diff --git a/CleanSpec.mk b/CleanSpec.mk
index e6d9ebf..94aac5c 100644
--- a/CleanSpec.mk
+++ b/CleanSpec.mk
@@ -47,6 +47,8 @@
 $(call add-clean-step, rm -rf $(PRODUCT_OUT)/obj/lib/libmedia_native.so)
 $(call add-clean-step, rm -rf $(PRODUCT_OUT)/symbols/system/lib/libmedia_native.so)
 $(call add-clean-step, rm -rf $(PRODUCT_OUT)/system/lib/libmedia_native.so)
+$(call add-clean-step, rm -rf $(PRODUCT_OUT)/obj/SHARED_LIBRARIES/libaudioflinger_intermediates)
+$(call add-clean-step, rm -rf $(PRODUCT_OUT)/system/lib/libaudioflinger.so)
 # ************************************************
 # NEWER CLEAN STEPS MUST BE AT THE END OF THE LIST
 # ************************************************
diff --git a/camera/Android.mk b/camera/Android.mk
index e633450..369d0c5 100644
--- a/camera/Android.mk
+++ b/camera/Android.mk
@@ -1,3 +1,17 @@
+# Copyright 2010 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.
+
 CAMERA_CLIENT_LOCAL_PATH:= $(call my-dir)
 include $(call all-subdir-makefiles)
 include $(CLEAR_VARS)
@@ -21,6 +35,7 @@
 	camera2/CaptureRequest.cpp \
 	ProCamera.cpp \
 	CameraBase.cpp \
+	VendorTagDescriptor.cpp
 
 LOCAL_SHARED_LIBRARIES := \
 	libcutils \
@@ -34,6 +49,7 @@
 
 LOCAL_C_INCLUDES += \
 	system/media/camera/include \
+	system/media/private/camera/include
 
 LOCAL_MODULE:= libcamera_client
 
diff --git a/camera/ICameraService.cpp b/camera/ICameraService.cpp
index 5fc89fb..b86651f 100644
--- a/camera/ICameraService.cpp
+++ b/camera/ICameraService.cpp
@@ -17,6 +17,7 @@
 
 #define LOG_TAG "BpCameraService"
 #include <utils/Log.h>
+#include <utils/Errors.h>
 
 #include <stdint.h>
 #include <sys/types.h>
@@ -34,6 +35,7 @@
 #include <camera/camera2/ICameraDeviceUser.h>
 #include <camera/camera2/ICameraDeviceCallbacks.h>
 #include <camera/CameraMetadata.h>
+#include <camera/VendorTagDescriptor.h>
 
 namespace android {
 
@@ -143,6 +145,24 @@
         return result;
     }
 
+    // Get enumeration and description of vendor tags for camera
+    virtual status_t getCameraVendorTagDescriptor(/*out*/sp<VendorTagDescriptor>& desc) {
+        Parcel data, reply;
+        data.writeInterfaceToken(ICameraService::getInterfaceDescriptor());
+        remote()->transact(BnCameraService::GET_CAMERA_VENDOR_TAG_DESCRIPTOR, data, &reply);
+
+        if (readExceptionCode(reply)) return -EPROTO;
+        status_t result = reply.readInt32();
+
+        if (reply.readInt32() != 0) {
+            sp<VendorTagDescriptor> d;
+            if (VendorTagDescriptor::createFromParcel(&reply, /*out*/d) == OK) {
+                desc = d;
+            }
+        }
+        return result;
+    }
+
     // connect to camera service (android.hardware.Camera)
     virtual status_t connect(const sp<ICameraClient>& cameraClient, int cameraId,
                              const String16 &clientPackageName, int clientUid,
@@ -275,6 +295,22 @@
             info.writeToParcel(reply);
             return NO_ERROR;
         } break;
+        case GET_CAMERA_VENDOR_TAG_DESCRIPTOR: {
+            CHECK_INTERFACE(ICameraService, data, reply);
+            sp<VendorTagDescriptor> d;
+            status_t result = getCameraVendorTagDescriptor(d);
+            reply->writeNoException();
+            reply->writeInt32(result);
+
+            // out-variables are after exception and return value
+            if (d == NULL) {
+                reply->writeInt32(0);
+            } else {
+                reply->writeInt32(1); // means the parcelable is included
+                d->writeToParcel(reply);
+            }
+            return NO_ERROR;
+        } break;
         case CONNECT: {
             CHECK_INTERFACE(ICameraService, data, reply);
             sp<ICameraClient> cameraClient =
@@ -284,7 +320,7 @@
             int32_t clientUid = data.readInt32();
             sp<ICamera> camera;
             status_t status = connect(cameraClient, cameraId,
-                    clientName, clientUid, /*out*/ camera);
+                    clientName, clientUid, /*out*/camera);
             reply->writeNoException();
             reply->writeInt32(status);
             if (camera != NULL) {
@@ -304,7 +340,7 @@
             int32_t clientUid = data.readInt32();
             sp<IProCameraUser> camera;
             status_t status = connectPro(cameraClient, cameraId,
-                    clientName, clientUid, /*out*/ camera);
+                    clientName, clientUid, /*out*/camera);
             reply->writeNoException();
             reply->writeInt32(status);
             if (camera != NULL) {
@@ -324,7 +360,7 @@
             int32_t clientUid = data.readInt32();
             sp<ICameraDeviceUser> camera;
             status_t status = connectDevice(cameraClient, cameraId,
-                    clientName, clientUid, /*out*/ camera);
+                    clientName, clientUid, /*out*/camera);
             reply->writeNoException();
             reply->writeInt32(status);
             if (camera != NULL) {
diff --git a/camera/VendorTagDescriptor.cpp b/camera/VendorTagDescriptor.cpp
new file mode 100644
index 0000000..a0a6a51
--- /dev/null
+++ b/camera/VendorTagDescriptor.cpp
@@ -0,0 +1,319 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "VenderTagDescriptor"
+
+#include <binder/Parcel.h>
+#include <utils/Errors.h>
+#include <utils/Log.h>
+#include <utils/Mutex.h>
+#include <utils/Vector.h>
+#include <system/camera_metadata.h>
+#include <camera_metadata_hidden.h>
+
+#include "camera/VendorTagDescriptor.h"
+
+#include <string.h>
+
+namespace android {
+
+extern "C" {
+
+static int vendor_tag_descriptor_get_tag_count(const vendor_tag_ops_t* v);
+static void vendor_tag_descriptor_get_all_tags(const vendor_tag_ops_t* v, uint32_t* tagArray);
+static const char* vendor_tag_descriptor_get_section_name(const vendor_tag_ops_t* v, uint32_t tag);
+static const char* vendor_tag_descriptor_get_tag_name(const vendor_tag_ops_t* v, uint32_t tag);
+static int vendor_tag_descriptor_get_tag_type(const vendor_tag_ops_t* v, uint32_t tag);
+
+} /* extern "C" */
+
+
+static Mutex sLock;
+static sp<VendorTagDescriptor> sGlobalVendorTagDescriptor;
+
+VendorTagDescriptor::VendorTagDescriptor() {}
+VendorTagDescriptor::~VendorTagDescriptor() {}
+
+status_t VendorTagDescriptor::createDescriptorFromOps(const vendor_tag_ops_t* vOps,
+            /*out*/
+            sp<VendorTagDescriptor>& descriptor) {
+    if (vOps == NULL) {
+        ALOGE("%s: vendor_tag_ops argument was NULL.", __FUNCTION__);
+        return BAD_VALUE;
+    }
+
+    int tagCount = vOps->get_tag_count(vOps);
+    if (tagCount < 0 || tagCount > INT32_MAX) {
+        ALOGE("%s: tag count %d from vendor ops is invalid.", __FUNCTION__, tagCount);
+        return BAD_VALUE;
+    }
+
+    Vector<uint32_t> tagArray;
+    LOG_ALWAYS_FATAL_IF(tagArray.resize(tagCount) != tagCount,
+            "%s: too many (%u) vendor tags defined.", __FUNCTION__, tagCount);
+
+    vOps->get_all_tags(vOps, /*out*/tagArray.editArray());
+
+    sp<VendorTagDescriptor> desc = new VendorTagDescriptor();
+    desc->mTagCount = tagCount;
+
+    for (size_t i = 0; i < static_cast<size_t>(tagCount); ++i) {
+        uint32_t tag = tagArray[i];
+        if (tag < CAMERA_METADATA_VENDOR_TAG_BOUNDARY) {
+            ALOGE("%s: vendor tag %d not in vendor tag section.", __FUNCTION__, tag);
+            return BAD_VALUE;
+        }
+        const char *tagName = vOps->get_tag_name(vOps, tag);
+        if (tagName == NULL) {
+            ALOGE("%s: no tag name defined for vendor tag %d.", __FUNCTION__, tag);
+            return BAD_VALUE;
+        }
+        desc->mTagToNameMap.add(tag, String8(tagName));
+        const char *sectionName = vOps->get_section_name(vOps, tag);
+        if (sectionName == NULL) {
+            ALOGE("%s: no section name defined for vendor tag %d.", __FUNCTION__, tag);
+            return BAD_VALUE;
+        }
+        desc->mTagToSectionMap.add(tag, String8(sectionName));
+        int tagType = vOps->get_tag_type(vOps, tag);
+        if (tagType < 0 || tagType >= NUM_TYPES) {
+            ALOGE("%s: tag type %d from vendor ops does not exist.", __FUNCTION__, tagType);
+            return BAD_VALUE;
+        }
+        desc->mTagToTypeMap.add(tag, tagType);
+    }
+    descriptor = desc;
+    return OK;
+}
+
+status_t VendorTagDescriptor::createFromParcel(const Parcel* parcel,
+            /*out*/
+            sp<VendorTagDescriptor>& descriptor) {
+    status_t res = OK;
+    if (parcel == NULL) {
+        ALOGE("%s: parcel argument was NULL.", __FUNCTION__);
+        return BAD_VALUE;
+    }
+
+    int32_t tagCount = 0;
+    if ((res = parcel->readInt32(&tagCount)) != OK) {
+        ALOGE("%s: could not read tag count from parcel", __FUNCTION__);
+        return res;
+    }
+
+    if (tagCount < 0 || tagCount > INT32_MAX) {
+        ALOGE("%s: tag count %d from vendor ops is invalid.", __FUNCTION__, tagCount);
+        return BAD_VALUE;
+    }
+
+    sp<VendorTagDescriptor> desc = new VendorTagDescriptor();
+    desc->mTagCount = tagCount;
+
+    uint32_t tag;
+    int32_t tagType;
+    for (int32_t i = 0; i < tagCount; ++i) {
+        if ((res = parcel->readInt32(reinterpret_cast<int32_t*>(&tag))) != OK) {
+            ALOGE("%s: could not read tag id from parcel for index %d", __FUNCTION__, i);
+            break;
+        }
+        if (tag < CAMERA_METADATA_VENDOR_TAG_BOUNDARY) {
+            ALOGE("%s: vendor tag %d not in vendor tag section.", __FUNCTION__, tag);
+            res = BAD_VALUE;
+            break;
+        }
+        if ((res = parcel->readInt32(&tagType)) != OK) {
+            ALOGE("%s: could not read tag type from parcel for tag %d", __FUNCTION__, tag);
+            break;
+        }
+        if (tagType < 0 || tagType >= NUM_TYPES) {
+            ALOGE("%s: tag type %d from vendor ops does not exist.", __FUNCTION__, tagType);
+            res = BAD_VALUE;
+            break;
+        }
+        String8 tagName = parcel->readString8();
+        if (tagName.isEmpty()) {
+            ALOGE("%s: parcel tag name was NULL for tag %d.", __FUNCTION__, tag);
+            res = NOT_ENOUGH_DATA;
+            break;
+        }
+        String8 sectionName = parcel->readString8();
+        if (sectionName.isEmpty()) {
+            ALOGE("%s: parcel section name was NULL for tag %d.", __FUNCTION__, tag);
+            res = NOT_ENOUGH_DATA;
+            break;
+        }
+
+        desc->mTagToNameMap.add(tag, tagName);
+        desc->mTagToSectionMap.add(tag, sectionName);
+        desc->mTagToTypeMap.add(tag, tagType);
+    }
+
+    if (res != OK) {
+        return res;
+    }
+
+    descriptor = desc;
+    return res;
+}
+
+int VendorTagDescriptor::getTagCount() const {
+    size_t size = mTagToNameMap.size();
+    if (size == 0) {
+        return VENDOR_TAG_COUNT_ERR;
+    }
+    return size;
+}
+
+void VendorTagDescriptor::getTagArray(uint32_t* tagArray) const {
+    size_t size = mTagToNameMap.size();
+    for (size_t i = 0; i < size; ++i) {
+        tagArray[i] = mTagToNameMap.keyAt(i);
+    }
+}
+
+const char* VendorTagDescriptor::getSectionName(uint32_t tag) const {
+    ssize_t index = mTagToSectionMap.indexOfKey(tag);
+    if (index < 0) {
+        return VENDOR_SECTION_NAME_ERR;
+    }
+    return mTagToSectionMap.valueAt(index).string();
+}
+
+const char* VendorTagDescriptor::getTagName(uint32_t tag) const {
+    ssize_t index = mTagToNameMap.indexOfKey(tag);
+    if (index < 0) {
+        return VENDOR_TAG_NAME_ERR;
+    }
+    return mTagToNameMap.valueAt(index).string();
+}
+
+int VendorTagDescriptor::getTagType(uint32_t tag) const {
+    ssize_t index = mTagToNameMap.indexOfKey(tag);
+    if (index < 0) {
+        return VENDOR_TAG_TYPE_ERR;
+    }
+    return mTagToTypeMap.valueFor(tag);
+}
+
+status_t VendorTagDescriptor::writeToParcel(Parcel* parcel) const {
+    status_t res = OK;
+    if (parcel == NULL) {
+        ALOGE("%s: parcel argument was NULL.", __FUNCTION__);
+        return BAD_VALUE;
+    }
+
+    if ((res = parcel->writeInt32(mTagCount)) != OK) {
+        return res;
+    }
+
+    size_t size = mTagToNameMap.size();
+    uint32_t tag;
+    int32_t tagType;
+    for (size_t i = 0; i < size; ++i) {
+        tag = mTagToNameMap.keyAt(i);
+        String8 tagName = mTagToNameMap[i];
+        String8 sectionName = mTagToSectionMap.valueFor(tag);
+        tagType = mTagToTypeMap.valueFor(tag);
+        if ((res = parcel->writeInt32(tag)) != OK) break;
+        if ((res = parcel->writeInt32(tagType)) != OK) break;
+        if ((res = parcel->writeString8(tagName)) != OK) break;
+        if ((res = parcel->writeString8(sectionName)) != OK) break;
+    }
+
+    return res;
+}
+
+status_t VendorTagDescriptor::setAsGlobalVendorTagDescriptor(const sp<VendorTagDescriptor>& desc) {
+    status_t res = OK;
+    Mutex::Autolock al(sLock);
+    sGlobalVendorTagDescriptor = desc;
+
+    vendor_tag_ops_t* opsPtr = NULL;
+    if (desc != NULL) {
+        opsPtr = &(desc->mVendorOps);
+        opsPtr->get_tag_count = vendor_tag_descriptor_get_tag_count;
+        opsPtr->get_all_tags = vendor_tag_descriptor_get_all_tags;
+        opsPtr->get_section_name = vendor_tag_descriptor_get_section_name;
+        opsPtr->get_tag_name = vendor_tag_descriptor_get_tag_name;
+        opsPtr->get_tag_type = vendor_tag_descriptor_get_tag_type;
+    }
+    if((res = set_camera_metadata_vendor_ops(opsPtr)) != OK) {
+        ALOGE("%s: Could not set vendor tag descriptor, received error %s (%d)."
+                , __FUNCTION__, strerror(-res), res);
+    }
+    return res;
+}
+
+void VendorTagDescriptor::clearGlobalVendorTagDescriptor() {
+    Mutex::Autolock al(sLock);
+    set_camera_metadata_vendor_ops(NULL);
+    sGlobalVendorTagDescriptor.clear();
+}
+
+sp<VendorTagDescriptor> VendorTagDescriptor::getGlobalVendorTagDescriptor() {
+    Mutex::Autolock al(sLock);
+    return sGlobalVendorTagDescriptor;
+}
+
+extern "C" {
+
+int vendor_tag_descriptor_get_tag_count(const vendor_tag_ops_t* v) {
+    Mutex::Autolock al(sLock);
+    if (sGlobalVendorTagDescriptor == NULL) {
+        ALOGE("%s: Vendor tag descriptor not initialized.", __FUNCTION__);
+        return VENDOR_TAG_COUNT_ERR;
+    }
+    return sGlobalVendorTagDescriptor->getTagCount();
+}
+
+void vendor_tag_descriptor_get_all_tags(const vendor_tag_ops_t* v, uint32_t* tagArray) {
+    Mutex::Autolock al(sLock);
+    if (sGlobalVendorTagDescriptor == NULL) {
+        ALOGE("%s: Vendor tag descriptor not initialized.", __FUNCTION__);
+        return;
+    }
+    sGlobalVendorTagDescriptor->getTagArray(tagArray);
+}
+
+const char* vendor_tag_descriptor_get_section_name(const vendor_tag_ops_t* v, uint32_t tag) {
+    Mutex::Autolock al(sLock);
+    if (sGlobalVendorTagDescriptor == NULL) {
+        ALOGE("%s: Vendor tag descriptor not initialized.", __FUNCTION__);
+        return VENDOR_SECTION_NAME_ERR;
+    }
+    return sGlobalVendorTagDescriptor->getSectionName(tag);
+}
+
+const char* vendor_tag_descriptor_get_tag_name(const vendor_tag_ops_t* v, uint32_t tag) {
+    Mutex::Autolock al(sLock);
+    if (sGlobalVendorTagDescriptor == NULL) {
+        ALOGE("%s: Vendor tag descriptor not initialized.", __FUNCTION__);
+        return VENDOR_TAG_NAME_ERR;
+    }
+    return sGlobalVendorTagDescriptor->getTagName(tag);
+}
+
+int vendor_tag_descriptor_get_tag_type(const vendor_tag_ops_t* v, uint32_t tag) {
+    Mutex::Autolock al(sLock);
+    if (sGlobalVendorTagDescriptor == NULL) {
+        ALOGE("%s: Vendor tag descriptor not initialized.", __FUNCTION__);
+        return VENDOR_TAG_TYPE_ERR;
+    }
+    return sGlobalVendorTagDescriptor->getTagType(tag);
+}
+
+} /* extern "C" */
+} /* namespace android */
diff --git a/camera/tests/Android.mk b/camera/tests/Android.mk
index ec13911..61385e5 100644
--- a/camera/tests/Android.mk
+++ b/camera/tests/Android.mk
@@ -1,9 +1,24 @@
+# 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.
+
 LOCAL_PATH:= $(call my-dir)
 include $(CLEAR_VARS)
 
