     */
-    for (size_t i = 0; i < kDumpLockAttempts; ++i) {
-        if (mLock.tryLock() == NO_ERROR) {
-            locked = true;
-            break;
-        } else {
-            usleep(kDumpSleepDuration);
-        }
-    }
-
-    if (!locked) {
-        ALOGW("%s: Possible deadlock detected", __FUNCTION__);
-    }
-
     CameraMetadata retVal;
 
     if (mRequestThread != NULL) {
         retVal = mRequestThread->getLatestRequest();
     }
 
-    if (locked) {
-        mLock.unlock();
-    }
-
     return retVal;
 }
 
@@ -1447,9 +1883,11 @@
  */
 
 Camera3Device::RequestThread::RequestThread(wp<Camera3Device> parent,
+        sp<StatusTracker> statusTracker,
         camera3_device_t *hal3Device) :
         Thread(false),
         mParent(parent),
+        mStatusTracker(statusTracker),
         mHal3Device(hal3Device),
         mId(getId(parent)),
         mReconfigured(false),
@@ -1457,6 +1895,7 @@
         mPaused(true),
         mFrameNumber(0),
         mLatestRequestId(NAME_NOT_FOUND) {
+    mStatusId = statusTracker->addComponent();
 }
 
 void Camera3Device::RequestThread::configurationComplete() {
@@ -1562,19 +2001,6 @@
     mDoPauseSignal.signal();
 }
 
-status_t Camera3Device::RequestThread::waitUntilPaused(nsecs_t timeout) {
-    ATRACE_CALL();
-    status_t res;
-    Mutex::Autolock l(mPauseLock);
-    while (!mPaused) {
-        res = mPausedSignal.waitRelative(mPauseLock, timeout);
-        if (res == TIMED_OUT) {
-            return res;
-        }
-    }
-    return OK;
-}
-
 status_t Camera3Device::RequestThread::waitUntilRequestProcessed(
         int32_t requestId, nsecs_t timeout) {
     Mutex::Autolock l(mLatestRequestMutex);
@@ -1591,7 +2017,13 @@
     return OK;
 }
 
-
+void Camera3Device::RequestThread::requestExit() {
+    // Call parent to set up shutdown
+    Thread::requestExit();
+    // The exit from any possible waits
+    mDoPauseSignal.signal();
+    mRequestSignal.signal();
+}
 
 bool Camera3Device::RequestThread::threadLoop() {
 
@@ -1613,6 +2045,18 @@
     camera3_capture_request_t request = camera3_capture_request_t();
     Vector<camera3_stream_buffer_t> outputBuffers;
 
+    // Get the request ID, if any
+    int requestId;
+    camera_metadata_entry_t requestIdEntry =
+            nextRequest->mSettings.find(ANDROID_REQUEST_ID);
+    if (requestIdEntry.count > 0) {
+        requestId = requestIdEntry.data.i32[0];
+    } else {
+        ALOGW("%s: Did not have android.request.id set in the request",
+                __FUNCTION__);
+        requestId = NAME_NOT_FOUND;
+    }
+
     // Insert any queued triggers (before metadata is locked)
     int32_t triggerCount;
     res = insertTriggers(nextRequest);
@@ -1679,7 +2123,7 @@
         request.input_buffer = &inputBuffer;
         res = nextRequest->mInputStream->getInputBuffer(&inputBuffer);
         if (res != OK) {
-            SET_ERR("RequestThread: Can't get input buffer, skipping request:"
+            ALOGE("RequestThread: Can't get input buffer, skipping request:"
                     " %s (%d)", strerror(-res), res);
             cleanUpFailedRequest(request, nextRequest, outputBuffers);
             return true;
@@ -1695,8 +2139,8 @@
         res = nextRequest->mOutputStreams.editItemAt(i)->
                 getBuffer(&outputBuffers.editItemAt(i));
         if (res != OK) {
-            SET_ERR("RequestThread: Can't get output buffer, skipping request:"
-                    "%s (%d)", strerror(-res), res);
+            ALOGE("RequestThread: Can't get output buffer, skipping request:"
+                    " %s (%d)", strerror(-res), res);
             cleanUpFailedRequest(request, nextRequest, outputBuffers);
             return true;
         }
@@ -1713,7 +2157,7 @@
         return false;
     }
 
-    res = parent->registerInFlight(request.frame_number,
+    res = parent->registerInFlight(request.frame_number, requestId,
             request.num_output_buffers);
     if (res != OK) {
         SET_ERR("RequestThread: Unable to register new in-flight request:"
@@ -1722,6 +2166,14 @@
         return false;
     }
 
+    // Inform waitUntilRequestProcessed thread of a new request ID
+    {
+        Mutex::Autolock al(mLatestRequestMutex);
+
+        mLatestRequestId = requestId;
+        mLatestRequestSignal.signal();
+    }
+
     // Submit request and block until ready for next one
     ATRACE_ASYNC_BEGIN("frame capture", request.frame_number);
     ATRACE_BEGIN("camera3->process_capture_request");
@@ -1757,24 +2209,6 @@
     }
     mPrevTriggers = triggerCount;
 
-    // Read android.request.id from the request settings metadata
-    // - inform waitUntilRequestProcessed thread of a new request ID
-    {
-        Mutex::Autolock al(mLatestRequestMutex);
-
-        camera_metadata_entry_t requestIdEntry =
-                nextRequest->mSettings.find(ANDROID_REQUEST_ID);
-        if (requestIdEntry.count > 0) {
-            mLatestRequestId = requestIdEntry.data.i32[0];
-        } else {
-            ALOGW("%s: Did not have android.request.id set in the request",
-                  __FUNCTION__);
-            mLatestRequestId = NAME_NOT_FOUND;
-        }
-
-        mLatestRequestSignal.signal();
-    }
-
     // Return input buffer back to framework
     if (request.input_buffer != NULL) {
         Camera3Stream *stream =
@@ -1790,8 +2224,6 @@
         }
     }
 
-
-
     return true;
 }
 
@@ -1849,12 +2281,17 @@
 
         res = mRequestSignal.waitRelative(mRequestLock, kRequestTimeout);
 
-        if (res == TIMED_OUT) {
-            // Signal that we're paused by starvation
+        if ((mRequestQueue.empty() && mRepeatingRequests.empty()) ||
+                exitPending()) {
             Mutex::Autolock pl(mPauseLock);
             if (mPaused == false) {
+                ALOGV("%s: RequestThread: Going idle", __FUNCTION__);
                 mPaused = true;
-                mPausedSignal.signal();
+                // Let the tracker know
+                sp<StatusTracker> statusTracker = mStatusTracker.promote();
+                if (statusTracker != 0) {
+                    statusTracker->markComponentIdle(mStatusId, Fence::NO_FENCE);
+                }
             }
             // Stop waiting for now and let thread management happen
             return NULL;
@@ -1874,6 +2311,13 @@
     // update internal pause state (capture/setRepeatingRequest unpause
     // directly).
     Mutex::Autolock pl(mPauseLock);
+    if (mPaused) {
+        ALOGV("%s: RequestThread: Unpaused", __FUNCTION__);
+        sp<StatusTracker> statusTracker = mStatusTracker.promote();
+        if (statusTracker != 0) {
+            statusTracker->markComponentActive(mStatusId);
+        }
+    }
     mPaused = false;
 