 LOCAL_SRC_FILES:= \
 	main.cpp \
 	ProCameraTests.cpp \
+	VendorTagDescriptorTests.cpp
 
 LOCAL_SHARED_LIBRARIES := \
 	libutils \
@@ -26,6 +41,8 @@
 	external/gtest/include \
 	external/stlport/stlport \
 	system/media/camera/include \
+	system/media/private/camera/include \
+	system/media/camera/tests \
 	frameworks/av/services/camera/libcameraservice \
 	frameworks/av/include/camera \
 	frameworks/native/include \
diff --git a/camera/tests/VendorTagDescriptorTests.cpp b/camera/tests/VendorTagDescriptorTests.cpp
new file mode 100644
index 0000000..6624e79
--- /dev/null
+++ b/camera/tests/VendorTagDescriptorTests.cpp
@@ -0,0 +1,204 @@
+/*
+ * 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 "VendorTagDescriptorTests"
+
+#include <binder/Parcel.h>
+#include <camera/VendorTagDescriptor.h>
+#include <camera_metadata_tests_fake_vendor.h>
+#include <camera_metadata_hidden.h>
+#include <system/camera_vendor_tags.h>
+#include <utils/Errors.h>
+#include <utils/Log.h>
+#include <utils/RefBase.h>
+
+#include <gtest/gtest.h>
+#include <stdint.h>
+
+using namespace android;
+
+enum {
+    BAD_TAG_ARRAY = 0xDEADBEEFu,
+    BAD_TAG = 0x8DEADBADu,
+};
+
+#define ARRAY_SIZE(a)      (sizeof(a) / sizeof((a)[0]))
+
+static bool ContainsTag(uint32_t* tagArray, size_t size, uint32_t tag) {
+    for (size_t i = 0; i < size; ++i) {
+        if (tag == tagArray[i]) return true;
+    }
+    return false;
+}
+
+#define EXPECT_CONTAINS_TAG(t, a) \
+    EXPECT_TRUE(ContainsTag(a, ARRAY_SIZE(a), t))
+
+#define ASSERT_NOT_NULL(x) \
+    ASSERT_TRUE((x) != NULL)
+
+extern "C" {
+
+static int default_get_tag_count(const vendor_tag_ops_t* vOps) {
+    return VENDOR_TAG_COUNT_ERR;
+}
+
+static void default_get_all_tags(const vendor_tag_ops_t* vOps, uint32_t* tagArray) {
+    //Noop
+}
+
+static const char* default_get_section_name(const vendor_tag_ops_t* vOps, uint32_t tag) {
+    return VENDOR_SECTION_NAME_ERR;
+}
+
+static const char* default_get_tag_name(const vendor_tag_ops_t* vOps, uint32_t tag) {
+    return VENDOR_TAG_NAME_ERR;
+}
+
+static int default_get_tag_type(const vendor_tag_ops_t* vOps, uint32_t tag) {
+    return VENDOR_TAG_TYPE_ERR;
+}
+
+} /*extern "C"*/
+
+// Set default vendor operations for a vendor_tag_ops struct
+static void FillWithDefaults(vendor_tag_ops_t* vOps) {
+    ASSERT_NOT_NULL(vOps);
+    vOps->get_tag_count = default_get_tag_count;
+    vOps->get_all_tags = default_get_all_tags;
+    vOps->get_section_name = default_get_section_name;
+    vOps->get_tag_name = default_get_tag_name;
+    vOps->get_tag_type = default_get_tag_type;
+}
+
+/**
+ * Test if values from VendorTagDescriptor methods match corresponding values
+ * from vendor_tag_ops functions.
+ */
+TEST(VendorTagDescriptorTest, ConsistentWithVendorTags) {
+    sp<VendorTagDescriptor> vDesc;
+    const vendor_tag_ops_t *vOps = &fakevendor_ops;
+    EXPECT_EQ(OK, VendorTagDescriptor::createDescriptorFromOps(vOps, /*out*/vDesc));
+
+    ASSERT_NOT_NULL(vDesc);
+
+    // Ensure reasonable tag count
+    int tagCount = vDesc->getTagCount();
+    EXPECT_EQ(tagCount, vOps->get_tag_count(vOps));
+
+    uint32_t descTagArray[tagCount];
+    uint32_t opsTagArray[tagCount];
+
+    // Get all tag ids
+    vDesc->getTagArray(descTagArray);
+    vOps->get_all_tags(vOps, opsTagArray);
+
+    ASSERT_NOT_NULL(descTagArray);
+    ASSERT_NOT_NULL(opsTagArray);
+
+    uint32_t tag;
+    for (int i = 0; i < tagCount; ++i) {
+        // For each tag id, check whether type, section name, tag name match
+        tag = descTagArray[i];
+        EXPECT_CONTAINS_TAG(tag, opsTagArray);
+        EXPECT_EQ(vDesc->getTagType(tag), vOps->get_tag_type(vOps, tag));
+        EXPECT_STREQ(vDesc->getSectionName(tag), vOps->get_section_name(vOps, tag));
+        EXPECT_STREQ(vDesc->getTagName(tag), vOps->get_tag_name(vOps, tag));
+    }
+}
+
+/**
+ * Test if values from VendorTagDescriptor methods stay consistent after being
+ * parcelled/unparcelled.
+ */
+TEST(VendorTagDescriptorTest, ConsistentAcrossParcel) {
+    sp<VendorTagDescriptor> vDescOriginal, vDescParceled;
+    const vendor_tag_ops_t *vOps = &fakevendor_ops;
+    EXPECT_EQ(OK, VendorTagDescriptor::createDescriptorFromOps(vOps, /*out*/vDescOriginal));
+
+    ASSERT_TRUE(vDescOriginal != NULL);
+
+    Parcel p;
+
+    // Check whether parcel read/write succeed
+    EXPECT_EQ(OK, vDescOriginal->writeToParcel(&p));
+    p.setDataPosition(0);
+    ASSERT_EQ(OK, VendorTagDescriptor::createFromParcel(&p, vDescParceled));
+
+    // Ensure consistent tag count
+    int tagCount = vDescOriginal->getTagCount();
+    ASSERT_EQ(tagCount, vDescParceled->getTagCount());
+
+    uint32_t descTagArray[tagCount];
+    uint32_t desc2TagArray[tagCount];
+
+    // Get all tag ids
+    vDescOriginal->getTagArray(descTagArray);
+    vDescParceled->getTagArray(desc2TagArray);
+
+    ASSERT_NOT_NULL(descTagArray);
+    ASSERT_NOT_NULL(desc2TagArray);
+
+    uint32_t tag;
+    for (int i = 0; i < tagCount; ++i) {
+        // For each tag id, check consistency between the two vendor tag
+        // descriptors for each type, section name, tag name
+        tag = descTagArray[i];
+        EXPECT_CONTAINS_TAG(tag, desc2TagArray);
+        EXPECT_EQ(vDescOriginal->getTagType(tag), vDescParceled->getTagType(tag));
+        EXPECT_STREQ(vDescOriginal->getSectionName(tag), vDescParceled->getSectionName(tag));
+        EXPECT_STREQ(vDescOriginal->getTagName(tag), vDescParceled->getTagName(tag));
+    }
+}
+
+/**
+ * Test defaults and error conditions.
+ */
+TEST(VendorTagDescriptorTest, ErrorConditions) {
+    sp<VendorTagDescriptor> vDesc;
+    vendor_tag_ops_t vOps;
+    FillWithDefaults(&vOps);
+
+    // Ensure create fails when using null vOps
+    EXPECT_EQ(BAD_VALUE, VendorTagDescriptor::createDescriptorFromOps(/*vOps*/NULL, vDesc));
+
+    // Ensure create works when there are no vtags defined in a well-formed vOps
+    ASSERT_EQ(OK, VendorTagDescriptor::createDescriptorFromOps(&vOps, vDesc));
+
+    // Ensure defaults are returned when no vtags are defined, or tag is unknown
+    EXPECT_EQ(VENDOR_TAG_COUNT_ERR, vDesc->getTagCount());
+    uint32_t* tagArray = reinterpret_cast<uint32_t*>(BAD_TAG_ARRAY);
+    uint32_t* testArray = tagArray;
+    vDesc->getTagArray(tagArray);
+    EXPECT_EQ(testArray, tagArray);
+    EXPECT_EQ(VENDOR_SECTION_NAME_ERR, vDesc->getSectionName(BAD_TAG));
+    EXPECT_EQ(VENDOR_TAG_NAME_ERR, vDesc->getTagName(BAD_TAG));
+    EXPECT_EQ(VENDOR_TAG_TYPE_ERR, vDesc->getTagType(BAD_TAG));
+
+    // Make sure global can be set/cleared
+    const vendor_tag_ops_t *fakeOps = &fakevendor_ops;
+    sp<VendorTagDescriptor> prevGlobal = VendorTagDescriptor::getGlobalVendorTagDescriptor();
+    VendorTagDescriptor::clearGlobalVendorTagDescriptor();
+
+    EXPECT_TRUE(VendorTagDescriptor::getGlobalVendorTagDescriptor() == NULL);
+    EXPECT_EQ(OK, VendorTagDescriptor::setAsGlobalVendorTagDescriptor(vDesc));
+    EXPECT_TRUE(VendorTagDescriptor::getGlobalVendorTagDescriptor() != NULL);
+    EXPECT_EQ(VENDOR_SECTION_NAME_ERR, vDesc->getSectionName(BAD_TAG));
+    EXPECT_EQ(OK, VendorTagDescriptor::setAsGlobalVendorTagDescriptor(prevGlobal));
+    EXPECT_EQ(prevGlobal, VendorTagDescriptor::getGlobalVendorTagDescriptor());
+}
+
diff --git a/cmds/screenrecord/Android.mk b/cmds/screenrecord/Android.mk
index 6747e60..6ee2884 100644
--- a/cmds/screenrecord/Android.mk
+++ b/cmds/screenrecord/Android.mk
@@ -41,4 +41,6 @@
 
 LOCAL_MODULE:= screenrecord
 
+LOCAL_32_BIT_ONLY := true
+
 include $(BUILD_EXECUTABLE)
diff --git a/cmds/screenrecord/FrameOutput.cpp b/cmds/screenrecord/FrameOutput.cpp
index b5cf2f9..06b1f70 100644
--- a/cmds/screenrecord/FrameOutput.cpp
+++ b/cmds/screenrecord/FrameOutput.cpp
@@ -67,8 +67,10 @@
         return UNKNOWN_ERROR;
     }
 
-    mBufferQueue = new BufferQueue(/*new GraphicBufferAlloc()*/);
-    mGlConsumer = new GLConsumer(mBufferQueue, mExtTextureName,
+    sp<IGraphicBufferProducer> producer;
+    sp<IGraphicBufferConsumer> consumer;
+    BufferQueue::createBufferQueue(&producer, &consumer);
+    mGlConsumer = new GLConsumer(consumer, mExtTextureName,
                 GL_TEXTURE_EXTERNAL_OES);
     mGlConsumer->setName(String8("virtual display"));
     mGlConsumer->setDefaultBufferSize(width, height);
@@ -79,7 +81,7 @@
 
     mPixelBuf = new uint8_t[width * height * kGlBytesPerPixel];
 
-    *pBufferProducer = mBufferQueue;
+    *pBufferProducer = producer;
 