     // Check if we've reconfigured since last time, and reset the preview
@@ -1890,13 +2334,18 @@
     status_t res;
     Mutex::Autolock l(mPauseLock);
     while (mDoPause) {
-        // Signal that we're paused by request
         if (mPaused == false) {
             mPaused = true;
-            mPausedSignal.signal();
+            ALOGV("%s: RequestThread: Paused", __FUNCTION__);
+            // Let the tracker know
+            sp<StatusTracker> statusTracker = mStatusTracker.promote();
+            if (statusTracker != 0) {
+                statusTracker->markComponentIdle(mStatusId, Fence::NO_FENCE);
+            }
         }
+
         res = mDoPauseSignal.waitRelative(mPauseLock, kRequestTimeout);
-        if (res == TIMED_OUT) {
+        if (res == TIMED_OUT || exitPending()) {
             return true;
         }
     }
@@ -1909,8 +2358,16 @@
     // With work to do, mark thread as unpaused.
     // If paused by request (setPaused), don't resume, to avoid
     // extra signaling/waiting overhead to waitUntilPaused
+    mRequestSignal.signal();
     Mutex::Autolock p(mPauseLock);
     if (!mDoPause) {
+        ALOGV("%s: RequestThread: Going active", __FUNCTION__);
+        if (mPaused) {
+            sp<StatusTracker> statusTracker = mStatusTracker.promote();
+            if (statusTracker != 0) {
+                statusTracker->markComponentActive(mStatusId);
+            }
+        }
         mPaused = false;
     }
 }

services/camera/libcameraservice/device3/Camera3Device.h

diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index 61caf13..468f641 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -26,6 +26,7 @@
 #include <hardware/camera3.h>
 
 #include "common/CameraDeviceBase.h"
+#include "device3/StatusTracker.h"
 
 /**
  * Function pointer types with C calling convention to
@@ -126,29 +127,47 @@
 
     virtual status_t flush();
 
+    // Methods called by subclasses
+    void             notifyStatus(bool idle); // updates from StatusTracker
+
   private:
     static const size_t        kDumpLockAttempts  = 10;
     static const size_t        kDumpSleepDuration = 100000; // 0.10 sec
     static const size_t        kInFlightWarnLimit = 20;
     static const nsecs_t       kShutdownTimeout   = 5000000000; // 5 sec
+    static const nsecs_t       kActiveTimeout     = 500000000;  // 500 ms
     struct                     RequestTrigger;
 
+    // A lock to enforce serialization on the input/configure side
+    // of the public interface.
+    // Only locked by public methods inherited from CameraDeviceBase.
+    // Not locked by methods guarded by mOutputLock, since they may act
+    // concurrently to the input/configure side of the interface.
+    // Must be locked before mLock if both will be locked by a method
+    Mutex                      mInterfaceLock;
+
+    // The main lock on internal state
     Mutex                      mLock;
 
+    // Camera device ID
+    const int                  mId;
+
     /**** Scope for mLock ****/
 
-    const int                  mId;
     camera3_device_t          *mHal3Device;
 
     CameraMetadata             mDeviceInfo;
     vendor_tag_query_ops_t     mVendorTagOps;
 
-    enum {
+    enum Status {
         STATUS_ERROR,
         STATUS_UNINITIALIZED,
-        STATUS_IDLE,
+        STATUS_UNCONFIGURED,
+        STATUS_CONFIGURED,
         STATUS_ACTIVE
     }                          mStatus;
+    Vector<Status>             mRecentStatusUpdates;
+    Condition                  mStatusChanged;
 
     // Tracking cause of fatal errors when in STATUS_ERROR
     String8                    mErrorCause;
@@ -162,9 +181,16 @@
     int                        mNextStreamId;
     bool                       mNeedConfig;
 
+    // Whether to send state updates upstream
+    // Pause when doing transparent reconfiguration
+    bool                       mPauseStateNotify;
+
     // Need to hold on to stream references until configure completes.
     Vector<sp<camera3::Camera3StreamInterface> > mDeletedStreams;
 
+    // Whether quirk ANDROID_QUIRKS_USE_PARTIAL_RESULT is enabled
+    bool                       mUsePartialResultQuirk;
+
     /**** End scope for mLock ****/
 
     class CaptureRequest : public LightRefBase<CaptureRequest> {
@@ -181,13 +207,34 @@
      *
      * Takes mLock.
      */
-    virtual CameraMetadata getLatestRequest();
+    virtual CameraMetadata getLatestRequestLocked();
 
     /**
-     * Lock-held version of waitUntilDrained. Will transition to IDLE on
-     * success.
+     * Pause processing and flush everything, but don't tell the clients.
+     * This is for reconfiguring outputs transparently when according to the
+     * CameraDeviceBase interface we shouldn't need to.
+     * Must be called with mLock and mInterfaceLock both held.
      */
-    status_t           waitUntilDrainedLocked();
+    status_t internalPauseAndWaitLocked();
+
+    /**
+     * Resume work after internalPauseAndWaitLocked()
+     * Must be called with mLock and mInterfaceLock both held.
+     */
+    status_t internalResumeLocked();
+
+    /**
+     * Wait until status tracker tells us we've transitioned to the target state
+     * set, which is either ACTIVE when active==true or IDLE (which is any
+     * non-ACTIVE state) when active==false.
+     *
+     * Needs to be called with mLock and mInterfaceLock held.  This means there
+     * can ever only be one waiter at most.
+     *
+     * During the wait mLock is released.
+     *
+     */
+    status_t waitUntilStateThenRelock(bool active, nsecs_t timeout);
 
     /**
      * Do common work for setting up a streaming or single capture request.
@@ -217,6 +264,12 @@
     void               setErrorStateLocked(const char *fmt, ...);
     void               setErrorStateLockedV(const char *fmt, va_list args);
 
+    /**
+     * Debugging trylock/spin method
+     * Try to acquire a lock a few times with sleeps between before giving up.
+     */
+    bool               tryLockSpinRightRound(Mutex& lock);
+
     struct RequestTrigger {
         // Metadata tag number, e.g. android.control.aePrecaptureTrigger
         uint32_t metadataTag;
@@ -242,6 +295,7 @@
       public:
 
         RequestThread(wp<Camera3Device> parent,
+                sp<camera3::StatusTracker> statusTracker,
                 camera3_device_t *hal3Device);
 
         /**
@@ -279,13 +333,6 @@
         void     setPaused(bool paused);
 
         /**
-         * Wait until thread is paused, either due to setPaused(true)
-         * or due to lack of input requests. Returns TIMED_OUT in case
-         * the thread does not pause within the timeout.
-         */
-        status_t waitUntilPaused(nsecs_t timeout);
-
-        /**
          * Wait until thread processes the capture request with settings'
          * android.request.id == requestId.
          *
@@ -295,6 +342,12 @@
         status_t waitUntilRequestProcessed(int32_t requestId, nsecs_t timeout);
 
         /**
+         * Shut down the thread. Shutdown is asynchronous, so thread may
+         * still be running once this method returns.
+         */
+        virtual void requestExit();
+
+        /**
          * Get the latest request that was sent to the HAL
          * with process_capture_request.
          */
@@ -339,9 +392,12 @@
         void               setErrorState(const char *fmt, ...);
 
         wp<Camera3Device>  mParent;
+        wp<camera3::StatusTracker>  mStatusTracker;
         camera3_device_t  *mHal3Device;
 
-        const int          mId;
+        const int          mId;       // The camera ID
+        int                mStatusId; // The RequestThread's component ID for
+                                      // status tracking
 
         Mutex              mRequestLock;
         Condition          mRequestSignal;
@@ -381,22 +437,42 @@
      */
 
     struct InFlightRequest {
+        // android.request.id for the request
+        int     requestId;
         // Set by notify() SHUTTER call.
         nsecs_t captureTimestamp;
+        int     requestStatus;
         // Set by process_capture_result call with valid metadata
         bool    haveResultMetadata;
         // Decremented by calls to process_capture_result with valid output
         // buffers
         int     numBuffersLeft;
 
+        // Fields used by the partial result quirk only
+        struct PartialResultQuirkInFlight {
+            // Set by process_capture_result once 3A has been sent to clients
+            bool    haveSent3A;
+            // Result metadata collected so far, when partial results are in use
+            CameraMetadata collectedResult;
+
+            PartialResultQuirkInFlight():
+                    haveSent3A(false) {
+            }
+        } partialResultQuirk;
+
+        // Default constructor needed by KeyedVector
         InFlightRequest() :
+                requestId(0),
                 captureTimestamp(0),
+                requestStatus(OK),
                 haveResultMetadata(false),
                 numBuffersLeft(0) {
         }
 