     ALOGD("FrameOutput::createInputSurface OK");
     return NO_ERROR;
diff --git a/cmds/screenrecord/FrameOutput.h b/cmds/screenrecord/FrameOutput.h
index bb66e05..c1148d0 100644
--- a/cmds/screenrecord/FrameOutput.h
+++ b/cmds/screenrecord/FrameOutput.h
@@ -77,10 +77,6 @@
     // 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;
diff --git a/cmds/screenrecord/Overlay.cpp b/cmds/screenrecord/Overlay.cpp
index 2e98874..94f560d 100644
--- a/cmds/screenrecord/Overlay.cpp
+++ b/cmds/screenrecord/Overlay.cpp
@@ -84,7 +84,7 @@
     assert(mState == RUNNING);
 
     ALOGV("Overlay::start successful");
-    *pBufferProducer = mBufferQueue;
+    *pBufferProducer = mProducer;
     return NO_ERROR;
 }
 
@@ -169,8 +169,9 @@
         return UNKNOWN_ERROR;
     }
 
-    mBufferQueue = new BufferQueue(/*new GraphicBufferAlloc()*/);
-    mGlConsumer = new GLConsumer(mBufferQueue, mExtTextureName,
+    sp<IGraphicBufferConsumer> consumer;
+    BufferQueue::createBufferQueue(&mProducer, &consumer);
+    mGlConsumer = new GLConsumer(consumer, mExtTextureName,
                 GL_TEXTURE_EXTERNAL_OES);
     mGlConsumer->setName(String8("virtual display"));
     mGlConsumer->setDefaultBufferSize(width, height);
@@ -187,7 +188,7 @@
     ALOGV("Overlay::release_l");
     mOutputSurface.clear();
     mGlConsumer.clear();
-    mBufferQueue.clear();
+    mProducer.clear();
 
     mTexProgram.release();
     mExtTexProgram.release();
diff --git a/cmds/screenrecord/Overlay.h b/cmds/screenrecord/Overlay.h
index 48e48e0..b1b5c29 100644
--- a/cmds/screenrecord/Overlay.h
+++ b/cmds/screenrecord/Overlay.h
@@ -122,9 +122,9 @@
     // 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;
+    // Producer side of queue, passed into the virtual display.
+    // The consumer end feeds into our GLConsumer.
+    sp<IGraphicBufferProducer> mProducer;
 
     // This receives frames from the virtual display and makes them available
     // as an external texture.
diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk
index 561ce02..e2e389b 100644
--- a/cmds/stagefright/Android.mk
+++ b/cmds/stagefright/Android.mk
@@ -23,6 +23,8 @@
 
 LOCAL_MODULE:= stagefright
 
+LOCAL_32_BIT_ONLY := true
+
 include $(BUILD_EXECUTABLE)
 
 ################################################################################
@@ -46,6 +48,8 @@
 
 LOCAL_MODULE:= record
 
+LOCAL_32_BIT_ONLY := true
+
 include $(BUILD_EXECUTABLE)
 
 ################################################################################
@@ -69,6 +73,8 @@
 
 LOCAL_MODULE:= recordvideo
 
+LOCAL_32_BIT_ONLY := true
+
 include $(BUILD_EXECUTABLE)
 
 
@@ -93,6 +99,8 @@
 
 LOCAL_MODULE:= audioloop
 
+LOCAL_32_BIT_ONLY := true
+
 include $(BUILD_EXECUTABLE)
 
 ################################################################################
@@ -116,6 +124,8 @@
 
 LOCAL_MODULE:= stream
 
+LOCAL_32_BIT_ONLY := true
+
 include $(BUILD_EXECUTABLE)
 
 ################################################################################
@@ -139,6 +149,8 @@
 
 LOCAL_MODULE:= sf2
 
+LOCAL_32_BIT_ONLY := true
+
 include $(BUILD_EXECUTABLE)
 
 ################################################################################
@@ -163,6 +175,8 @@
 
 LOCAL_MODULE:= codec
 
+LOCAL_32_BIT_ONLY := true
+
 include $(BUILD_EXECUTABLE)
 
 ################################################################################
@@ -186,4 +200,6 @@
 
 LOCAL_MODULE:= muxer
 
+LOCAL_32_BIT_ONLY := true
+
 include $(BUILD_EXECUTABLE)
diff --git a/drm/drmserver/Android.mk b/drm/drmserver/Android.mk
index dc973da..aa0ab9b 100644
--- a/drm/drmserver/Android.mk
+++ b/drm/drmserver/Android.mk
@@ -39,4 +39,6 @@
 
 LOCAL_MODULE_TAGS := optional
 
+LOCAL_32_BIT_ONLY := true
+
 include $(BUILD_EXECUTABLE)
diff --git a/include/camera/ICameraService.h b/include/camera/ICameraService.h
index f342122..6e48f22 100644
--- a/include/camera/ICameraService.h
+++ b/include/camera/ICameraService.h
@@ -31,6 +31,7 @@
 class ICameraDeviceUser;
 class ICameraDeviceCallbacks;
 class CameraMetadata;
+class VendorTagDescriptor;
 
 class ICameraService : public IInterface
 {
@@ -47,6 +48,7 @@
         ADD_LISTENER,
         REMOVE_LISTENER,
         GET_CAMERA_CHARACTERISTICS,
+        GET_CAMERA_VENDOR_TAG_DESCRIPTOR,
     };
 
     enum {
@@ -58,10 +60,16 @@
 
     virtual int32_t  getNumberOfCameras() = 0;
     virtual status_t getCameraInfo(int cameraId,
-                                          struct CameraInfo* cameraInfo) = 0;
+            /*out*/
+            struct CameraInfo* cameraInfo) = 0;
 
     virtual status_t getCameraCharacteristics(int cameraId,
-                                              CameraMetadata* cameraInfo) = 0;
+            /*out*/
+            CameraMetadata* cameraInfo) = 0;
+
+    virtual status_t getCameraVendorTagDescriptor(
+            /*out*/
+            sp<VendorTagDescriptor>& desc) = 0;
 
     // Returns 'OK' if operation succeeded
     // - Errors: ALREADY_EXISTS if the listener was already added
diff --git a/include/camera/VendorTagDescriptor.h b/include/camera/VendorTagDescriptor.h
new file mode 100644
index 0000000..ea21d31
--- /dev/null
+++ b/include/camera/VendorTagDescriptor.h
@@ -0,0 +1,124 @@
+/*
+ * 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 VENDOR_TAG_DESCRIPTOR_H
+
+#include <utils/KeyedVector.h>
+#include <utils/String8.h>
+#include <utils/RefBase.h>
+#include <system/camera_vendor_tags.h>
+
+#include <stdint.h>
+
+namespace android {
+
+class Parcel;
+
+/**
+ * VendorTagDescriptor objects are parcelable containers for the vendor tag
+ * definitions provided, and are typically used to pass the vendor tag
+ * information enumerated by the HAL to clients of the camera service.
+ */
+class VendorTagDescriptor
+        : public LightRefBase<VendorTagDescriptor> {
+    public:
+        virtual ~VendorTagDescriptor();
+
+        /**
+         * The following 'get*' methods implement the corresponding
+         * functions defined in
+         * system/media/camera/include/system/camera_vendor_tags.h
+         */
+
+        // Returns the number of vendor tags defined.
+        int getTagCount() const;
+
+        // Returns an array containing the id's of vendor tags defined.
+        void getTagArray(uint32_t* tagArray) const;
+
+        // Returns the section name string for a given vendor tag id.
+        const char* getSectionName(uint32_t tag) const;
+
+        // Returns the tag name string for a given vendor tag id.
+        const char* getTagName(uint32_t tag) const;
+
+        // Returns the tag type for a given vendor tag id.
+        int getTagType(uint32_t tag) const;
+
+        /**
+         * Write the VendorTagDescriptor object into the given parcel.
+         *
+         * Returns OK on success, or a negative error code.
+         */
+        status_t writeToParcel(
+                /*out*/
+                Parcel* parcel) const;
+
+        // Static methods:
+
+        /**
+         * Create a VendorTagDescriptor object from the given parcel.
+         *
+         * Returns OK on success, or a negative error code.
+         */
+        static status_t createFromParcel(const Parcel* parcel,
+                /*out*/
+                sp<VendorTagDescriptor>& descriptor);
+
+        /**
+         * Create a VendorTagDescriptor object from the given vendor_tag_ops_t
+         * struct.
+         *
+         * Returns OK on success, or a negative error code.
+         */
+        static status_t createDescriptorFromOps(const vendor_tag_ops_t* vOps,
+                /*out*/
+                sp<VendorTagDescriptor>& descriptor);
+
+        /**
+         * Sets the global vendor tag descriptor to use for this process.
+         * Camera metadata operations that access vendor tags will use the
+         * vendor tag definitions set this way.
+         *
+         * Returns OK on success, or a negative error code.
+         */
+        static status_t setAsGlobalVendorTagDescriptor(const sp<VendorTagDescriptor>& desc);
+
+        /**
+         * Clears the global vendor tag descriptor used by this process.
+         */
+        static void clearGlobalVendorTagDescriptor();
+
+        /**
+         * Returns the global vendor tag descriptor used by this process.
+         * This will contain NULL if no vendor tags are defined.
+         */
+        static sp<VendorTagDescriptor> getGlobalVendorTagDescriptor();
+    protected:
+        VendorTagDescriptor();
+        KeyedVector<uint32_t, String8> mTagToNameMap;
+        KeyedVector<uint32_t, String8> mTagToSectionMap;
+        KeyedVector<uint32_t, int32_t> mTagToTypeMap;
+        // must be int32_t to be compatible with Parcel::writeInt32
+        int32_t mTagCount;
+    private:
+        vendor_tag_ops mVendorOps;
+};
+
+} /* namespace android */
+
+#define VENDOR_TAG_DESCRIPTOR_H
+#endif /* VENDOR_TAG_DESCRIPTOR_H */
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 716eaa1..647748b 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -748,7 +748,6 @@
     sp<AudioTrackClientProxy>       mProxy;         // primary owner of the memory
 
     bool                    mInUnderrun;            // whether track is currently in underrun state
-    String8                 mName;                  // server's name for this IAudioTrack
     uint32_t                mPausedPosition;
 
 private:
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 7c5f33a..9101f06 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -73,10 +73,6 @@
                                 audio_io_handle_t output,
                                 pid_t tid,  // -1 means unused, otherwise must be valid non-0
                                 int *sessionId,
-                                // input: ignored
-                                // output: server's description of IAudioTrack for display in logs.
-                                // Don't attempt to parse, as the format could change.
-                                String8& name,
                                 int clientUid,
                                 status_t *status) = 0;
 
diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h
index 36f2a67..863a7d5 100644
--- a/include/media/stagefright/ACodec.h
+++ b/include/media/stagefright/ACodec.h
@@ -67,8 +67,6 @@
 
     void signalRequestIDRFrame();
 
-    bool isConfiguredForAdaptivePlayback() { return mIsConfiguredForAdaptivePlayback; }
-
     struct PortDescription : public RefBase {
         size_t countBuffers();
         IOMX::buffer_id bufferIDAt(size_t index) const;
@@ -178,6 +176,8 @@
     sp<MemoryDealer> mDealer[2];
 
     sp<ANativeWindow> mNativeWindow;
+    sp<AMessage> mInputFormat;
+    sp<AMessage> mOutputFormat;
 
     Vector<BufferInfo> mBuffers[2];
     bool mPortEOS[2];
@@ -189,7 +189,6 @@
     bool mIsEncoder;
     bool mUseMetadataOnEncoderOutput;
     bool mShutdownInProgress;
-    bool mIsConfiguredForAdaptivePlayback;
 
     // If "mKeepComponentAllocated" we only transition back to Loaded state
     // and do not release the component instance.
@@ -203,6 +202,7 @@
     unsigned mDequeueCounter;
     bool mStoreMetaDataInOutputBuffers;
     int32_t mMetaDataBuffersToSubmit;
+    size_t mNumUndequeuedBuffers;
 
     int64_t mRepeatFrameDelayUs;
     int64_t mMaxPtsGapUs;
@@ -305,6 +305,7 @@
     void processDeferredMessages();
 
     void sendFormatChange(const sp<AMessage> &reply);
+    status_t getPortFormat(OMX_U32 portIndex, sp<AMessage> &notify);
 
     void signalError(
             OMX_ERRORTYPE error = OMX_ErrorUndefined,
diff --git a/include/media/stagefright/MediaCodec.h b/include/media/stagefright/MediaCodec.h
index 76aa503..276543b 100644
--- a/include/media/stagefright/MediaCodec.h
+++ b/include/media/stagefright/MediaCodec.h
@@ -106,6 +106,7 @@
     status_t signalEndOfInputStream();
 
     status_t getOutputFormat(sp<AMessage> *format) const;
+    status_t getInputFormat(sp<AMessage> *format) const;
 
     status_t getInputBuffers(Vector<sp<ABuffer> > *buffers) const;
     status_t getOutputBuffers(Vector<sp<ABuffer> > *buffers) const;
@@ -159,6 +160,7 @@
         kWhatGetBuffers                     = 'getB',
         kWhatFlush                          = 'flus',
         kWhatGetOutputFormat                = 'getO',
+        kWhatGetInputFormat                 = 'getI',
         kWhatDequeueInputTimedOut           = 'dITO',
         kWhatDequeueOutputTimedOut          = 'dOTO',
         kWhatCodecNotify                    = 'codc',
@@ -199,6 +201,7 @@
     sp<Surface> mNativeWindow;
     SoftwareRenderer *mSoftRenderer;
     sp<AMessage> mOutputFormat;
+    sp<AMessage> mInputFormat;
 
     List<size_t> mAvailPortBuffers[2];
     Vector<BufferInfo> mPortBuffers[2];
diff --git a/libvideoeditor/lvpp/Android.mk b/libvideoeditor/lvpp/Android.mk
index 860d351..77a21ac 100755
--- a/libvideoeditor/lvpp/Android.mk
+++ b/libvideoeditor/lvpp/Android.mk
@@ -99,6 +99,8 @@
     -DUSE_STAGEFRIGHT_READERS \
     -DUSE_STAGEFRIGHT_3GPP_READER
 
+LOCAL_32_BIT_ONLY := true
+
 include $(BUILD_SHARED_LIBRARY)
 
 #include $(call all-makefiles-under,$(LOCAL_PATH))
diff --git a/libvideoeditor/lvpp/NativeWindowRenderer.cpp b/libvideoeditor/lvpp/NativeWindowRenderer.cpp
index 8b362ef..be0f747 100755
--- a/libvideoeditor/lvpp/NativeWindowRenderer.cpp
+++ b/libvideoeditor/lvpp/NativeWindowRenderer.cpp
@@ -568,9 +568,11 @@
 RenderInput::RenderInput(NativeWindowRenderer* renderer, GLuint textureId)
     : mRenderer(renderer)
     , mTextureId(textureId) {
-    sp<BufferQueue> bq = new BufferQueue();
-    mST = new GLConsumer(bq, mTextureId);
-    mSTC = new Surface(bq);
+    sp<IGraphicBufferProducer> producer;
+    sp<IGraphicBufferConsumer> consumer;
+    BufferQueue::createBufferQueue(&producer, &consumer);
+    mST = new GLConsumer(consumer, mTextureId);
+    mSTC = new Surface(producer);
     native_window_connect(mSTC.get(), NATIVE_WINDOW_API_MEDIA);
 }
 
diff --git a/libvideoeditor/vss/src/Android.mk b/libvideoeditor/vss/src/Android.mk
index 0caa15b..8856c41 100755
--- a/libvideoeditor/vss/src/Android.mk
+++ b/libvideoeditor/vss/src/Android.mk
@@ -96,4 +96,6 @@
     -DM4xVSS_RESERVED_MOOV_DISK_SPACEno \
     -DDECODE_GIF_ON_SAVING
 
+LOCAL_32_BIT_ONLY := true
+
 include $(BUILD_SHARED_LIBRARY)
diff --git a/libvideoeditor/vss/stagefrightshells/src/Android.mk b/libvideoeditor/vss/stagefrightshells/src/Android.mk
index 9188942..a060c0d 100755
--- a/libvideoeditor/vss/stagefrightshells/src/Android.mk
+++ b/libvideoeditor/vss/stagefrightshells/src/Android.mk
@@ -64,4 +64,6 @@
 
 LOCAL_MODULE_TAGS := optional
 
+LOCAL_32_BIT_ONLY := true
+
 include $(BUILD_STATIC_LIBRARY)
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 60ed626..20c1cdb 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -1024,7 +1024,6 @@
                                                       output,
                                                       tid,
                                                       &mSessionId,
-                                                      mName,
                                                       mClientUid,
                                                       &status);
 
@@ -1281,8 +1280,7 @@
     if (mState == STATE_ACTIVE) {
         audio_track_cblk_t* cblk = mCblk;
         if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) {
-            ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting",
-                    this, mName.string());
+            ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this);
             // FIXME ignoring status
             mAudioTrack->start();
         }
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index a9a9f1a..762681e 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -95,7 +95,6 @@
                                 audio_io_handle_t output,
                                 pid_t tid,
                                 int *sessionId,
-                                String8& name,
                                 int clientUid,
                                 status_t *status)
     {
@@ -140,7 +139,6 @@
             if (sessionId != NULL) {
                 *sessionId = lSessionId;
             }
-            name = reply.readString8();
             lStatus = reply.readInt32();
             track = interface_cast<IAudioTrack>(reply.readStrongBinder());
             if (lStatus == NO_ERROR) {
@@ -808,7 +806,6 @@
             pid_t tid = (pid_t) data.readInt32();
             int sessionId = data.readInt32();
             int clientUid = data.readInt32();
-            String8 name;
             status_t status;
             sp<IAudioTrack> track;
             if ((haveSharedBuffer && (buffer == 0)) ||
@@ -819,13 +816,12 @@
                 track = createTrack(
                         (audio_stream_type_t) streamType, sampleRate, format,
                         channelMask, &frameCount, &flags, buffer, output, tid,
-                        &sessionId, name, clientUid, &status);
+                        &sessionId, clientUid, &status);
                 LOG_ALWAYS_FATAL_IF((track != 0) != (status == NO_ERROR));
             }
             reply->writeInt32(frameCount);
             reply->writeInt32(flags);
             reply->writeInt32(sessionId);
-            reply->writeString8(name);
             reply->writeInt32(status);
             reply->writeStrongBinder(track->asBinder());
             return NO_ERROR;
diff --git a/media/libmediaplayerservice/Android.mk b/media/libmediaplayerservice/Android.mk
index 4189a5e..caf2dfc 100644
--- a/media/libmediaplayerservice/Android.mk
+++ b/media/libmediaplayerservice/Android.mk
@@ -53,6 +53,8 @@
 
 LOCAL_MODULE:= libmediaplayerservice
 
+LOCAL_32_BIT_ONLY := true
+
 include $(BUILD_SHARED_LIBRARY)
 
 include $(call all-makefiles-under,$(LOCAL_PATH))
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index a750ad0..d8d939a 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -31,13 +31,10 @@
 
 #include "ATSParser.h"
 
-#include "SoftwareRenderer.h"
-
 #include <media/stagefright/foundation/hexdump.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/ACodec.h>
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/MediaErrors.h>
 #include <media/stagefright/MetaData.h>
@@ -146,7 +143,6 @@
     : mUIDValid(false),
       mSourceFlags(0),
       mVideoIsAVC(false),
-      mNeedsSwRenderer(false),
       mAudioEOS(false),
       mVideoEOS(false),
       mScanSourcesPending(false),
@@ -442,7 +438,6 @@
             ALOGV("kWhatStart");
 
             mVideoIsAVC = false;
-            mNeedsSwRenderer = false;
             mAudioEOS = false;
             mVideoEOS = false;
             mSkipRenderingAudioUntilMediaTimeUs = -1;
@@ -533,24 +528,21 @@
         {
             bool audio = msg->what() == kWhatAudioNotify;
 
-            sp<AMessage> codecRequest;
-            CHECK(msg->findMessage("codec-request", &codecRequest));
-
             int32_t what;
-            CHECK(codecRequest->findInt32("what", &what));
+            CHECK(msg->findInt32("what", &what));
 
-            if (what == ACodec::kWhatFillThisBuffer) {
+            if (what == Decoder::kWhatFillThisBuffer) {
                 status_t err = feedDecoderInputData(
-                        audio, codecRequest);
+                        audio, msg);
 
                 if (err == -EWOULDBLOCK) {
                     if (mSource->feedMoreTSData() == OK) {
                         msg->post(10000ll);
                     }
                 }
-            } else if (what == ACodec::kWhatEOS) {
+            } else if (what == Decoder::kWhatEOS) {
                 int32_t err;
-                CHECK(codecRequest->findInt32("err", &err));
+                CHECK(msg->findInt32("err", &err));
 
                 if (err == ERROR_END_OF_STREAM) {
                     ALOGV("got %s decoder EOS", audio ? "audio" : "video");
@@ -561,7 +553,7 @@
                 }
 
                 mRenderer->queueEOS(audio, err);
-            } else if (what == ACodec::kWhatFlushCompleted) {
+            } else if (what == Decoder::kWhatFlushCompleted) {
                 bool needShutdown;
 
                 if (audio) {
@@ -590,14 +582,17 @@
                 }
 
                 finishFlushIfPossible();
-            } else if (what == ACodec::kWhatOutputFormatChanged) {
+            } else if (what == Decoder::kWhatOutputFormatChanged) {
+                sp<AMessage> format;
+                CHECK(msg->findMessage("format", &format));
+
                 if (audio) {
                     int32_t numChannels;
-                    CHECK(codecRequest->findInt32(
+                    CHECK(format->findInt32(
                                 "channel-count", &numChannels));
 
                     int32_t sampleRate;
-                    CHECK(codecRequest->findInt32("sample-rate", &sampleRate));
+                    CHECK(format->findInt32("sample-rate", &sampleRate));
 
                     ALOGV("Audio output format changed to %d Hz, %d channels",
                          sampleRate, numChannels);
@@ -621,7 +616,7 @@
                     }
 
                     int32_t channelMask;
-                    if (!codecRequest->findInt32("channel-mask", &channelMask)) {
+                    if (!format->findInt32("channel-mask", &channelMask)) {
                         channelMask = CHANNEL_MASK_USE_CHANNEL_ORDER;
                     }
 
@@ -642,11 +637,11 @@
                     // video
 
                     int32_t width, height;
-                    CHECK(codecRequest->findInt32("width", &width));
-                    CHECK(codecRequest->findInt32("height", &height));
+                    CHECK(format->findInt32("width", &width));
+                    CHECK(format->findInt32("height", &height));
 
                     int32_t cropLeft, cropTop, cropRight, cropBottom;
-                    CHECK(codecRequest->findRect(
+                    CHECK(format->findRect(
                                 "crop",
                                 &cropLeft, &cropTop, &cropRight, &cropBottom));
 
@@ -679,22 +674,8 @@
 
                     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) {
+            } else if (what == Decoder::kWhatShutdownCompleted) {
                 ALOGV("%s shutdown completed", audio ? "audio" : "video");
                 if (audio) {
                     mAudioDecoder.clear();
@@ -709,22 +690,15 @@
                 }
 
                 finishFlushIfPossible();
-            } else if (what == ACodec::kWhatError) {
+            } else if (what == Decoder::kWhatError) {
                 ALOGE("Received error from %s decoder, aborting playback.",
                      audio ? "audio" : "video");
 
                 mRenderer->queueEOS(audio, UNKNOWN_ERROR);
-            } else if (what == ACodec::kWhatDrainThisBuffer) {
-                renderBuffer(audio, codecRequest);
-            } 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'.",
+            } else if (what == Decoder::kWhatDrainThisBuffer) {
+                renderBuffer(audio, msg);
+            } else {
+                ALOGV("Unhandled decoder notification %d '%c%c%c%c'.",
                       what,
                       what >> 24,
                       (what >> 16) & 0xff,
@@ -925,8 +899,7 @@
 
     *decoder = audio ? new Decoder(notify) :
                        new Decoder(notify, mNativeWindow);
-    looper()->registerHandler(*decoder);
-
+    (*decoder)->init();
     (*decoder)->configure(format);
 
     return OK;
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.h b/media/libmediaplayerservice/nuplayer/NuPlayer.h
index 9dfe4a0..f1d3d55 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.h
@@ -24,7 +24,6 @@
 
 namespace android {
 
-struct ACodec;
 struct MetaData;
 struct NuPlayerDriver;
 
@@ -118,7 +117,6 @@
     sp<MediaPlayerBase::AudioSink> mAudioSink;
     sp<Decoder> mVideoDecoder;
     bool mVideoIsAVC;
-    bool mNeedsSwRenderer;
     sp<Decoder> mAudioDecoder;
     sp<Renderer> mRenderer;
 
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
index 2423fd5..469c9ca 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
@@ -17,14 +17,17 @@
 //#define LOG_NDEBUG 0
 #define LOG_TAG "NuPlayerDecoder"
 #include <utils/Log.h>
+#include <inttypes.h>
 
 #include "NuPlayerDecoder.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/ACodec.h>
+#include <media/stagefright/MediaCodec.h>
 #include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaErrors.h>
 
 namespace android {
 
@@ -32,70 +35,404 @@
         const sp<AMessage> &notify,
         const sp<NativeWindowWrapper> &nativeWindow)
     : mNotify(notify),
-      mNativeWindow(nativeWindow) {
+      mNativeWindow(nativeWindow),
+      mBufferGeneration(0),
+      mComponentName("decoder") {
+    // Every decoder has its own looper because MediaCodec operations
+    // are blocking, but NuPlayer needs asynchronous operations.
+    mDecoderLooper = new ALooper;
+    mDecoderLooper->setName("NuPlayerDecoder");
+    mDecoderLooper->start(false, false, ANDROID_PRIORITY_AUDIO);
+
+    mCodecLooper = new ALooper;
+    mCodecLooper->setName("NuPlayerDecoder-MC");
+    mCodecLooper->start(false, false, ANDROID_PRIORITY_AUDIO);
 }
 
 NuPlayer::Decoder::~Decoder() {
 }
 
-void NuPlayer::Decoder::configure(const sp<AMessage> &format) {
+void NuPlayer::Decoder::onConfigure(const sp<AMessage> &format) {
     CHECK(mCodec == NULL);
 
+    ++mBufferGeneration;
+
     AString mime;
     CHECK(format->findString("mime", &mime));
 
-    sp<AMessage> notifyMsg =
-        new AMessage(kWhatCodecNotify, id());
-
-    mCSDIndex = 0;
-    for (size_t i = 0;; ++i) {
-        sp<ABuffer> csd;
-        if (!format->findBuffer(StringPrintf("csd-%d", i).c_str(), &csd)) {
-            break;
-        }
-
-        mCSD.push(csd);
+    sp<Surface> surface = NULL;
+    if (mNativeWindow != NULL) {
+        surface = mNativeWindow->getSurfaceTextureClient();
     }
 
+    mComponentName = mime;
+    mComponentName.append(" decoder");
+    ALOGV("[%s] onConfigure (surface=%p)", mComponentName.c_str(), surface.get());
+
+    mCodec = MediaCodec::CreateByType(mCodecLooper, mime.c_str(), false /* encoder */);
+    if (mCodec == NULL) {
+        ALOGE("Failed to create %s decoder", mime.c_str());
+        handleError(UNKNOWN_ERROR);
+        return;
+    }
+
+    mCodec->getName(&mComponentName);
+
     if (mNativeWindow != NULL) {
-        format->setObject("native-window", mNativeWindow);
+        // disconnect from surface as MediaCodec will reconnect
+        CHECK_EQ((int)NO_ERROR,
+                native_window_api_disconnect(
+                        surface.get(),
+                        NATIVE_WINDOW_API_MEDIA));
+    }
+    status_t err = mCodec->configure(
+            format, surface, NULL /* crypto */, 0 /* flags */);
+    if (err != OK) {
+        ALOGE("Failed to configure %s decoder (err=%d)", mComponentName.c_str(), err);
+        handleError(err);
+        return;
+    }
+    // the following should work in configured state
+    CHECK_EQ((status_t)OK, mCodec->getOutputFormat(&mOutputFormat));
+    CHECK_EQ((status_t)OK, mCodec->getInputFormat(&mInputFormat));
+
+    err = mCodec->start();
+    if (err != OK) {
+        ALOGE("Failed to start %s decoder (err=%d)", mComponentName.c_str(), err);
+        handleError(err);
+        return;
     }
 
-    // Current video decoders do not return from OMX_FillThisBuffer
-    // quickly, violating the OpenMAX specs, until that is remedied
-    // we need to invest in an extra looper to free the main event
-    // queue.
-    bool needDedicatedLooper = !strncasecmp(mime.c_str(), "video/", 6);
+    // the following should work after start
+    CHECK_EQ((status_t)OK, mCodec->getInputBuffers(&mInputBuffers));
+    CHECK_EQ((status_t)OK, mCodec->getOutputBuffers(&mOutputBuffers));
+    ALOGV("[%s] got %zu input and %zu output buffers",
+            mComponentName.c_str(),
+            mInputBuffers.size(),
+            mOutputBuffers.size());
 
-    mFormat = format;
-    mCodec = new ACodec;
+    requestCodecNotification();
+}
 
-    if (needDedicatedLooper && mCodecLooper == NULL) {
-        mCodecLooper = new ALooper;
-        mCodecLooper->setName("NuPlayerDecoder");
-        mCodecLooper->start(false, false, ANDROID_PRIORITY_AUDIO);
+void NuPlayer::Decoder::requestCodecNotification() {
+    if (mCodec != NULL) {
+        sp<AMessage> reply = new AMessage(kWhatCodecNotify, id());
+        reply->setInt32("generation", mBufferGeneration);
+        mCodec->requestActivityNotification(reply);
+    }
+}
+
+bool NuPlayer::Decoder::isStaleReply(const sp<AMessage> &msg) {
+    int32_t generation;
+    CHECK(msg->findInt32("generation", &generation));
+    return generation != mBufferGeneration;
+}
+
+void NuPlayer::Decoder::init() {
+    mDecoderLooper->registerHandler(this);
+}
+
+void NuPlayer::Decoder::configure(const sp<AMessage> &format) {
+    sp<AMessage> msg = new AMessage(kWhatConfigure, id());
+    msg->setMessage("format", format);
+    msg->post();
+}
+
+void NuPlayer::Decoder::handleError(int32_t err)
+{
+    sp<AMessage> notify = mNotify->dup();
+    notify->setInt32("what", kWhatError);
+    notify->setInt32("err", err);
+    notify->post();
+}
+
+bool NuPlayer::Decoder::handleAnInputBuffer() {
+    size_t bufferIx = -1;
+    status_t res = mCodec->dequeueInputBuffer(&bufferIx);
+    ALOGV("[%s] dequeued input: %d",
+            mComponentName.c_str(), res == OK ? (int)bufferIx : res);
+    if (res != OK) {
+        if (res != -EAGAIN) {
+            handleError(res);
+        }
+        return false;
     }
 