-        explicit InFlightRequest(int numBuffers) :
+        InFlightRequest(int id, int numBuffers) :
+                requestId(id),
                 captureTimestamp(0),
+                requestStatus(OK),
                 haveResultMetadata(false),
                 numBuffersLeft(numBuffers) {
         }
@@ -407,7 +483,29 @@
     Mutex                  mInFlightLock; // Protects mInFlightMap
     InFlightMap            mInFlightMap;
 
-    status_t registerInFlight(int32_t frameNumber, int32_t numBuffers);
+    status_t registerInFlight(int32_t frameNumber, int32_t requestId,
+            int32_t numBuffers);
+
+    /**
+     * For the partial result quirk, check if all 3A state fields are available
+     * and if so, queue up 3A-only result to the client. Returns true if 3A
+     * is sent.
+     */
+    bool processPartial3AQuirk(int32_t frameNumber, int32_t requestId,
+            const CameraMetadata& partial);
+
+    // Helpers for reading and writing 3A metadata into to/from partial results
+    template<typename T>
+    bool get3AResult(const CameraMetadata& result, int32_t tag,
+            T* value, int32_t frameNumber);
+
+    template<typename T>
+    bool insert3AResult(CameraMetadata &result, int32_t tag, const T* value,
+            int32_t frameNumber);
+    /**
+     * Tracking for idle detection
+     */
+    sp<camera3::StatusTracker> mStatusTracker;
 
     /**
      * Output result queue and current HAL device 3A state

services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp

diff --git a/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp b/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp
index 0850566..42e02d8 100644
--- a/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp
+++ b/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp
@@ -23,7 +23,8 @@
 
 #include <utils/Log.h>
 #include <utils/Trace.h>
-#include "Camera3IOStreamBase.h"
+#include "device3/Camera3IOStreamBase.h"
+#include "device3/StatusTracker.h"
 
 namespace android {
 
@@ -62,53 +63,6 @@
     return false;
 }
 
-status_t Camera3IOStreamBase::waitUntilIdle(nsecs_t timeout) {
-    status_t res;
-    {
-        Mutex::Autolock l(mLock);
-        while (mDequeuedBufferCount > 0) {
-            if (timeout != TIMEOUT_NEVER) {
-                nsecs_t startTime = systemTime();
-                res = mBufferReturnedSignal.waitRelative(mLock, timeout);
-                if (res == TIMED_OUT) {
-                    return res;
-                } else if (res != OK) {
-                    ALOGE("%s: Error waiting for outstanding buffers: %s (%d)",
-                            __FUNCTION__, strerror(-res), res);
-                    return res;
-                }
-                nsecs_t deltaTime = systemTime() - startTime;
-                if (timeout <= deltaTime) {
-                    timeout = 0;
-                } else {
-                    timeout -= deltaTime;
-                }
-            } else {
-                res = mBufferReturnedSignal.wait(mLock);
-                if (res != OK) {
-                    ALOGE("%s: Error waiting for outstanding buffers: %s (%d)",
-                            __FUNCTION__, strerror(-res), res);
-                    return res;
-                }
-            }
-        }
-    }
-
-    // No lock
-
-    unsigned int timeoutMs;
-    if (timeout == TIMEOUT_NEVER) {
-        timeoutMs = Fence::TIMEOUT_NEVER;
-    } else if (timeout == 0) {
-        timeoutMs = 0;
-    } else {
-        // Round up to wait at least 1 ms
-        timeoutMs = (timeout + 999999) / 1000000;
-    }
-
-    return mCombinedFence->wait(timeoutMs);
-}
-
 void Camera3IOStreamBase::dump(int fd, const Vector<String16> &args) const {
     (void) args;
     String8 lines;
@@ -116,12 +70,12 @@
     lines.appendFormat("      Dims: %d x %d, format 0x%x\n",
             camera3_stream::width, camera3_stream::height,
             camera3_stream::format);
-    lines.appendFormat("      Max size: %d\n", mMaxSize);
+    lines.appendFormat("      Max size: %zu\n", mMaxSize);
     lines.appendFormat("      Usage: %d, max HAL buffers: %d\n",
             camera3_stream::usage, camera3_stream::max_buffers);
     lines.appendFormat("      Frames produced: %d, last timestamp: %lld ns\n",
             mFrameCount, mLastTimestamp);
-    lines.appendFormat("      Total buffers: %d, currently dequeued: %d\n",
+    lines.appendFormat("      Total buffers: %zu, currently dequeued: %zu\n",
             mTotalBufferCount, mDequeuedBufferCount);
     write(fd, lines.string(), lines.size());
 }
@@ -190,6 +144,14 @@
     buffer.release_fence = releaseFence;
     buffer.status = status;
 
+    // Inform tracker about becoming busy
+    if (mDequeuedBufferCount == 0 && mState != STATE_IN_CONFIG &&
+            mState != STATE_IN_RECONFIG) {
+        sp<StatusTracker> statusTracker = mStatusTracker.promote();
+        if (statusTracker != 0) {
+            statusTracker->markComponentActive(mStatusId);
+        }
+    }
     mDequeuedBufferCount++;
 }
 
@@ -252,20 +214,32 @@
     sp<Fence> releaseFence;
     res = returnBufferCheckedLocked(buffer, timestamp, output,
                                     &releaseFence);
-    if (res != OK) {
-        return res;
+    // Res may be an error, but we still want to decrement our owned count
+    // to enable clean shutdown. So we'll just return the error but otherwise
+    // carry on
+
+    if (releaseFence != 0) {
+        mCombinedFence = Fence::merge(mName, mCombinedFence, releaseFence);
     }
 
-    mCombinedFence = Fence::merge(mName, mCombinedFence, releaseFence);
-
     mDequeuedBufferCount--;
+    if (mDequeuedBufferCount == 0 && mState != STATE_IN_CONFIG &&
+            mState != STATE_IN_RECONFIG) {
+        ALOGV("%s: Stream %d: All buffers returned; now idle", __FUNCTION__,
+                mId);
+        sp<StatusTracker> statusTracker = mStatusTracker.promote();
+        if (statusTracker != 0) {
+            statusTracker->markComponentIdle(mStatusId, mCombinedFence);
+        }
+    }
+
     mBufferReturnedSignal.signal();
 
     if (output) {
         mLastTimestamp = timestamp;
     }
 
-    return OK;
+    return res;
 }
 
 

services/camera/libcameraservice/device3/Camera3IOStreamBase.h

diff --git a/services/camera/libcameraservice/device3/Camera3IOStreamBase.h b/services/camera/libcameraservice/device3/Camera3IOStreamBase.h
index 9432a59..fcb9d04 100644
--- a/services/camera/libcameraservice/device3/Camera3IOStreamBase.h
+++ b/services/camera/libcameraservice/device3/Camera3IOStreamBase.h
@@ -43,7 +43,6 @@
      * Camera3Stream interface
      */
 
-    virtual status_t waitUntilIdle(nsecs_t timeout);
     virtual void     dump(int fd, const Vector<String16> &args) const;
 
   protected:

services/camera/libcameraservice/device3/Camera3InputStream.cpp

diff --git a/services/camera/libcameraservice/device3/Camera3InputStream.cpp b/services/camera/libcameraservice/device3/Camera3InputStream.cpp
index c80f512..e1c492b 100644
--- a/services/camera/libcameraservice/device3/Camera3InputStream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3InputStream.cpp
@@ -115,7 +115,6 @@
                 bufferFound = true;
                 bufferItem = tmp;
                 mBuffersInFlight.erase(it);
-                mDequeuedBufferCount--;
             }
         }
     }
@@ -148,12 +147,11 @@
     if (res != OK) {
         ALOGE("%s: Stream %d: Error releasing buffer back to buffer queue:"
                 " %s (%d)", __FUNCTION__, mId, strerror(-res), res);
-        return res;
     }
 
     *releaseFenceOut = releaseFence;
 