-    (needDedicatedLooper ? mCodecLooper : looper())->registerHandler(mCodec);
+    CHECK_LT(bufferIx, mInputBuffers.size());
 
-    mCodec->setNotificationMessage(notifyMsg);
-    mCodec->initiateSetup(format);
+    sp<AMessage> reply = new AMessage(kWhatInputBufferFilled, id());
+    reply->setSize("buffer-ix", bufferIx);
+    reply->setInt32("generation", mBufferGeneration);
+
+    sp<AMessage> notify = mNotify->dup();
+    notify->setInt32("what", kWhatFillThisBuffer);
+    notify->setBuffer("buffer", mInputBuffers[bufferIx]);
+    notify->setMessage("reply", reply);
+    notify->post();
+    return true;
+}
+
+void android::NuPlayer::Decoder::onInputBufferFilled(const sp<AMessage> &msg) {
+    size_t bufferIx;
+    CHECK(msg->findSize("buffer-ix", &bufferIx));
+    CHECK_LT(bufferIx, mInputBuffers.size());
+    sp<ABuffer> codecBuffer = mInputBuffers[bufferIx];
+
+    sp<ABuffer> buffer;
+    bool hasBuffer = msg->findBuffer("buffer", &buffer);
+    if (buffer == NULL /* includes !hasBuffer */) {
+        int32_t streamErr = ERROR_END_OF_STREAM;
+        CHECK(msg->findInt32("err", &streamErr) || !hasBuffer);
+
+        if (streamErr == OK) {
+            /* buffers are returned to hold on to */
+            return;
+        }
+
+        // attempt to queue EOS
+        status_t err = mCodec->queueInputBuffer(
+                bufferIx,
+                0,
+                0,
+                0,
+                MediaCodec::BUFFER_FLAG_EOS);
+        if (streamErr == ERROR_END_OF_STREAM && err != OK) {
+            streamErr = err;
+            // err will not be ERROR_END_OF_STREAM
+        }
+
+        if (streamErr != ERROR_END_OF_STREAM) {
+            handleError(streamErr);
+        }
+    } else {
+        int64_t timeUs = 0;
+        uint32_t flags = 0;
+        CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
+
+        int32_t eos;
+        // we do not expect CODECCONFIG or SYNCFRAME for decoder
+        if (buffer->meta()->findInt32("eos", &eos) && eos) {
+            flags |= MediaCodec::BUFFER_FLAG_EOS;
+        }
+
+        // copy into codec buffer
+        if (buffer != codecBuffer) {
+            CHECK_LE(buffer->size(), codecBuffer->capacity());
+            codecBuffer->setRange(0, buffer->size());
+            memcpy(codecBuffer->data(), buffer->data(), buffer->size());
+        }
+
+        status_t err = mCodec->queueInputBuffer(
+                        bufferIx,
+                        codecBuffer->offset(),
+                        codecBuffer->size(),
+                        timeUs,
+                        flags);
+        if (err != OK) {
+            ALOGE("Failed to queue input buffer for %s (err=%d)",
+                    mComponentName.c_str(), err);
+            handleError(err);
+        }
+    }
+}
+
+bool NuPlayer::Decoder::handleAnOutputBuffer() {
+    size_t bufferIx = -1;
+    size_t offset;
+    size_t size;
+    int64_t timeUs;
+    uint32_t flags;
+    status_t res = mCodec->dequeueOutputBuffer(
+            &bufferIx, &offset, &size, &timeUs, &flags);
+
+    if (res != OK) {
+        ALOGV("[%s] dequeued output: %d", mComponentName.c_str(), res);
+    } else {
+        ALOGV("[%s] dequeued output: %d (time=%lld flags=%" PRIu32 ")",
+                mComponentName.c_str(), (int)bufferIx, timeUs, flags);
+    }
+
+    if (res == INFO_OUTPUT_BUFFERS_CHANGED) {
+        res = mCodec->getOutputBuffers(&mOutputBuffers);
+        if (res != OK) {
+            ALOGE("Failed to get output buffers for %s after INFO event (err=%d)",
+                    mComponentName.c_str(), res);
+            handleError(res);
+            return false;
+        }
+        // NuPlayer ignores this
+        return true;
+    } else if (res == INFO_FORMAT_CHANGED) {
+        sp<AMessage> format = new AMessage();
+        res = mCodec->getOutputFormat(&format);
+        if (res != OK) {
+            ALOGE("Failed to get output format for %s after INFO event (err=%d)",
+                    mComponentName.c_str(), res);
+            handleError(res);
+            return false;
+        }
+
+        sp<AMessage> notify = mNotify->dup();
+        notify->setInt32("what", kWhatOutputFormatChanged);
+        notify->setMessage("format", format);
+        notify->post();
+        return true;
+    } else if (res == INFO_DISCONTINUITY) {
+        // nothing to do
+        return true;
+    } else if (res != OK) {
+        if (res != -EAGAIN) {
+            handleError(res);
+        }
+        return false;
+    }
+
+    CHECK_LT(bufferIx, mOutputBuffers.size());
+    sp<ABuffer> buffer = mOutputBuffers[bufferIx];
+    buffer->setRange(offset, size);
+    buffer->meta()->clear();
+    buffer->meta()->setInt64("timeUs", timeUs);
+    if (flags & MediaCodec::BUFFER_FLAG_EOS) {
+        buffer->meta()->setInt32("eos", true);
+    }
+    // we do not expect CODECCONFIG or SYNCFRAME for decoder
+
+    sp<AMessage> reply = new AMessage(kWhatRenderBuffer, id());
+    reply->setSize("buffer-ix", bufferIx);
+    reply->setInt32("generation", mBufferGeneration);
+
+    sp<AMessage> notify = mNotify->dup();
+    notify->setInt32("what", kWhatDrainThisBuffer);
+    notify->setBuffer("buffer", buffer);
+    notify->setMessage("reply", reply);
+    notify->post();
+
+    // FIXME: This should be handled after rendering is complete,
+    // but Renderer needs it now
+    if (flags & MediaCodec::BUFFER_FLAG_EOS) {
+        ALOGV("queueing eos [%s]", mComponentName.c_str());
+        sp<AMessage> notify = mNotify->dup();
+        notify->setInt32("what", kWhatEOS);
+        notify->setInt32("err", ERROR_END_OF_STREAM);
+        notify->post();
+    }
+    return true;
+}
+
+void NuPlayer::Decoder::onRenderBuffer(const sp<AMessage> &msg) {
+    status_t err;
+    int32_t render;
+    size_t bufferIx;
+    CHECK(msg->findSize("buffer-ix", &bufferIx));
+    if (msg->findInt32("render", &render) && render) {
+        err = mCodec->renderOutputBufferAndRelease(bufferIx);
+    } else {
+        err = mCodec->releaseOutputBuffer(bufferIx);
+    }
+    if (err != OK) {
+        ALOGE("failed to release output buffer for %s (err=%d)",
+                mComponentName.c_str(), err);
+        handleError(err);
+    }
+}
+
+void NuPlayer::Decoder::onFlush() {
+    status_t err = OK;
+    if (mCodec != NULL) {
+        err = mCodec->flush();
+        ++mBufferGeneration;
+    }
+
+    if (err != OK) {
+        ALOGE("failed to flush %s (err=%d)", mComponentName.c_str(), err);
+        handleError(err);
+        return;
+    }
+
+    sp<AMessage> notify = mNotify->dup();
+    notify->setInt32("what", kWhatFlushCompleted);
+    notify->post();
+}
+
+void NuPlayer::Decoder::onShutdown() {
+    status_t err = OK;
+    if (mCodec != NULL) {
+        err = mCodec->release();
+        mCodec = NULL;
+        ++mBufferGeneration;
+
+        if (mNativeWindow != NULL) {
+            // reconnect to surface as MediaCodec disconnected from it
+            CHECK_EQ((int)NO_ERROR,
+                    native_window_api_connect(
+                            mNativeWindow->getNativeWindow().get(),
+                            NATIVE_WINDOW_API_MEDIA));
+        }
+        mComponentName = "decoder";
+    }
+
+    if (err != OK) {
+        ALOGE("failed to release %s (err=%d)", mComponentName.c_str(), err);
+        handleError(err);
+        return;
+    }
+
+    sp<AMessage> notify = mNotify->dup();
+    notify->setInt32("what", kWhatShutdownCompleted);
+    notify->post();
 }
 
 void NuPlayer::Decoder::onMessageReceived(const sp<AMessage> &msg) {
+    ALOGV("[%s] onMessage: %s", mComponentName.c_str(), msg->debugString().c_str());
+
     switch (msg->what()) {
+        case kWhatConfigure:
+        {
+            sp<AMessage> format;
+            CHECK(msg->findMessage("format", &format));
+            onConfigure(format);
+            break;
+        }
+
         case kWhatCodecNotify:
         {
-            int32_t what;
-            CHECK(msg->findInt32("what", &what));
+            if (!isStaleReply(msg)) {
+                while (handleAnInputBuffer()) {
+                }
 
-            if (what == ACodec::kWhatFillThisBuffer) {
-                onFillThisBuffer(msg);
-            } else {
-                sp<AMessage> notify = mNotify->dup();
-                notify->setMessage("codec-request", msg);
-                notify->post();
+                while (handleAnOutputBuffer()) {
+                }
             }
+
+            requestCodecNotification();
+            break;
+        }
+
+        case kWhatInputBufferFilled:
+        {
+            if (!isStaleReply(msg)) {
+                onInputBufferFilled(msg);
+            }
+            break;
+        }
+
+        case kWhatRenderBuffer:
+        {
+            if (!isStaleReply(msg)) {
+                onRenderBuffer(msg);
+            }
+            break;
+        }
+
+        case kWhatFlush:
+        {
+            onFlush();
+            break;
+        }
+
+        case kWhatShutdown:
+        {
+            onShutdown();
             break;
         }
 
@@ -105,47 +442,16 @@
     }
 }
 
-void NuPlayer::Decoder::onFillThisBuffer(const sp<AMessage> &msg) {
-    sp<AMessage> reply;
-    CHECK(msg->findMessage("reply", &reply));
-
-#if 0
-    sp<ABuffer> outBuffer;
-    CHECK(msg->findBuffer("buffer", &outBuffer));
-#else
-    sp<ABuffer> outBuffer;
-#endif
-
-    if (mCSDIndex < mCSD.size()) {
-        outBuffer = mCSD.editItemAt(mCSDIndex++);
-        outBuffer->meta()->setInt64("timeUs", 0);
-
-        reply->setBuffer("buffer", outBuffer);
-        reply->post();
-        return;
-    }
-
-    sp<AMessage> notify = mNotify->dup();
-    notify->setMessage("codec-request", msg);
-    notify->post();
-}
-
 void NuPlayer::Decoder::signalFlush() {
-    if (mCodec != NULL) {
-        mCodec->signalFlush();
-    }
+    (new AMessage(kWhatFlush, id()))->post();
 }
 
 void NuPlayer::Decoder::signalResume() {
-    if (mCodec != NULL) {
-        mCodec->signalResume();
-    }
+    // nothing to do
 }
 
 void NuPlayer::Decoder::initiateShutdown() {
-    if (mCodec != NULL) {
-        mCodec->initiateShutdown();
-    }
+    (new AMessage(kWhatShutdown, id()))->post();
 }
 
 bool NuPlayer::Decoder::supportsSeamlessAudioFormatChange(const sp<AMessage> &targetFormat) const {
@@ -163,14 +469,16 @@
         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) {
+            if (!mOutputFormat->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 (mOutputFormat->findBuffer("csd-0", &oldBuf) &&
+                targetFormat->findBuffer("csd-0", &newBuf)) {
             if (oldBuf->size() != newBuf->size()) {
                 return false;
             }
@@ -181,7 +489,7 @@
 }
 
 bool NuPlayer::Decoder::supportsSeamlessFormatChange(const sp<AMessage> &targetFormat) const {
-    if (mFormat == NULL) {
+    if (mOutputFormat == NULL) {
         return false;
     }
 
@@ -190,7 +498,7 @@
     }
 
     AString oldMime, newMime;
-    if (!mFormat->findString("mime", &oldMime)
+    if (!mOutputFormat->findString("mime", &oldMime)
             || !targetFormat->findString("mime", &newMime)
             || !(oldMime == newMime)) {
         return false;
@@ -201,7 +509,10 @@
     if (audio) {
         seamless = supportsSeamlessAudioFormatChange(targetFormat);
     } else {
-        seamless = mCodec != NULL && mCodec->isConfiguredForAdaptivePlayback();
+        int32_t isAdaptive;
+        seamless = (mCodec != NULL &&
+                mInputFormat->findInt32("adaptive-playback", &isAdaptive) &&
+                isAdaptive);
     }
 
     ALOGV("%s seamless support for %s", seamless ? "yes" : "no", oldMime.c_str());
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
index 78ea74a..94243fc 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
@@ -25,12 +25,14 @@
 namespace android {
 
 struct ABuffer;
+struct MediaCodec;
 
 struct NuPlayer::Decoder : public AHandler {
     Decoder(const sp<AMessage> &notify,
             const sp<NativeWindowWrapper> &nativeWindow = NULL);
 
     void configure(const sp<AMessage> &format);
+    void init();
 
     void signalFlush();
     void signalResume();
@@ -38,7 +40,18 @@
 
     bool supportsSeamlessFormatChange(const sp<AMessage> &to) const;
 
+    enum {
+        kWhatFillThisBuffer      = 'flTB',
+        kWhatDrainThisBuffer     = 'drTB',
+        kWhatOutputFormatChanged = 'fmtC',
+        kWhatFlushCompleted      = 'flsC',
+        kWhatShutdownCompleted   = 'shDC',
+        kWhatEOS                 = 'eos ',
+        kWhatError               = 'err ',
+    };
+
 protected:
+
     virtual ~Decoder();
 
     virtual void onMessageReceived(const sp<AMessage> &msg);
@@ -46,21 +59,40 @@
 private:
     enum {
         kWhatCodecNotify        = 'cdcN',
+        kWhatConfigure          = 'conf',
+        kWhatInputBufferFilled  = 'inpF',
+        kWhatRenderBuffer       = 'rndr',
+        kWhatFlush              = 'flus',
+        kWhatShutdown           = 'shuD',
     };
 
     sp<AMessage> mNotify;
     sp<NativeWindowWrapper> mNativeWindow;
 
-    sp<AMessage> mFormat;
-    sp<ACodec> mCodec;
+    sp<AMessage> mInputFormat;
+    sp<AMessage> mOutputFormat;
+    sp<MediaCodec> mCodec;
     sp<ALooper> mCodecLooper;
+    sp<ALooper> mDecoderLooper;
 
-    Vector<sp<ABuffer> > mCSD;
-    size_t mCSDIndex;
+    Vector<sp<ABuffer> > mInputBuffers;
+    Vector<sp<ABuffer> > mOutputBuffers;
 
-    sp<AMessage> makeFormat(const sp<MetaData> &meta);
+    void handleError(int32_t err);
+    bool handleAnInputBuffer();
+    bool handleAnOutputBuffer();
 
-    void onFillThisBuffer(const sp<AMessage> &msg);
+    void requestCodecNotification();
+    bool isStaleReply(const sp<AMessage> &msg);
+
+    void onConfigure(const sp<AMessage> &format);
+    void onFlush();
+    void onInputBufferFilled(const sp<AMessage> &msg);
+    void onRenderBuffer(const sp<AMessage> &msg);
+    void onShutdown();
+
+    int32_t mBufferGeneration;
+    AString mComponentName;
 
     bool supportsSeamlessAudioFormatChange(const sp<AMessage> &targetFormat) const;
 
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index bf5271e..a070c1a 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -20,8 +20,6 @@
 
 #include "NuPlayerRenderer.h"
 
-#include "SoftwareRenderer.h"
-
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AMessage.h>
@@ -36,7 +34,6 @@
         const sp<AMessage> &notify,
         uint32_t flags)
     : mAudioSink(sink),
-      mSoftRenderer(NULL),
       mNotify(notify),
       mFlags(flags),
       mNumFramesWritten(0),
@@ -60,12 +57,6 @@
 }
 
 NuPlayer::Renderer::~Renderer() {
-    delete mSoftRenderer;
-}
-
-void NuPlayer::Renderer::setSoftRenderer(SoftwareRenderer *softRenderer) {
-    delete mSoftRenderer;
-    mSoftRenderer = softRenderer;
 }
 
 void NuPlayer::Renderer::queueBuffer(
@@ -425,9 +416,6 @@
         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);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h
index 9124e03..94a05ea 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h
@@ -23,7 +23,6 @@
 namespace android {
 
 struct ABuffer;
-class SoftwareRenderer;
 
 struct NuPlayer::Renderer : public AHandler {
     enum Flags {
@@ -57,8 +56,6 @@
         kWhatMediaRenderingStart = 'mdrd',
     };
 
-    void setSoftRenderer(SoftwareRenderer *softRenderer);
-
 protected:
     virtual ~Renderer();
 
@@ -86,7 +83,6 @@
     static const int64_t kMinPositionUpdateDelayUs;
 
     sp<MediaPlayerBase::AudioSink> mAudioSink;
-    SoftwareRenderer *mSoftRenderer;
     sp<AMessage> mNotify;
     uint32_t mFlags;
     List<QueueEntry> mAudioQueue;
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 9c48587..e9e96d1 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -365,7 +365,6 @@
       mIsEncoder(false),
       mUseMetadataOnEncoderOutput(false),
       mShutdownInProgress(false),
-      mIsConfiguredForAdaptivePlayback(false),
       mEncoderDelay(0),
       mEncoderPadding(0),
       mChannelMaskPresent(false),
@@ -643,18 +642,33 @@
         return err;
     }
 
-    // XXX: Is this the right logic to use?  It's not clear to me what the OMX
-    // buffer counts refer to - how do they account for the renderer holding on
-    // to buffers?
-    if (def.nBufferCountActual < def.nBufferCountMin + *minUndequeuedBuffers) {
-        OMX_U32 newBufferCount = def.nBufferCountMin + *minUndequeuedBuffers;
+    // FIXME: assume that surface is controlled by app (native window
+    // returns the number for the case when surface is not controlled by app)
+    (*minUndequeuedBuffers)++;
+
+
+    // Use conservative allocation while also trying to reduce starvation
+    //
+    // 1. allocate at least nBufferCountMin + minUndequeuedBuffers - that is the
+    //    minimum needed for the consumer to be able to work
+    // 2. try to allocate two (2) additional buffers to reduce starvation from
+    //    the consumer
+    for (OMX_U32 extraBuffers = 2; /* condition inside loop */; extraBuffers--) {
+        OMX_U32 newBufferCount =
+            def.nBufferCountMin + *minUndequeuedBuffers + extraBuffers;
         def.nBufferCountActual = newBufferCount;
         err = mOMX->setParameter(
                 mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
 
-        if (err != OK) {
-            ALOGE("[%s] setting nBufferCountActual to %lu failed: %d",
-                    mComponentName.c_str(), newBufferCount, err);
+        if (err == OK) {
+            *minUndequeuedBuffers += extraBuffers;
+            break;
+        }
+
+        ALOGW("[%s] setting nBufferCountActual to %lu failed: %d",
+                mComponentName.c_str(), newBufferCount, err);
+        /* exit condition */
+        if (extraBuffers == 0) {
             return err;
         }
     }
@@ -679,6 +693,7 @@
             &bufferCount, &bufferSize, &minUndequeuedBuffers);
     if (err != 0)
         return err;
+    mNumUndequeuedBuffers = minUndequeuedBuffers;
 
     ALOGV("[%s] Allocating %lu buffers from a native window of size %lu on "
          "output port",
@@ -744,6 +759,7 @@
             &bufferCount, &bufferSize, &minUndequeuedBuffers);
     if (err != 0)
         return err;
+    mNumUndequeuedBuffers = minUndequeuedBuffers;
 
     ALOGV("[%s] Allocating %lu meta buffers on output port",
          mComponentName.c_str(), bufferCount);
@@ -1041,6 +1057,9 @@
         encoder = false;
     }
 
+    sp<AMessage> inputFormat = new AMessage();
+    sp<AMessage> outputFormat = new AMessage();
+
     mIsEncoder = encoder;
 
     status_t err = setComponentRole(encoder /* isEncoder */, mime);
@@ -1142,7 +1161,9 @@
     int32_t haveNativeWindow = msg->findObject("native-window", &obj) &&
             obj != NULL;
     mStoreMetaDataInOutputBuffers = false;
-    mIsConfiguredForAdaptivePlayback = false;
+    if (video && !encoder) {
+        inputFormat->setInt32("adaptive-playback", false);
+    }
     if (!encoder && video && haveNativeWindow) {
         err = mOMX->storeMetaDataInBuffers(mNode, kPortIndexOutput, OMX_TRUE);
         if (err != OK) {
@@ -1187,14 +1208,19 @@
                 ALOGW_IF(err != OK,
                         "[%s] prepareForAdaptivePlayback failed w/ err %d",
                         mComponentName.c_str(), err);
-                mIsConfiguredForAdaptivePlayback = (err == OK);
+
+                if (err == OK) {
+                    inputFormat->setInt32("max-width", maxWidth);
+                    inputFormat->setInt32("max-height", maxHeight);
+                    inputFormat->setInt32("adaptive-playback", true);
+                }
             }
             // allow failure
             err = OK;
         } else {
             ALOGV("[%s] storeMetaDataInBuffers succeeded", mComponentName.c_str());
             mStoreMetaDataInOutputBuffers = true;
-            mIsConfiguredForAdaptivePlayback = true;
+            inputFormat->setInt32("adaptive-playback", true);
         }
 
         int32_t push;
@@ -1334,6 +1360,11 @@
         err = setMinBufferSize(kPortIndexInput, 8192);  // XXX
     }
 
+    CHECK_EQ(getPortFormat(kPortIndexInput, inputFormat), (status_t)OK);
+    CHECK_EQ(getPortFormat(kPortIndexOutput, outputFormat), (status_t)OK);
+    mInputFormat = inputFormat;
+    mOutputFormat = outputFormat;
+
     return err;
 }
 
@@ -2498,19 +2529,7 @@
         return;
     }
 
-    int minUndequeuedBufs = 0;
-    status_t err = mNativeWindow->query(
-            mNativeWindow.get(), NATIVE_WINDOW_MIN_UNDEQUEUED_BUFFERS,
-            &minUndequeuedBufs);
-
-    if (err != OK) {
-        ALOGE("[%s] NATIVE_WINDOW_MIN_UNDEQUEUED_BUFFERS query failed: %s (%d)",
-                mComponentName.c_str(), strerror(-err), -err);
-
-        minUndequeuedBufs = 0;
-    }
-
-    while (countBuffersOwnedByNativeWindow() > (size_t)minUndequeuedBufs
+    while (countBuffersOwnedByNativeWindow() > mNumUndequeuedBuffers
             && dequeueBufferFromNativeWindow() != NULL) {
         // these buffers will be submitted as regular buffers; account for this
         if (mStoreMetaDataInOutputBuffers && mMetaDataBuffersToSubmit > 0) {
@@ -2556,79 +2575,78 @@
     }
 }
 
-void ACodec::sendFormatChange(const sp<AMessage> &reply) {
-    sp<AMessage> notify = mNotify->dup();
-    notify->setInt32("what", kWhatOutputFormatChanged);
-
+status_t ACodec::getPortFormat(OMX_U32 portIndex, sp<AMessage> &notify) {
+    // TODO: catch errors an return them instead of using CHECK
     OMX_PARAM_PORTDEFINITIONTYPE def;
     InitOMXParams(&def);
-    def.nPortIndex = kPortIndexOutput;
+    def.nPortIndex = portIndex;
 
     CHECK_EQ(mOMX->getParameter(
                 mNode, OMX_IndexParamPortDefinition, &def, sizeof(def)),
              (status_t)OK);
 
-    CHECK_EQ((int)def.eDir, (int)OMX_DirOutput);
+    CHECK_EQ((int)def.eDir,
+            (int)(portIndex == kPortIndexOutput ? OMX_DirOutput : OMX_DirInput));
 
     switch (def.eDomain) {
         case OMX_PortDomainVideo:
         {
             OMX_VIDEO_PORTDEFINITIONTYPE *videoDef = &def.format.video;
+            switch ((int)videoDef->eCompressionFormat) {
+                case OMX_VIDEO_CodingUnused:
+                {
+                    CHECK(mIsEncoder ^ (portIndex == kPortIndexOutput));
+                    notify->setString("mime", MEDIA_MIMETYPE_VIDEO_RAW);
 
-            AString mime;
-            if (!mIsEncoder) {
-                notify->setString("mime", MEDIA_MIMETYPE_VIDEO_RAW);
-            } else if (GetMimeTypeForVideoCoding(
+                    notify->setInt32("stride", videoDef->nStride);
+                    notify->setInt32("slice-height", videoDef->nSliceHeight);
+                    notify->setInt32("color-format", videoDef->eColorFormat);
+
+                    OMX_CONFIG_RECTTYPE rect;
+                    InitOMXParams(&rect);
+                    rect.nPortIndex = kPortIndexOutput;
+
+                    if (mOMX->getConfig(
+                                mNode, OMX_IndexConfigCommonOutputCrop,
+                                &rect, sizeof(rect)) != OK) {
+                        rect.nLeft = 0;
+                        rect.nTop = 0;
+                        rect.nWidth = videoDef->nFrameWidth;
+                        rect.nHeight = videoDef->nFrameHeight;
+                    }
+
+                    CHECK_GE(rect.nLeft, 0);
+                    CHECK_GE(rect.nTop, 0);
+                    CHECK_GE(rect.nWidth, 0u);
+                    CHECK_GE(rect.nHeight, 0u);
+                    CHECK_LE(rect.nLeft + rect.nWidth - 1, videoDef->nFrameWidth);
+                    CHECK_LE(rect.nTop + rect.nHeight - 1, videoDef->nFrameHeight);
+
+                    notify->setRect(
+                            "crop",
+                            rect.nLeft,
+                            rect.nTop,
+                            rect.nLeft + rect.nWidth - 1,
+                            rect.nTop + rect.nHeight - 1);
+
+                    break;
+                }
+                default:
+                {
+                    CHECK(mIsEncoder ^ (portIndex == kPortIndexInput));
+                    AString mime;
+                    if (GetMimeTypeForVideoCoding(
                         videoDef->eCompressionFormat, &mime) != OK) {
-                notify->setString("mime", "application/octet-stream");
-            } else {
-                notify->setString("mime", mime.c_str());
+                        notify->setString("mime", "application/octet-stream");
+                    } else {
+                        notify->setString("mime", mime.c_str());
+                    }
+                    break;
+                }
             }
 
             notify->setInt32("width", videoDef->nFrameWidth);
             notify->setInt32("height", videoDef->nFrameHeight);
-
-            if (!mIsEncoder) {
-                notify->setInt32("stride", videoDef->nStride);
-                notify->setInt32("slice-height", videoDef->nSliceHeight);
-                notify->setInt32("color-format", videoDef->eColorFormat);
-
-                OMX_CONFIG_RECTTYPE rect;
-                InitOMXParams(&rect);
-                rect.nPortIndex = kPortIndexOutput;
-
-                if (mOMX->getConfig(
-                            mNode, OMX_IndexConfigCommonOutputCrop,
-                            &rect, sizeof(rect)) != OK) {
-                    rect.nLeft = 0;
-                    rect.nTop = 0;
-                    rect.nWidth = videoDef->nFrameWidth;
-                    rect.nHeight = videoDef->nFrameHeight;
-                }
-
-                CHECK_GE(rect.nLeft, 0);
-                CHECK_GE(rect.nTop, 0);
-                CHECK_GE(rect.nWidth, 0u);
-                CHECK_GE(rect.nHeight, 0u);
-                CHECK_LE(rect.nLeft + rect.nWidth - 1, videoDef->nFrameWidth);
-                CHECK_LE(rect.nTop + rect.nHeight - 1, videoDef->nFrameHeight);
-
-                notify->setRect(
-                        "crop",
-                        rect.nLeft,
-                        rect.nTop,
-                        rect.nLeft + rect.nWidth - 1,
-                        rect.nTop + rect.nHeight - 1);
-
-                if (mNativeWindow != NULL) {
-                    reply->setRect(
-                            "crop",
-                            rect.nLeft,
-                            rect.nTop,
-                            rect.nLeft + rect.nWidth,
-                            rect.nTop + rect.nHeight);
-                }
-            }
             break;
         }
 
@@ -2641,7 +2659,7 @@
                 {
                     OMX_AUDIO_PARAM_PCMMODETYPE params;
                     InitOMXParams(&params);
-                    params.nPortIndex = kPortIndexOutput;
+                    params.nPortIndex = portIndex;
 
                     CHECK_EQ(mOMX->getParameter(
                                 mNode, OMX_IndexParamAudioPcm,
@@ -2661,20 +2679,6 @@
                     notify->setString("mime", MEDIA_MIMETYPE_AUDIO_RAW);
                     notify->setInt32("channel-count", params.nChannels);
                     notify->setInt32("sample-rate", params.nSamplingRate);
-                    if (mEncoderDelay + mEncoderPadding) {
-                        size_t frameSize = params.nChannels * sizeof(int16_t);
-                        if (mSkipCutBuffer != NULL) {
-                            size_t prevbufsize = mSkipCutBuffer->size();
-                            if (prevbufsize != 0) {
-                                ALOGW("Replacing SkipCutBuffer holding %d "
-                                      "bytes",
-                                      prevbufsize);
-                            }
-                        }
-                        mSkipCutBuffer = new SkipCutBuffer(
-                                mEncoderDelay * frameSize,
-                                mEncoderPadding * frameSize);
-                    }
 
                     if (mChannelMaskPresent) {
                         notify->setInt32("channel-mask", mChannelMask);
@@ -2686,7 +2690,7 @@
                 {
                     OMX_AUDIO_PARAM_AACPROFILETYPE params;
                     InitOMXParams(&params);
-                    params.nPortIndex = kPortIndexOutput;
+                    params.nPortIndex = portIndex;
 