-    return OK;
+    return res;
 }
 
 status_t Camera3InputStream::returnInputBufferLocked(
@@ -201,13 +199,33 @@
     assert(mMaxSize == 0);
     assert(camera3_stream::format != HAL_PIXEL_FORMAT_BLOB);
 
-    mTotalBufferCount = BufferQueue::MIN_UNDEQUEUED_BUFFERS +
-                        camera3_stream::max_buffers;
     mDequeuedBufferCount = 0;
     mFrameCount = 0;
 
     if (mConsumer.get() == 0) {
         sp<BufferQueue> bq = new BufferQueue();
+
+        int minUndequeuedBuffers = 0;
+        res = bq->query(NATIVE_WINDOW_MIN_UNDEQUEUED_BUFFERS, &minUndequeuedBuffers);
+        if (res != OK || minUndequeuedBuffers < 0) {
+            ALOGE("%s: Stream %d: Could not query min undequeued buffers (error %d, bufCount %d)",
+                  __FUNCTION__, mId, res, minUndequeuedBuffers);
+            return res;
+        }
+        size_t minBufs = static_cast<size_t>(minUndequeuedBuffers);
+        /*
+         * We promise never to 'acquire' more than camera3_stream::max_buffers
+         * at any one time.
+         *
+         * Boost the number up to meet the minimum required buffer count.
+         *
+         * (Note that this sets consumer-side buffer count only,
+         * and not the sum of producer+consumer side as in other camera streams).
+         */
+        mTotalBufferCount = camera3_stream::max_buffers > minBufs ?
+            camera3_stream::max_buffers : minBufs;
+        // TODO: somehow set the total buffer count when producer connects?
+
         mConsumer = new BufferItemConsumer(bq, camera3_stream::usage,
                                            mTotalBufferCount);
         mConsumer->setName(String8::format("Camera3-InputStream-%d", mId));

services/camera/libcameraservice/device3/Camera3InputStream.h

diff --git a/services/camera/libcameraservice/device3/Camera3InputStream.h b/services/camera/libcameraservice/device3/Camera3InputStream.h
index 681d684..ae49467 100644
--- a/services/camera/libcameraservice/device3/Camera3InputStream.h
+++ b/services/camera/libcameraservice/device3/Camera3InputStream.h
@@ -44,6 +44,8 @@
 
     virtual void     dump(int fd, const Vector<String16> &args) const;
 
+    // TODO: expose an interface to get the IGraphicBufferProducer
+
   private:
 
     typedef BufferItemConsumer::BufferItem BufferItem;

services/camera/libcameraservice/device3/Camera3OutputStream.cpp

diff --git a/services/camera/libcameraservice/device3/Camera3OutputStream.cpp b/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
index 35cb5ba..682755d 100644
--- a/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
@@ -92,7 +92,22 @@
     ANativeWindowBuffer* anb;
     int fenceFd;
 
-    res = mConsumer->dequeueBuffer(mConsumer.get(), &anb, &fenceFd);
+    /**
+     * Release the lock briefly to avoid deadlock for below scenario:
+     * Thread 1: StreamingProcessor::startStream -> Camera3Stream::isConfiguring().
+     * This thread acquired StreamingProcessor lock and try to lock Camera3Stream lock.
+     * Thread 2: Camera3Stream::returnBuffer->StreamingProcessor::onFrameAvailable().
+     * This thread acquired Camera3Stream lock and bufferQueue lock, and try to lock
+     * StreamingProcessor lock.
+     * Thread 3: Camera3Stream::getBuffer(). This thread acquired Camera3Stream lock
+     * and try to lock bufferQueue lock.
+     * Then there is circular locking dependency.
+     */
+    sp<ANativeWindow> currentConsumer = mConsumer;
+    mLock.unlock();
+
+    res = currentConsumer->dequeueBuffer(currentConsumer.get(), &anb, &fenceFd);
+    mLock.lock();
     if (res != OK) {
         ALOGE("%s: Stream %d: Can't dequeue next output buffer: %s (%d)",
                 __FUNCTION__, mId, strerror(-res), res);
@@ -198,12 +213,11 @@
     mLock.lock();
     if (res != OK) {
         close(anwReleaseFence);
-        return res;
     }
 
     *releaseFenceOut = releaseFence;
 
-    return OK;
+    return res;
 }
 
 void Camera3OutputStream::dump(int fd, const Vector<String16> &args) const {

services/camera/libcameraservice/device3/Camera3Stream.cpp

diff --git a/services/camera/libcameraservice/device3/Camera3Stream.cpp b/services/camera/libcameraservice/device3/Camera3Stream.cpp
index a6872aa..6d2cf94 100644
--- a/services/camera/libcameraservice/device3/Camera3Stream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Stream.cpp
@@ -20,13 +20,18 @@
 
 #include <utils/Log.h>
 #include <utils/Trace.h>
-#include "Camera3Stream.h"
+#include "device3/Camera3Stream.h"
+#include "device3/StatusTracker.h"
 
 namespace android {
 
 namespace camera3 {
 
 Camera3Stream::~Camera3Stream() {
+    sp<StatusTracker> statusTracker = mStatusTracker.promote();
+    if (statusTracker != 0 && mStatusId != StatusTracker::NO_STATUS_ID) {
+        statusTracker->removeComponent(mStatusId);
+    }
 }
 
 Camera3Stream* Camera3Stream::cast(camera3_stream *stream) {
@@ -44,7 +49,8 @@
     mId(id),
     mName(String8::format("Camera3Stream[%d]", id)),
     mMaxSize(maxSize),
-    mState(STATE_CONSTRUCTED) {
+    mState(STATE_CONSTRUCTED),
+    mStatusId(StatusTracker::NO_STATUS_ID) {
 
     camera3_stream::stream_type = type;
     camera3_stream::width = width;
@@ -119,6 +125,15 @@
         return NULL;
     }
 
+    // Stop tracking if currently doing so
+    if (mStatusId != StatusTracker::NO_STATUS_ID) {
+        sp<StatusTracker> statusTracker = mStatusTracker.promote();
+        if (statusTracker != 0) {
+            statusTracker->removeComponent(mStatusId);
+        }
+        mStatusId = StatusTracker::NO_STATUS_ID;
+    }
+
     if (mState == STATE_CONSTRUCTED) {
         mState = STATE_IN_CONFIG;
     } else { // mState == STATE_CONFIGURED
@@ -154,6 +169,12 @@
             return INVALID_OPERATION;
     }
 
+    // Register for idle tracking
+    sp<StatusTracker> statusTracker = mStatusTracker.promote();
+    if (statusTracker != 0) {
+        mStatusId = statusTracker->addComponent();
+    }
+
     // Check if the stream configuration is unchanged, and skip reallocation if
     // so. As documented in hardware/camera3.h:configure_streams().
     if (mState == STATE_IN_RECONFIG &&
@@ -265,6 +286,18 @@
     return hasOutstandingBuffersLocked();
 }
 
+status_t Camera3Stream::setStatusTracker(sp<StatusTracker> statusTracker) {
+    Mutex::Autolock l(mLock);
+    sp<StatusTracker> oldTracker = mStatusTracker.promote();
+    if (oldTracker != 0 && mStatusId != StatusTracker::NO_STATUS_ID) {
+        oldTracker->removeComponent(mStatusId);
+    }
+    mStatusId = StatusTracker::NO_STATUS_ID;
+    mStatusTracker = statusTracker;
+
+    return OK;
+}
+
 status_t Camera3Stream::disconnect() {
     ATRACE_CALL();
     Mutex::Autolock l(mLock);

services/camera/libcameraservice/device3/Camera3Stream.h

diff --git a/services/camera/libcameraservice/device3/Camera3Stream.h b/services/camera/libcameraservice/device3/Camera3Stream.h
index b64fd86..6eeb721 100644
--- a/services/camera/libcameraservice/device3/Camera3Stream.h
+++ b/services/camera/libcameraservice/device3/Camera3Stream.h
@@ -190,12 +190,11 @@
     enum {
         TIMEOUT_NEVER = -1
     };
+
     /**
-     * Wait until the HAL is done with all of this stream's buffers, including
-     * signalling all release fences. Returns TIMED_OUT if the timeout is exceeded,
-     * OK on success. Pass in TIMEOUT_NEVER for timeout to indicate an indefinite wait.
+     * Set the status tracker to notify about idle transitions
      */
-    virtual status_t waitUntilIdle(nsecs_t timeout) = 0;
+    virtual status_t setStatusTracker(sp<StatusTracker> statusTracker);
 