                     CHECK_EQ(mOMX->getParameter(
                                 mNode, OMX_IndexParamAudioAac,
@@ -2703,7 +2707,7 @@
                 {
                     OMX_AUDIO_PARAM_AMRTYPE params;
                     InitOMXParams(&params);
-                    params.nPortIndex = kPortIndexOutput;
+                    params.nPortIndex = portIndex;
 
                     CHECK_EQ(mOMX->getParameter(
                                 mNode, OMX_IndexParamAudioAmr,
@@ -2729,7 +2733,7 @@
                 {
                     OMX_AUDIO_PARAM_FLACTYPE params;
                     InitOMXParams(&params);
-                    params.nPortIndex = kPortIndexOutput;
+                    params.nPortIndex = portIndex;
 
                     CHECK_EQ(mOMX->getParameter(
                                 mNode, OMX_IndexParamAudioFlac,
@@ -2742,11 +2746,45 @@
                     break;
                 }
 
+                case OMX_AUDIO_CodingMP3:
+                {
+                    OMX_AUDIO_PARAM_MP3TYPE params;
+                    InitOMXParams(&params);
+                    params.nPortIndex = portIndex;
+
+                    CHECK_EQ(mOMX->getParameter(
+                                mNode, OMX_IndexParamAudioMp3,
+                                &params, sizeof(params)),
+                             (status_t)OK);
+
+                    notify->setString("mime", MEDIA_MIMETYPE_AUDIO_MPEG);
+                    notify->setInt32("channel-count", params.nChannels);
+                    notify->setInt32("sample-rate", params.nSampleRate);
+                    break;
+                }
+
+                case OMX_AUDIO_CodingVORBIS:
+                {
+                    OMX_AUDIO_PARAM_VORBISTYPE params;
+                    InitOMXParams(&params);
+                    params.nPortIndex = portIndex;
+
+                    CHECK_EQ(mOMX->getParameter(
+                                mNode, OMX_IndexParamAudioVorbis,
+                                &params, sizeof(params)),
+                             (status_t)OK);
+
+                    notify->setString("mime", MEDIA_MIMETYPE_AUDIO_VORBIS);
+                    notify->setInt32("channel-count", params.nChannels);
+                    notify->setInt32("sample-rate", params.nSampleRate);
+                    break;
+                }
+
                 case OMX_AUDIO_CodingAndroidAC3:
                 {
                     OMX_AUDIO_PARAM_ANDROID_AC3TYPE params;
                     InitOMXParams(&params);
-                    params.nPortIndex = kPortIndexOutput;
+                    params.nPortIndex = portIndex;
 
                     CHECK_EQ((status_t)OK, mOMX->getParameter(
                             mNode,
@@ -2761,6 +2799,7 @@
                 }
 
                 default:
+                    ALOGE("UNKNOWN AUDIO CODING: %d\n", audioDef->eEncoding);
                     TRESPASS();
             }
             break;
@@ -2770,6 +2809,43 @@
             TRESPASS();
     }
 
+    return OK;
+}
+
+void ACodec::sendFormatChange(const sp<AMessage> &reply) {
+    sp<AMessage> notify = mNotify->dup();
+    notify->setInt32("what", kWhatOutputFormatChanged);
+
+    CHECK_EQ(getPortFormat(kPortIndexOutput, notify), (status_t)OK);
+
+    AString mime;
+    CHECK(notify->findString("mime", &mime));
+
+    int32_t left, top, right, bottom;
+    if (mime == MEDIA_MIMETYPE_VIDEO_RAW &&
+        mNativeWindow != NULL &&
+        notify->findRect("crop", &left, &top, &right, &bottom)) {
+        // notify renderer of the crop change
+        // NOTE: native window uses extended right-bottom coordinate
+        reply->setRect("crop", left, top, right + 1, bottom + 1);
+    } else if (mime == MEDIA_MIMETYPE_AUDIO_RAW &&
+               (mEncoderDelay || mEncoderPadding)) {
+        int32_t channelCount;
+        CHECK(notify->findInt32("channel-count", &channelCount));
+        size_t frameSize = channelCount * sizeof(int16_t);
+        if (mSkipCutBuffer != NULL) {
+            size_t prevbufsize = mSkipCutBuffer->size();
+            if (prevbufsize != 0) {
+                ALOGW("Replacing SkipCutBuffer holding %d "
+                      "bytes",
+                      prevbufsize);
+            }
+        }
+        mSkipCutBuffer = new SkipCutBuffer(
+                mEncoderDelay * frameSize,
+                mEncoderPadding * frameSize);
+    }
+
     notify->post();
 
     mSentFormat = true;
@@ -3799,7 +3875,8 @@
     mCodec->mDequeueCounter = 0;
     mCodec->mMetaDataBuffersToSubmit = 0;
     mCodec->mRepeatFrameDelayUs = -1ll;
-    mCodec->mIsConfiguredForAdaptivePlayback = false;
+    mCodec->mInputFormat.clear();
+    mCodec->mOutputFormat.clear();
 
     if (mCodec->mShutdownInProgress) {
         bool keepComponentAllocated = mCodec->mKeepComponentAllocated;
@@ -3913,6 +3990,8 @@
     {
         sp<AMessage> notify = mCodec->mNotify->dup();
         notify->setInt32("what", ACodec::kWhatComponentConfigured);
+        notify->setMessage("input-format", mCodec->mInputFormat);
+        notify->setMessage("output-format", mCodec->mOutputFormat);
         notify->post();
     }
 
diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index 714b5e0..a9b0c73 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -118,6 +118,8 @@
 
 LOCAL_MODULE_TAGS := optional
 
+LOCAL_32_BIT_ONLY := true
+
 include $(BUILD_SHARED_LIBRARY)
 
 include $(call all-makefiles-under,$(LOCAL_PATH))
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index fe21296..e0419ca 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -352,6 +352,20 @@
     return OK;
 }
 
+status_t MediaCodec::getInputFormat(sp<AMessage> *format) const {
+    sp<AMessage> msg = new AMessage(kWhatGetInputFormat, id());
+
+    sp<AMessage> response;
+    status_t err;
+    if ((err = PostAndAwaitResponse(msg, &response)) != OK) {
+        return err;
+    }
+
+    CHECK(response->findMessage("format", format));
+
+    return OK;
+}
+
 status_t MediaCodec::getName(AString *name) const {
     sp<AMessage> msg = new AMessage(kWhatGetName, id());
 
@@ -642,6 +656,9 @@
                     // reset input surface flag
                     mHaveInputSurface = false;
 
+                    CHECK(msg->findMessage("input-format", &mInputFormat));
+                    CHECK(msg->findMessage("output-format", &mOutputFormat));
+
                     (new AMessage)->postReply(mReplyID);
                     break;
                 }
@@ -1330,14 +1347,19 @@
             break;
         }
 
+        case kWhatGetInputFormat:
         case kWhatGetOutputFormat:
         {
+            sp<AMessage> format =
+                (msg->what() == kWhatGetOutputFormat ? mOutputFormat : mInputFormat);
+
             uint32_t replyID;
             CHECK(msg->senderAwaitsResponse(&replyID));
 
-            if ((mState != STARTED && mState != FLUSHING)
+            if ((mState != CONFIGURED && mState != STARTING &&
+                 mState != STARTED && mState != FLUSHING)
                     || (mFlags & kFlagStickyError)
-                    || mOutputFormat == NULL) {
+                    || format == NULL) {
                 sp<AMessage> response = new AMessage;
                 response->setInt32("err", INVALID_OPERATION);
 
@@ -1346,7 +1368,7 @@
             }
 
             sp<AMessage> response = new AMessage;
-            response->setMessage("format", mOutputFormat);
+            response->setMessage("format", format);
             response->postReply(replyID);
             break;
         }
diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp
index 4d3b5bd..545ca9d 100644
--- a/media/libstagefright/OMXCodec.cpp
+++ b/media/libstagefright/OMXCodec.cpp
@@ -94,6 +94,7 @@
 
 #define CODEC_LOGI(x, ...) ALOGI("[%s] "x, mComponentName, ##__VA_ARGS__)
 #define CODEC_LOGV(x, ...) ALOGV("[%s] "x, mComponentName, ##__VA_ARGS__)
+#define CODEC_LOGW(x, ...) ALOGW("[%s] "x, mComponentName, ##__VA_ARGS__)
 #define CODEC_LOGE(x, ...) ALOGE("[%s] "x, mComponentName, ##__VA_ARGS__)
 
 struct OMXCodecObserver : public BnOMXObserver {
@@ -1803,21 +1804,40 @@
                 strerror(-err), -err);
         return err;
     }
+    // FIXME: assume that surface is controlled by app (native window
+    // returns the number for the case when surface is not controlled by app)
+    minUndequeuedBufs++;
 
-    // XXX: Is this the right logic to use?  It's not clear to me what the OMX
-    // buffer counts refer to - how do they account for the renderer holding on
-    // to buffers?
-    if (def.nBufferCountActual < def.nBufferCountMin + minUndequeuedBufs) {
-        OMX_U32 newBufferCount = def.nBufferCountMin + minUndequeuedBufs;
+    // Use conservative allocation while also trying to reduce starvation
+    //
+    // 1. allocate at least nBufferCountMin + minUndequeuedBuffers - that is the
+    //    minimum needed for the consumer to be able to work
+    // 2. try to allocate two (2) additional buffers to reduce starvation from
+    //    the consumer
+    CODEC_LOGI("OMX-buffers: min=%u actual=%u undeq=%d",
+            def.nBufferCountMin, def.nBufferCountActual, minUndequeuedBufs);
+
+    for (OMX_U32 extraBuffers = 2; /* condition inside loop */; extraBuffers--) {
+        OMX_U32 newBufferCount =
+            def.nBufferCountMin + minUndequeuedBufs + extraBuffers;
         def.nBufferCountActual = newBufferCount;
         err = mOMX->setParameter(
                 mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
-        if (err != OK) {
-            CODEC_LOGE("setting nBufferCountActual to %lu failed: %d",
-                    newBufferCount, err);
+
+        if (err == OK) {
+            minUndequeuedBufs += extraBuffers;
+            break;
+        }
+
+        CODEC_LOGW("setting nBufferCountActual to %lu failed: %d",
+                newBufferCount, err);
+        /* exit condition */
+        if (extraBuffers == 0) {
             return err;
         }
     }
+    CODEC_LOGI("OMX-buffers: min=%u actual=%u undeq=%d",
+            def.nBufferCountMin, def.nBufferCountActual, minUndequeuedBufs);
 
     err = native_window_set_buffer_count(
             mNativeWindow.get(), def.nBufferCountActual);
diff --git a/media/libstagefright/codecs/aacenc/Android.mk b/media/libstagefright/codecs/aacenc/Android.mk
index 58ec3ba..04dc487 100644
--- a/media/libstagefright/codecs/aacenc/Android.mk
+++ b/media/libstagefright/codecs/aacenc/Android.mk
@@ -117,6 +117,7 @@
 
   LOCAL_MODULE := libstagefright_soft_aacenc
   LOCAL_MODULE_TAGS := optional
+  LOCAL_32_BIT_ONLY := true
 
   include $(BUILD_SHARED_LIBRARY)
 
diff --git a/media/libstagefright/codecs/avc/enc/Android.mk b/media/libstagefright/codecs/avc/enc/Android.mk
index 537ba42..c2e7b81 100644
--- a/media/libstagefright/codecs/avc/enc/Android.mk
+++ b/media/libstagefright/codecs/avc/enc/Android.mk
@@ -20,6 +20,7 @@
 
 
 LOCAL_MODULE := libstagefright_avcenc
+LOCAL_32_BIT_ONLY := true
 
 LOCAL_C_INCLUDES := \
     $(LOCAL_PATH)/src \
@@ -70,6 +71,7 @@
 
 LOCAL_MODULE := libstagefright_soft_h264enc
 LOCAL_MODULE_TAGS := optional
+LOCAL_32_BIT_ONLY := true
 
 LOCAL_CFLAGS += -Werror
 
diff --git a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
index a09ab7c..5396022 100644
--- a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
+++ b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
@@ -146,6 +146,23 @@
             return OMX_ErrorNone;
         }
 
+        case OMX_IndexParamAudioMp3:
+        {
+            OMX_AUDIO_PARAM_MP3TYPE *mp3Params =
+                (OMX_AUDIO_PARAM_MP3TYPE *)params;
+
+            if (mp3Params->nPortIndex > 1) {
+                return OMX_ErrorUndefined;
+            }
+
+            mp3Params->nChannels = mNumChannels;
+            mp3Params->nBitRate = 0 /* unknown */;
+            mp3Params->nSampleRate = mSamplingRate;
+            // other fields are encoder-only
+
+            return OMX_ErrorNone;
+        }
+
         default:
             return SimpleSoftOMXComponent::internalGetParameter(index, params);
     }
diff --git a/media/libstagefright/rtsp/MyHandler.h b/media/libstagefright/rtsp/MyHandler.h
index 16549ff..f3dfc59 100644
--- a/media/libstagefright/rtsp/MyHandler.h
+++ b/media/libstagefright/rtsp/MyHandler.h
@@ -488,21 +488,32 @@
                     sp<ARTSPResponse> response =
                         static_cast<ARTSPResponse *>(obj.get());
 
-                    if (response->mStatusCode == 302) {
+                    if (response->mStatusCode == 301 || response->mStatusCode == 302) {
                         ssize_t i = response->mHeaders.indexOfKey("location");
                         CHECK_GE(i, 0);
 
-                        mSessionURL = response->mHeaders.valueAt(i);
+                        mOriginalSessionURL = response->mHeaders.valueAt(i);
+                        mSessionURL = mOriginalSessionURL;
 
-                        AString request;
-                        request = "DESCRIBE ";
-                        request.append(mSessionURL);
-                        request.append(" RTSP/1.0\r\n");
-                        request.append("Accept: application/sdp\r\n");
-                        request.append("\r\n");
+                        // Strip any authentication info from the session url, we don't
+                        // want to transmit user/pass in cleartext.
+                        AString host, path, user, pass;
+                        unsigned port;
+                        if (ARTSPConnection::ParseURL(
+                                    mSessionURL.c_str(), &host, &port, &path, &user, &pass)
+                                && user.size() > 0) {
+                            mSessionURL.clear();
+                            mSessionURL.append("rtsp://");
+                            mSessionURL.append(host);
+                            mSessionURL.append(":");
+                            mSessionURL.append(StringPrintf("%u", port));
+                            mSessionURL.append(path);
 
-                        sp<AMessage> reply = new AMessage('desc', id());
-                        mConn->sendRequest(request.c_str(), reply);
+                            ALOGI("rewritten session url: '%s'", mSessionURL.c_str());
+                        }
+
+                        sp<AMessage> reply = new AMessage('conn', id());
+                        mConn->connect(mOriginalSessionURL.c_str(), reply);
                         break;
                     }
 
diff --git a/media/mediaserver/Android.mk b/media/mediaserver/Android.mk
index f848054..d3e546a 100644
--- a/media/mediaserver/Android.mk
+++ b/media/mediaserver/Android.mk
@@ -15,6 +15,7 @@
 