     /**
      * Disconnect stream from its non-HAL endpoint. After this,
@@ -267,6 +266,11 @@
     // INVALID_OPERATION if they cannot be obtained.
     virtual status_t getEndpointUsage(uint32_t *usage) = 0;
 
+    // Tracking for idle state
+    wp<StatusTracker> mStatusTracker;
+    // Status tracker component ID
+    int mStatusId;
+
   private:
     uint32_t oldUsage;
     uint32_t oldMaxBuffers;

services/camera/libcameraservice/device3/Camera3StreamInterface.h

diff --git a/services/camera/libcameraservice/device3/Camera3StreamInterface.h b/services/camera/libcameraservice/device3/Camera3StreamInterface.h
index 4768536..c93ae15 100644
--- a/services/camera/libcameraservice/device3/Camera3StreamInterface.h
+++ b/services/camera/libcameraservice/device3/Camera3StreamInterface.h
@@ -26,6 +26,8 @@
 
 namespace camera3 {
 
+class StatusTracker;
+
 /**
  * An interface for managing a single stream of input and/or output data from
  * the camera device.
@@ -128,13 +130,11 @@
     enum {
         TIMEOUT_NEVER = -1
     };
+
     /**
-     * Wait until the HAL is done with all of this stream's buffers, including
-     * signalling all release fences. Returns TIMED_OUT if the timeout is
-     * exceeded, OK on success. Pass in TIMEOUT_NEVER for timeout to indicate
-     * an indefinite wait.
+     * Set the state tracker to use for signaling idle transitions.
      */
-    virtual status_t waitUntilIdle(nsecs_t timeout) = 0;
+    virtual status_t setStatusTracker(sp<StatusTracker> statusTracker) = 0;
 
     /**
      * Disconnect stream from its non-HAL endpoint. After this,

services/camera/libcameraservice/device3/Camera3ZslStream.cpp

diff --git a/services/camera/libcameraservice/device3/Camera3ZslStream.cpp b/services/camera/libcameraservice/device3/Camera3ZslStream.cpp
index 04f5dc5..e509350 100644
--- a/services/camera/libcameraservice/device3/Camera3ZslStream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3ZslStream.cpp
@@ -109,14 +109,14 @@
 } // namespace anonymous
 
 Camera3ZslStream::Camera3ZslStream(int id, uint32_t width, uint32_t height,
-        int depth) :
+        int bufferCount) :
         Camera3OutputStream(id, CAMERA3_STREAM_BIDIRECTIONAL,
                             width, height,
                             HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED),
-        mDepth(depth) {
+        mDepth(bufferCount) {
 
     sp<BufferQueue> bq = new BufferQueue();
-    mProducer = new RingBufferConsumer(bq, GRALLOC_USAGE_HW_CAMERA_ZSL, depth);
+    mProducer = new RingBufferConsumer(bq, GRALLOC_USAGE_HW_CAMERA_ZSL, bufferCount);
     mConsumer = new Surface(bq);
 }
 
@@ -271,7 +271,7 @@
     Camera3IOStreamBase::dump(fd, args);
 
     lines = String8();
-    lines.appendFormat("      Input buffers pending: %d, in flight %d\n",
+    lines.appendFormat("      Input buffers pending: %zu, in flight %zu\n",
             mInputBufferQueue.size(), mBuffersInFlight.size());
     write(fd, lines.string(), lines.size());
 }

services/camera/libcameraservice/device3/Camera3ZslStream.h

diff --git a/services/camera/libcameraservice/device3/Camera3ZslStream.h b/services/camera/libcameraservice/device3/Camera3ZslStream.h
index c7f4490..6721832 100644
--- a/services/camera/libcameraservice/device3/Camera3ZslStream.h
+++ b/services/camera/libcameraservice/device3/Camera3ZslStream.h
@@ -37,10 +37,10 @@
         public Camera3OutputStream {
   public:
     /**
-     * Set up a ZSL stream of a given resolution. Depth is the number of buffers
+     * Set up a ZSL stream of a given resolution. bufferCount is the number of buffers
      * cached within the stream that can be retrieved for input.
      */
-    Camera3ZslStream(int id, uint32_t width, uint32_t height, int depth);
+    Camera3ZslStream(int id, uint32_t width, uint32_t height, int bufferCount);
     ~Camera3ZslStream();
 
     virtual void     dump(int fd, const Vector<String16> &args) const;

services/camera/libcameraservice/device3/StatusTracker.cpp

diff --git a/services/camera/libcameraservice/device3/StatusTracker.cpp b/services/camera/libcameraservice/device3/StatusTracker.cpp
new file mode 100644
index 0000000..ab5419f
--- /dev/null
+++ b/services/camera/libcameraservice/device3/StatusTracker.cpp
@@ -0,0 +1,219 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "Camera3-Status"
+#define ATRACE_TAG ATRACE_TAG_CAMERA
+//#define LOG_NDEBUG 0
+
+// This is needed for stdint.h to define INT64_MAX in C++
+#define __STDC_LIMIT_MACROS
+
+#include <utils/Log.h>
+#include <utils/Trace.h>
+#include <ui/Fence.h>
+
+#include "device3/StatusTracker.h"
+#include "device3/Camera3Device.h"
+
+namespace android {
+
+namespace camera3 {
+
+StatusTracker::StatusTracker(wp<Camera3Device> parent) :
+        mComponentsChanged(false),
+        mParent(parent),
+        mNextComponentId(0),
+        mIdleFence(new Fence()),
+        mDeviceState(IDLE) {
+}
+
+StatusTracker::~StatusTracker() {
+}
+
+int StatusTracker::addComponent() {
+    int id;
+    ssize_t err;
+    {
+        Mutex::Autolock l(mLock);
+        id = mNextComponentId++;
+        ALOGV("%s: Adding new component %d", __FUNCTION__, id);
+
+        err = mStates.add(id, IDLE);
+        ALOGE_IF(err < 0, "%s: Can't add new component %d: %s (%d)",
+                __FUNCTION__, id, strerror(-err), err);
+    }
+
+    if (err >= 0) {
+        Mutex::Autolock pl(mPendingLock);
+        mComponentsChanged = true;
+        mPendingChangeSignal.signal();
+    }
+
+    return err < 0 ? err : id;
+}
+
+void StatusTracker::removeComponent(int id) {
+    ssize_t idx;
+    {
+        Mutex::Autolock l(mLock);
+        ALOGV("%s: Removing component %d", __FUNCTION__, id);
+        idx = mStates.removeItem(id);
+    }
+
+    if (idx >= 0) {
+        Mutex::Autolock pl(mPendingLock);
+        mComponentsChanged = true;
+        mPendingChangeSignal.signal();
+    }
+
+    return;
+}
+
+
+void StatusTracker::markComponentIdle(int id, const sp<Fence>& componentFence) {
+    markComponent(id, IDLE, componentFence);
+}
+
+void StatusTracker::markComponentActive(int id) {
+    markComponent(id, ACTIVE, Fence::NO_FENCE);
+}
+
+void StatusTracker::markComponent(int id, ComponentState state,
+        const sp<Fence>& componentFence) {
+    ALOGV("%s: Component %d is now %s", __FUNCTION__, id,
+            state == IDLE ? "idle" : "active");
+    Mutex::Autolock l(mPendingLock);
+
+    StateChange newState = {
+        id,
+        state,
+        componentFence
+    };
+
+    mPendingChangeQueue.add(newState);
+    mPendingChangeSignal.signal();
+}
+
+void StatusTracker::requestExit() {
+    // First mark thread dead
+    Thread::requestExit();
+    // Then exit any waits
+    mPendingChangeSignal.signal();
+}
+
+StatusTracker::ComponentState StatusTracker::getDeviceStateLocked() {
+    for (size_t i = 0; i < mStates.size(); i++) {
+        if (mStates.valueAt(i) == ACTIVE) {
+            ALOGV("%s: Component %d not idle", __FUNCTION__,
+                    mStates.keyAt(i));
+            return ACTIVE;
+        }
+    }
+    // - If not yet signaled, getSignalTime returns INT64_MAX
+    // - If invalid fence or error, returns -1
+    // - Otherwise returns time of signalling.
+    // Treat -1 as 'signalled', since HAL may not be using fences, and want
+    // to be able to idle in case of errors.
+    nsecs_t signalTime = mIdleFence->getSignalTime();
+    bool fencesDone = signalTime != INT64_MAX;
+
+    ALOGV_IF(!fencesDone, "%s: Fences still to wait on", __FUNCTION__);
+
+    return fencesDone ? IDLE : ACTIVE;
+}
+
+bool StatusTracker::threadLoop() {
+    status_t res;
+
+    // Wait for state updates
+    {
+        Mutex::Autolock pl(mPendingLock);
+        while (mPendingChangeQueue.size() == 0 && !mComponentsChanged) {
+            res = mPendingChangeSignal.waitRelative(mPendingLock,
+                    kWaitDuration);
+            if (exitPending()) return false;
+            if (res != OK) {
+                if (res != TIMED_OUT) {
+                    ALOGE("%s: Error waiting on state changes: %s (%d)",
+                            __FUNCTION__, strerror(-res), res);
+                }
+                // TIMED_OUT is expected
+                break;
+            }
+        }
+    }
+
+    // After new pending states appear, or timeout, check if we're idle.  Even
+    // with timeout, need to check to account for fences that may still be
+    // clearing out
+    sp<Camera3Device> parent;
+    {
+        Mutex::Autolock pl(mPendingLock);
+        Mutex::Autolock l(mLock);
+
+        // Collect all pending state updates and see if the device
+        // collectively transitions between idle and active for each one
+
+        // First pass for changed components or fence completions
+        ComponentState prevState = getDeviceStateLocked();
+        if (prevState != mDeviceState) {
+            // Only collect changes to overall device state
+            mStateTransitions.add(prevState);
+        }
+        // For each pending component state update, check if we've transitioned
+        // to a new overall device state
+        for (size_t i = 0; i < mPendingChangeQueue.size(); i++) {
+            const StateChange &newState = mPendingChangeQueue[i];
+            ssize_t idx = mStates.indexOfKey(newState.id);
+            // Ignore notices for unknown components
+            if (idx >= 0) {
+                // Update single component state
+                mStates.replaceValueAt(idx, newState.state);
+                mIdleFence = Fence::merge(String8("idleFence"),
+                        mIdleFence, newState.fence);
+                // .. and see if overall device state has changed
+                ComponentState newState = getDeviceStateLocked();
+                if (newState != prevState) {
+                    mStateTransitions.add(newState);
+                }
+                prevState = newState;
+            }
+        }
+        mPendingChangeQueue.clear();
+        mComponentsChanged = false;
+
+        // Store final state after all pending state changes are done with
+
+        mDeviceState = prevState;
+        parent = mParent.promote();
+    }
+
+    // Notify parent for all intermediate transitions
+    if (mStateTransitions.size() > 0 && parent.get()) {
+        for (size_t i = 0; i < mStateTransitions.size(); i++) {
+            bool idle = (mStateTransitions[i] == IDLE);
+            ALOGV("Camera device is now %s", idle ? "idle" : "active");
+            parent->notifyStatus(idle);
+        }
+    }
+    mStateTransitions.clear();
+
+    return true;
+}
+
+} // namespace android
+
+} // namespace camera3

services/camera/libcameraservice/device3/StatusTracker.h

diff --git a/services/camera/libcameraservice/device3/StatusTracker.h b/services/camera/libcameraservice/device3/StatusTracker.h
new file mode 100644
index 0000000..49cecb3
--- /dev/null
+++ b/services/camera/libcameraservice/device3/StatusTracker.h
@@ -0,0 +1,130 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_SERVERS_CAMERA3_STATUSTRACKER_H
+#define ANDROID_SERVERS_CAMERA3_STATUSTRACKER_H
+
+#include <utils/Condition.h>
+#include <utils/Errors.h>
+#include <utils/List.h>
+#include <utils/Mutex.h>
+#include <utils/Thread.h>
+#include <utils/KeyedVector.h>
+#include <hardware/camera3.h>
+
+#include "common/CameraDeviceBase.h"
+
+namespace android {
+
+class Camera3Device;
+class Fence;
+
+namespace camera3 {
+
+/**
+ * State tracking for idle and other collective state transitions.
+ * Collects idle notifications from different sources and calls the
+ * parent when all of them become idle.
+ *
+ * The parent is responsible for synchronizing the status updates with its
+ * internal state correctly, which means the notifyStatus call to the parent may
+ * block for a while.
+ */
+class StatusTracker: public Thread {
+  public:
+    StatusTracker(wp<Camera3Device> parent);
+    ~StatusTracker();
+
+    // An always-invalid component ID
+    static const int NO_STATUS_ID = -1;
+
+    // Add a component to track; returns non-negative unique ID for the new
+    // component on success, negative error code on failure.
+    // New components start in the idle state.
+    int addComponent();
+
+    // Remove existing component from idle tracking. Ignores unknown IDs
+    void removeComponent(int id);
+
+    // Set the state of a tracked component to be idle. Ignores unknown IDs; can
+    // accept a fence to wait on to complete idle.  The fence is merged with any
+    // previous fences given, which means they all must signal before the
+    // component is considered idle.
+    void markComponentIdle(int id, const sp<Fence>& componentFence);
+
+    // Set the state of a tracked component to be active. Ignores unknown IDs.
+    void markComponentActive(int id);
+
+    virtual void requestExit();
+  protected:
+
+    virtual bool threadLoop();
+
+  private:
+    enum ComponentState {
+        IDLE,
+        ACTIVE
+    };
+
+    void markComponent(int id, ComponentState state,
+            const sp<Fence>& componentFence);
+
+    // Guards mPendingChange, mPendingStates, mComponentsChanged
+    Mutex mPendingLock;
+
+    Condition mPendingChangeSignal;
+
+    struct StateChange {
+        int id;
+        ComponentState state;
+        sp<Fence> fence;
+    };
+    // A queue of yet-to-be-processed state changes to components
+    Vector<StateChange> mPendingChangeQueue;
+    bool mComponentsChanged;
+
+    wp<Camera3Device> mParent;
+
+    // Guards rest of internals. Must be locked after mPendingLock if both used.
+    Mutex mLock;
+
+    int mNextComponentId;
+
+    // Current component states
+    KeyedVector<int, ComponentState> mStates;
+    // Merged fence for all processed state changes
+    sp<Fence> mIdleFence;
+    // Current overall device state
+    ComponentState mDeviceState;
+
+    // Private to threadLoop
+
+    // Determine current overall device state
+    // We're IDLE iff
+    // - All components are currently IDLE
+    // - The merged fence for all component updates has signalled
+    ComponentState getDeviceStateLocked();
+
+    Vector<ComponentState> mStateTransitions;
+
+    static const nsecs_t kWaitDuration = 250000000LL; // 250 ms
+};
+
+} // namespace camera3
+
+} // namespace android
+
+#endif

services/camera/libcameraservice/gui/RingBufferConsumer.cpp

diff --git a/services/camera/libcameraservice/gui/RingBufferConsumer.cpp b/services/camera/libcameraservice/gui/RingBufferConsumer.cpp
index ebc7ea7..9a6dc28 100644
--- a/services/camera/libcameraservice/gui/RingBufferConsumer.cpp
+++ b/services/camera/libcameraservice/gui/RingBufferConsumer.cpp
@@ -21,11 +21,11 @@
 
 #include <gui/RingBufferConsumer.h>
 
-#define BI_LOGV(x, ...) ALOGV("[%s] "x, mName.string(), ##__VA_ARGS__)
-#define BI_LOGD(x, ...) ALOGD("[%s] "x, mName.string(), ##__VA_ARGS__)
-#define BI_LOGI(x, ...) ALOGI("[%s] "x, mName.string(), ##__VA_ARGS__)
-#define BI_LOGW(x, ...) ALOGW("[%s] "x, mName.string(), ##__VA_ARGS__)
-#define BI_LOGE(x, ...) ALOGE("[%s] "x, mName.string(), ##__VA_ARGS__)
+#define BI_LOGV(x, ...) ALOGV("[%s] " x, mName.string(), ##__VA_ARGS__)
+#define BI_LOGD(x, ...) ALOGD("[%s] " x, mName.string(), ##__VA_ARGS__)
+#define BI_LOGI(x, ...) ALOGI("[%s] " x, mName.string(), ##__VA_ARGS__)
+#define BI_LOGW(x, ...) ALOGW("[%s] " x, mName.string(), ##__VA_ARGS__)
+#define BI_LOGE(x, ...) ALOGE("[%s] " x, mName.string(), ##__VA_ARGS__)
 
 #undef assert
 #define assert(x) ALOG_ASSERT((x), #x)

services/camera/libcameraservice/gui/RingBufferConsumer.h

diff --git a/services/camera/libcameraservice/gui/RingBufferConsumer.h b/services/camera/libcameraservice/gui/RingBufferConsumer.h
index b4ad824..a03736d 100644
--- a/services/camera/libcameraservice/gui/RingBufferConsumer.h
+++ b/services/camera/libcameraservice/gui/RingBufferConsumer.h
@@ -64,7 +64,7 @@
     // bufferCount parameter specifies how many buffers can be pinned for user
     // access at the same time.
     RingBufferConsumer(const sp<IGraphicBufferConsumer>& consumer, uint32_t consumerUsage,
-            int bufferCount = BufferQueue::MIN_UNDEQUEUED_BUFFERS);
+            int bufferCount);
 
     virtual ~RingBufferConsumer();
 

services/camera/libcameraservice/utils/CameraTraces.cpp

diff --git a/services/camera/libcameraservice/utils/CameraTraces.cpp b/services/camera/libcameraservice/utils/CameraTraces.cpp
new file mode 100644
index 0000000..346e15f
--- /dev/null
+++ b/services/camera/libcameraservice/utils/CameraTraces.cpp
@@ -0,0 +1,94 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "CameraTraces"
+#define ATRACE_TAG ATRACE_TAG_CAMERA
+//#define LOG_NDEBUG 0
+
+#include "utils/CameraTraces.h"
+#include <utils/ProcessCallStack.h>
+
+#include <utils/Mutex.h>
+#include <utils/List.h>
+
+#include <utils/Log.h>
+#include <cutils/trace.h>
+
+namespace android {
+namespace camera3 {
+
+struct CameraTracesImpl {
+    Mutex                    tracesLock;
+    List<ProcessCallStack>   pcsList;
+}; // class CameraTraces::Impl;
+
+static CameraTracesImpl gImpl;
+CameraTracesImpl& CameraTraces::sImpl = gImpl;
+
+void CameraTraces::saveTrace() {
+    ALOGV("%s: begin", __FUNCTION__);
+    ATRACE_BEGIN("CameraTraces::saveTrace");
+    Mutex::Autolock al(sImpl.tracesLock);
+
+    List<ProcessCallStack>& pcsList = sImpl.pcsList;
+
+    // Insert new ProcessCallStack, and immediately crawl all the threads
+    pcsList.push_front(ProcessCallStack());
+    ProcessCallStack& pcs = *pcsList.begin();
+    pcs.update();
+
+    if (pcsList.size() > MAX_TRACES) {
+        // Prune list periodically and discard oldest entry
+        pcsList.erase(--pcsList.end());
+    }
+
+    IF_ALOGV() {
+        pcs.log(LOG_TAG, ANDROID_LOG_VERBOSE);
+    }
+
+    ALOGD("Process trace saved. Use dumpsys media.camera to view.");
+
+    ATRACE_END();
+}
+
+status_t CameraTraces::dump(int fd, const Vector<String16> &args __attribute__((unused))) {
+    ALOGV("%s: fd = %d", __FUNCTION__, fd);
+    Mutex::Autolock al(sImpl.tracesLock);
+    List<ProcessCallStack>& pcsList = sImpl.pcsList;
+
+    if (fd < 0) {
+        ALOGW("%s: Negative FD (%d)", __FUNCTION__, fd);
+        return BAD_VALUE;
+    }
+
+    fdprintf(fd, "Camera traces (%zu):\n", pcsList.size());
+
+    if (pcsList.empty()) {
+        fdprintf(fd, "  No camera traces collected.\n");
+    }
+
+    // Print newest items first
+    List<ProcessCallStack>::iterator it, end;
+    for (it = pcsList.begin(), end = pcsList.end(); it != end; ++it) {
+        const ProcessCallStack& pcs = *it;
+        pcs.dump(fd, DUMP_INDENT);
+    }
+
+    return OK;
+}
+
+}; // namespace camera3
+}; // namespace android

services/camera/libcameraservice/utils/CameraTraces.h

diff --git a/services/camera/libcameraservice/utils/CameraTraces.h b/services/camera/libcameraservice/utils/CameraTraces.h
new file mode 100644
index 0000000..d10dbc9
--- /dev/null
+++ b/services/camera/libcameraservice/utils/CameraTraces.h
@@ -0,0 +1,66 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_SERVERS_CAMERA_TRACES_H_
+#define ANDROID_SERVERS_CAMERA_TRACES_H_
+
+#include <utils/Errors.h>
+#include <utils/String16.h>
+#include <utils/Vector.h>
+
+namespace android {
+namespace camera3 {
+
+class CameraTracesImpl;
+
+// Collect a list of the process's stack traces
+class CameraTraces {
+public:
+    /**
+     * Save the current stack trace for each thread in the process. At most
+     * MAX_TRACES will be saved, after which the oldest traces will be discarded.
+     *
+     * <p>Use CameraTraces::dump to print out the traces.</p>
+     */
+    static void     saveTrace();
+
+    /**
+     * Prints all saved traces to the specified file descriptor.
+     *
+     * <p>Each line is indented by DUMP_INDENT spaces.</p>
+     */
+    static status_t dump(int fd, const Vector<String16>& args);
+
+private:
+    enum {
+        // Don't collect more than 100 traces. Discard oldest.
+        MAX_TRACES = 100,
+
+        // Insert 2 spaces when dumping the traces
+        DUMP_INDENT = 2,
+    };
+
+    CameraTraces();
+    ~CameraTraces();
+    CameraTraces(CameraTraces& rhs);
+
+    static CameraTracesImpl& sImpl;
+}; // class CameraTraces
+
+}; // namespace camera3
+}; // namespace android
+
+#endif // ANDROID_SERVERS_CAMERA_TRACES_H_

services/medialog/MediaLogService.cpp

diff --git a/services/medialog/MediaLogService.cpp b/services/medialog/MediaLogService.cpp
index 683fdf3..0c7fbbd 100644
--- a/services/medialog/MediaLogService.cpp
+++ b/services/medialog/MediaLogService.cpp
@@ -54,7 +54,7 @@
     }
 }
 
-status_t MediaLogService::dump(int fd, const Vector<String16>& args)
+status_t MediaLogService::dump(int fd, const Vector<String16>& args __unused)
 {
     // FIXME merge with similar but not identical code at services/audioflinger/ServiceUtilities.cpp
     static const String16 sDump("android.permission.DUMP");

tools/resampler_tools/fir.cpp

diff --git a/tools/resampler_tools/fir.cpp b/tools/resampler_tools/fir.cpp
index cc3d509..3d6a74d 100644
--- a/tools/resampler_tools/fir.cpp
+++ b/tools/resampler_tools/fir.cpp
@@ -20,15 +20,25 @@
 #include <stdlib.h>
 #include <string.h>
 
-static double sinc(double x) {
+static inline double sinc(double x) {
     if (fabs(x) == 0.0f) return 1.0f;
     return sin(x) / x;
 }
 
-static double sqr(double x) {
+static inline double sqr(double x) {
     return x*x;
 }
 
+static inline int64_t toint(double x, int64_t maxval) {
+    int64_t v;
+
+    v = static_cast<int64_t>(floor(x * maxval + 0.5));
+    if (v >= maxval) {
+        return maxval - 1; // error!
+    }
+    return v;
+}
+
 static double I0(double x) {
     // from the Numerical Recipes in C p. 237
     double ax,ans,y;
@@ -54,11 +64,12 @@
     return I0(beta * sqrt(1.0 - sqr((2.0*k)/N - 1.0))) / I0(beta);
 }
 
-
 static void usage(char* name) {
     fprintf(stderr,
-            "usage: %s [-h] [-d] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings] [-f {float|fixed}] [-b beta] [-v dBFS] [-l lerp]\n"
-            "       %s [-h] [-d] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings] [-f {float|fixed}] [-b beta] [-v dBFS] -p M/N\n"
+            "usage: %s [-h] [-d] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings]"
+            " [-f {float|fixed|fixed16}] [-b beta] [-v dBFS] [-l lerp]\n"
+            "       %s [-h] [-d] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings]"
+            " [-f {float|fixed|fixed16}] [-b beta] [-v dBFS] -p M/N\n"
             "    -h    this help message\n"
             "    -d    debug, print comma-separated coefficient table\n"
             "    -p    generate poly-phase filter coefficients, with sample increment M/N\n"
@@ -66,6 +77,7 @@
             "    -c    cut-off frequency (20478)\n"
             "    -n    number of zero-crossings on one side (8)\n"
             "    -l    number of lerping bits (4)\n"
+            "    -m    number of polyphases (related to -l, default 16)\n"
             "    -f    output format, can be fixed-point or floating-point (fixed)\n"
             "    -b    kaiser window parameter beta (7.865 [-80dB])\n"
             "    -v    attenuation in dBFS (0)\n",
@@ -77,8 +89,7 @@
 int main(int argc, char** argv)
 {
     // nc is the number of bits to store the coefficients
-    const int nc = 32;
-
+    int nc = 32;
     bool polyphase = false;
     unsigned int polyM = 160;
     unsigned int polyN = 147;
@@ -88,7 +99,6 @@
     double atten = 1;
     int format = 0;
 
-
     // in order to keep the errors associated with the linear
     // interpolation of the coefficients below the quantization error
     // we must satisfy:
@@ -104,7 +114,6 @@
     // Smith, J.O. Digital Audio Resampling Home Page
     // https://ccrma.stanford.edu/~jos/resample/, 2011-03-29
     //
-    int nz = 4;
 
     //         | 0.1102*(A - 8.7)                         A > 50
     //  beta = | 0.5842*(A - 21)^0.4 + 0.07886*(A - 21)   21 <= A <= 50
@@ -123,7 +132,6 @@
     //   100 dB   10.056
     double beta = 7.865;
 
-
     // 2*nzc = (A - 8) / (2.285 * dw)
     //      with dw the transition width = 2*pi*dF/Fs
     //
@@ -148,8 +156,9 @@
     // nzc  = 20
     //
 
+    int M = 1 << 4; // number of phases for interpolation
     int ch;
-    while ((ch = getopt(argc, argv, ":hds:c:n:f:l:b:p:v:")) != -1) {
+    while ((ch = getopt(argc, argv, ":hds:c:n:f:l:m:b:p:v:z:")) != -1) {
         switch (ch) {
             case 'd':
                 debug = true;
@@ -169,13 +178,26 @@
             case 'n':
                 nzc = atoi(optarg);
                 break;
+            case 'm':
+                M = atoi(optarg);
+                break;
             case 'l':
-                nz = atoi(optarg);
+                M = 1 << atoi(optarg);
                 break;
             case 'f':
-                if (!strcmp(optarg,"fixed")) format = 0;
-                else if (!strcmp(optarg,"float")) format = 1;
-                else usage(argv[0]);
+                if (!strcmp(optarg, "fixed")) {
+                    format = 0;
+                }
+                else if (!strcmp(optarg, "fixed16")) {
+                    format = 0;
+                    nc = 16;
+                }
+                else if (!strcmp(optarg, "float")) {
+                    format = 1;
+                }
+                else {
+                    usage(argv[0]);
+                }
                 break;
             case 'b':
                 beta = atof(optarg);
@@ -193,11 +215,14 @@
     // cut off frequency ratio Fc/Fs
     double Fcr = Fc / Fs;
 
-
     // total number of coefficients (one side)
-    const int M = (1 << nz);
+
     const int N = M * nzc;
 
+    // lerp (which is most useful if M is a power of 2)
+
+    int nz = 0; // recalculate nz as the bits needed to represent M
+    for (int i = M-1 ; i; i>>=1, nz++);
     // generate the right half of the filter
     if (!debug) {
         printf("// cmd-line: ");
@@ -207,7 +232,7 @@
         printf("\n");
         if (!polyphase) {
             printf("const int32_t RESAMPLE_FIR_SIZE           = %d;\n", N);
-            printf("const int32_t RESAMPLE_FIR_LERP_INT_BITS  = %d;\n", nz);
+            printf("const int32_t RESAMPLE_FIR_INT_PHASES     = %d;\n", M);
             printf("const int32_t RESAMPLE_FIR_NUM_COEF       = %d;\n", nzc);
         } else {
             printf("const int32_t RESAMPLE_FIR_SIZE           = %d;\n", 2*nzc*polyN);
@@ -224,7 +249,7 @@
         for (int i=0 ; i<=M ; i++) { // an extra set of coefs for interpolation
             for (int j=0 ; j<nzc ; j++) {
                 int ix = j*M + i;
-                double x = (2.0 * M_PI * ix * Fcr) / (1 << nz);
+                double x = (2.0 * M_PI * ix * Fcr) / M;
                 double y = kaiser(ix+N, 2*N, beta) * sinc(x) * 2.0 * Fcr;
                 y *= atten;
 
@@ -232,11 +257,13 @@
                     if (j == 0)
                         printf("\n    ");
                 }
-
                 if (!format) {
-                    int64_t yi = floor(y * ((1ULL<<(nc-1))) + 0.5);
-                    if (yi >= (1LL<<(nc-1))) yi = (1LL<<(nc-1))-1;
-                    printf("0x%08x, ", int32_t(yi));
+                    int64_t yi = toint(y, 1ULL<<(nc-1));
+                    if (nc > 16) {
+                        printf("0x%08x, ", int32_t(yi));
+                    } else {
+                        printf("0x%04x, ", int32_t(yi)&0xffff);
+                    }
                 } else {
                     printf("%.9g%s ", y, debug ? "," : "f,");
                 }
@@ -254,9 +281,12 @@
                 double y = kaiser(i+N, 2*N, beta) * sinc(x) * 2.0 * Fcr;;
                 y *= atten;
                 if (!format) {
-                    int64_t yi = floor(y * ((1ULL<<(nc-1))) + 0.5);
-                    if (yi >= (1LL<<(nc-1))) yi = (1LL<<(nc-1))-1;
-                    printf("0x%08x", int32_t(yi));
+                    int64_t yi = toint(y, 1ULL<<(nc-1));
+                    if (nc > 16) {
+                        printf("0x%08x, ", int32_t(yi));
+                    } else {
+                        printf("0x%04x, ", int32_t(yi)&0xffff);
+                    }
                 } else {
                     printf("%.9g%s", y, debug ? "" : "f");
                 }
@@ -277,5 +307,3 @@
 }
 
 // http://www.csee.umbc.edu/help/sound/AFsp-V2R1/html/audio/ResampAudio.html
-
-