 LOCAL_SHARED_LIBRARIES := \
 	libaudioflinger \
+	libaudiopolicy \
 	libcamera_metadata\
 	libcameraservice \
 	libmedialogservice \
@@ -33,8 +34,10 @@
     frameworks/av/media/libmediaplayerservice \
     frameworks/av/services/medialog \
     frameworks/av/services/audioflinger \
+    frameworks/av/services/audiopolicy \
     frameworks/av/services/camera/libcameraservice
 
 LOCAL_MODULE:= mediaserver
+LOCAL_32_BIT_ONLY := true
 
 include $(BUILD_EXECUTABLE)
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index 4524d3c..de7e3c3 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -13,6 +13,16 @@
 
 include $(CLEAR_VARS)
 
+LOCAL_SRC_FILES := \
+    ServiceUtilities.cpp
+
+# FIXME Move this library to frameworks/native
+LOCAL_MODULE := libserviceutility
+
+include $(BUILD_STATIC_LIBRARY)
+
+include $(CLEAR_VARS)
+
 LOCAL_SRC_FILES:=               \
     AudioFlinger.cpp            \
     Threads.cpp                 \
@@ -20,8 +30,6 @@
     Effects.cpp                 \
     AudioMixer.cpp.arm          \
     AudioResampler.cpp.arm      \
-    AudioPolicyService.cpp      \
-    ServiceUtilities.cpp        \
     AudioResamplerCubic.cpp.arm \
     AudioResamplerSinc.cpp.arm  \
     AudioResamplerDyn.cpp.arm
@@ -29,6 +37,7 @@
 LOCAL_SRC_FILES += StateQueue.cpp
 
 LOCAL_C_INCLUDES := \
+    $(TOPDIR)frameworks/av/services/audiopolicy \
     $(call include-path-for, audio-effects) \
     $(call include-path-for, audio-utils)
 
@@ -50,9 +59,11 @@
 LOCAL_STATIC_LIBRARIES := \
     libscheduling_policy \
     libcpustats \
-    libmedia_helper
+    libmedia_helper \
+    libserviceutility
 
 LOCAL_MODULE:= libaudioflinger
+LOCAL_32_BIT_ONLY := true
 
 LOCAL_SRC_FILES += FastMixer.cpp FastMixerState.cpp AudioWatchdog.cpp
 
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 92ee30e..50179c5 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -509,7 +509,6 @@
         audio_io_handle_t output,
         pid_t tid,
         int *sessionId,
-        String8& name,
         int clientUid,
         status_t *status)
 {
@@ -559,7 +558,6 @@
     {
         Mutex::Autolock _l(mLock);
         PlaybackThread *thread = checkPlaybackThread_l(output);
-        PlaybackThread *effectThread = NULL;
         if (thread == NULL) {
             ALOGE("no playback thread found for output handle %d", output);
             lStatus = BAD_VALUE;
@@ -567,24 +565,23 @@
         }
 
         pid_t pid = IPCThreadState::self()->getCallingPid();
-
         client = registerPid_l(pid);
 
-        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
+        PlaybackThread *effectThread = NULL;
         if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
+            lSessionId = *sessionId;
             // 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++) {
                 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
                 if (mPlaybackThreads.keyAt(i) != output) {
-                    uint32_t sessions = t->hasAudioSession(*sessionId);
+                    uint32_t sessions = t->hasAudioSession(lSessionId);
                     if (sessions & PlaybackThread::EFFECT_SESSION) {
                         effectThread = t.get();
                         break;
                     }
                 }
             }
-            lSessionId = *sessionId;
         } else {
             // if no audio session id is provided, create one here
             lSessionId = nextUniqueId();
@@ -625,18 +622,17 @@
 
     }
 
-    if (lStatus == NO_ERROR) {
-        // s for server's pid, n for normal mixer name, f for fast index
-        name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0,
-                track->fastIndex());
-        trackHandle = new TrackHandle(track);
-    } else {
-        // remove local strong reference to Client before deleting the Track so that the Client
-        // destructor is called by the TrackBase destructor with mLock held
+    if (lStatus != NO_ERROR) {
+        // remove local strong reference to Client before deleting the Track so that the
+        // Client destructor is called by the TrackBase destructor with mLock held
         client.clear();
         track.clear();
+        goto Exit;
     }
 
+    // return handle to client
+    trackHandle = new TrackHandle(track);
+
 Exit:
     *status = lStatus;
     return trackHandle;
@@ -1324,8 +1320,6 @@
     sp<RecordHandle> recordHandle;
     sp<Client> client;
     status_t lStatus;
-    RecordThread *thread;
-    size_t inFrameCount;
     int lSessionId;
 
     // check calling permissions
@@ -1342,9 +1336,9 @@
         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) {
+    // we don't yet support anything other than 16-bit PCM
+    if (!(audio_is_valid_format(format) &&
+            audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) {
         ALOGE("openRecord() invalid format %#x", format);
         lStatus = BAD_VALUE;
         goto Exit;
@@ -1357,10 +1351,9 @@
         goto Exit;
     }
 
-    // add client to list
-    { // scope for mLock
+    {
         Mutex::Autolock _l(mLock);
-        thread = checkRecordThread_l(input);
+        RecordThread *thread = checkRecordThread_l(input);
         if (thread == NULL) {
             ALOGE("openRecord() checkRecordThread_l failed");
             lStatus = BAD_VALUE;
@@ -1377,17 +1370,17 @@
         pid_t pid = IPCThreadState::self()->getCallingPid();
         client = registerPid_l(pid);
 
-        // If no audio session id is provided, create one here
         if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
             lSessionId = *sessionId;
         } else {
+            // if no audio session id is provided, create one here
             lSessionId = nextUniqueId();
             if (sessionId != NULL) {
                 *sessionId = lSessionId;
             }
         }
-        // create new record track.
-        // The record track uses one track in mHardwareMixerThread by convention.
+        ALOGV("openRecord() lSessionId: %d", lSessionId);
+
         // TODO: the uid should be passed in as a parameter to openRecord
         recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
                                                   frameCount, lSessionId,
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index c2b516b..2367d7d 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -108,7 +108,6 @@
                                 audio_io_handle_t output,
                                 pid_t tid,
                                 int *sessionId,
-                                String8& name,
                                 int clientUid,
                                 status_t *status /*non-NULL*/);
 
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 8aee194..12d453e 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1340,7 +1340,9 @@
     }
     *pFrameCount = frameCount;
 
-    if (mType == DIRECT) {
+    switch (mType) {
+
+    case 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 %#x, channelMask 0x%08x "
@@ -1350,7 +1352,9 @@
                 goto Exit;
             }
         }
-    } else if (mType == OFFLOAD) {
+        break;
+
+    case OFFLOAD:
         if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
             ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \""
                     "for output %p with format %#x",
@@ -1358,7 +1362,9 @@
             lStatus = BAD_VALUE;
             goto Exit;
         }
-    } else {
+        break;
+
+    default:
         if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) {
                 ALOGE("createTrack_l() Bad parameter: format %#x \""
                         "for output %p with format %#x",
@@ -1372,11 +1378,13 @@
             lStatus = BAD_VALUE;
             goto Exit;
         }
+        break;
+
     }
 
     lStatus = initCheck();
     if (lStatus != NO_ERROR) {
-        ALOGE("Audio driver not initialized.");
+        ALOGE("createTrack_l() audio driver not initialized");
         goto Exit;
     }
 
@@ -1416,7 +1424,6 @@
             // track must be cleared from the caller as the caller has the AF lock
             goto Exit;
         }
-
         mTracks.add(track);
 
         sp<EffectChain> chain = getEffectChain_l(sessionId);
@@ -1786,8 +1793,9 @@
     // Originally this was int16_t[] array, need to remove legacy implications.
     free(mSinkBuffer);
     mSinkBuffer = NULL;
-    const size_t sinkBufferSize = mNormalFrameCount * mChannelCount
-            * audio_bytes_per_sample(mFormat);
+    // For sink buffer size, we use the frame size from the downstream sink to avoid problems
+    // with non PCM formats for compressed music, e.g. AAC, and Offload threads.
+    const size_t sinkBufferSize = mNormalFrameCount * mFrameSize;
     (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);
 
     // We resize the mMixerBuffer according to the requirements of the sink buffer which
@@ -5052,6 +5060,7 @@
     mInput->stream->common.standby(&mInput->stream->common);
 }
 
+// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held
 sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l(
         const sp<AudioFlinger::Client>& client,
         uint32_t sampleRate,
@@ -5068,12 +5077,6 @@
     sp<RecordTrack> track;
     status_t lStatus;
 
-    lStatus = initCheck();
-    if (lStatus != NO_ERROR) {
-        ALOGE("createRecordTrack_l() audio driver not initialized");
-        goto Exit;
-    }
-
     // client expresses a preference for FAST, but we get the final say
     if (*flags & IAudioFlinger::TRACK_FAST) {
       if (
@@ -5126,7 +5129,11 @@
     }
     *pFrameCount = frameCount;
 
-    // FIXME use flags and tid similar to createTrack_l()
+    lStatus = initCheck();
+    if (lStatus != NO_ERROR) {
+        ALOGE("createRecordTrack_l() audio driver not initialized");
+        goto Exit;
+    }
 
     { // scope for mLock
         Mutex::Autolock _l(mLock);
@@ -5155,6 +5162,7 @@
             sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp);
         }
     }
+
     lStatus = NO_ERROR;
 
 Exit:
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 92ed46a..f19cd88 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -350,39 +350,39 @@
     mResumeToStopping(false),
     mFlushHwPending(false)
 {
-    if (mCblk != NULL) {
-        if (sharedBuffer == 0) {
-            mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
-                    mFrameSize);
-        } else {
-            mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount,
-                    mFrameSize);
-        }
-        mServerProxy = mAudioTrackServerProxy;
-        // to avoid leaking a track name, do not allocate one unless there is an mCblk
-        mName = thread->getTrackName_l(channelMask, sessionId);
-        if (mName < 0) {
-            ALOGE("no more track names available");
-            return;
-        }
-        // only allocate a fast track index if we were able to allocate a normal track name
-        if (flags & IAudioFlinger::TRACK_FAST) {
-            mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads();
-            ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
-            int i = __builtin_ctz(thread->mFastTrackAvailMask);
-            ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks);
-            // FIXME This is too eager.  We allocate a fast track index before the
-            //       fast track becomes active.  Since fast tracks are a scarce resource,
-            //       this means we are potentially denying other more important fast tracks from
-            //       being created.  It would be better to allocate the index dynamically.
-            mFastIndex = i;
-            // Read the initial underruns because this field is never cleared by the fast mixer
-            mObservedUnderruns = thread->getFastTrackUnderruns(i);
-            thread->mFastTrackAvailMask &= ~(1 << i);
-        }
+    if (mCblk == NULL) {
+        return;
     }
-    ALOGV("Track constructor name %d, calling pid %d", mName,
-            IPCThreadState::self()->getCallingPid());
+
+    if (sharedBuffer == 0) {
+        mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
+                mFrameSize);
+    } else {
+        mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount,
+                mFrameSize);
+    }
+    mServerProxy = mAudioTrackServerProxy;
+
+    mName = thread->getTrackName_l(channelMask, sessionId);
+    if (mName < 0) {
+        ALOGE("no more track names available");
+        return;
+    }
+    // only allocate a fast track index if we were able to allocate a normal track name
+    if (flags & IAudioFlinger::TRACK_FAST) {
+        mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads();
+        ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
+        int i = __builtin_ctz(thread->mFastTrackAvailMask);
+        ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks);
+        // FIXME This is too eager.  We allocate a fast track index before the
+        //       fast track becomes active.  Since fast tracks are a scarce resource,
+        //       this means we are potentially denying other more important fast tracks from
+        //       being created.  It would be better to allocate the index dynamically.
+        mFastIndex = i;
+        // Read the initial underruns because this field is never cleared by the fast mixer
+        mObservedUnderruns = thread->getFastTrackUnderruns(i);
+        thread->mFastTrackAvailMask &= ~(1 << i);
+    }
 }
 
 AudioFlinger::PlaybackThread::Track::~Track()
@@ -1773,7 +1773,7 @@
 
 // ----------------------------------------------------------------------------
 
-// RecordTrack constructor must be called with AudioFlinger::mLock held
+// RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
 AudioFlinger::RecordThread::RecordTrack::RecordTrack(
             RecordThread *thread,
             const sp<Client>& client,
@@ -1789,11 +1789,12 @@
         // 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);
+    if (mCblk == NULL) {
+        return;
     }
 
+    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 &&
diff --git a/services/audiopolicy/Android.mk b/services/audiopolicy/Android.mk
new file mode 100644
index 0000000..84565bb
--- /dev/null
+++ b/services/audiopolicy/Android.mk
@@ -0,0 +1,32 @@
+LOCAL_PATH:= $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:= \
+    AudioPolicyService.cpp \
+    AudioPolicyInterfaceImpl.cpp \
+    AudioPolicyClientImpl.cpp
+
+LOCAL_C_INCLUDES := \
+    $(TOPDIR)frameworks/av/services/audioflinger \
+    $(call include-path-for, audio-effects) \
+    $(call include-path-for, audio-utils)
+
+LOCAL_SHARED_LIBRARIES := \
+    libcutils \
+    libutils \
+    liblog \
+    libbinder \
+    libmedia \
+    libhardware \
+    libhardware_legacy
+
+LOCAL_STATIC_LIBRARIES := \
+    libmedia_helper \
+    libserviceutility
+
+LOCAL_MODULE:= libaudiopolicy
+
+LOCAL_CFLAGS += -fvisibility=hidden
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/services/audiopolicy/AudioPolicyClientImpl.cpp b/services/audiopolicy/AudioPolicyClientImpl.cpp
new file mode 100644
index 0000000..53f3e2d
--- /dev/null
+++ b/services/audiopolicy/AudioPolicyClientImpl.cpp
@@ -0,0 +1,261 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioPolicyService"
+//#define LOG_NDEBUG 0
+
+#include "Configuration.h"
+#undef __STRICT_ANSI__
+#define __STDINT_LIMITS
+#define __STDC_LIMIT_MACROS
+#include <stdint.h>
+
+#include <sys/time.h>
+#include <binder/IServiceManager.h>
+#include <utils/Log.h>
+#include <cutils/properties.h>
+#include <binder/IPCThreadState.h>
+#include <utils/String16.h>
+#include <utils/threads.h>
+#include "AudioPolicyService.h"
+#include "ServiceUtilities.h"
+#include <hardware_legacy/power.h>
+#include <media/AudioEffect.h>
+#include <media/EffectsFactoryApi.h>
+//#include <media/IAudioFlinger.h>
+
+#include <hardware/hardware.h>
+#include <system/audio.h>
+#include <system/audio_policy.h>
+#include <hardware/audio_policy.h>
+#include <audio_effects/audio_effects_conf.h>
+#include <media/AudioParameter.h>
+
+
+namespace android {
+
+/* implementation of the interface to the policy manager */
+extern "C" {
+
+audio_module_handle_t aps_load_hw_module(void *service __unused,
+                                             const char *name)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) {
+        ALOGW("%s: could not get AudioFlinger", __func__);
+        return 0;
+    }
+
+    return af->loadHwModule(name);
+}
+
+// deprecated: replaced by aps_open_output_on_module()
+audio_io_handle_t aps_open_output(void *service __unused,
+                                         audio_devices_t *pDevices,
+                                         uint32_t *pSamplingRate,
+                                         audio_format_t *pFormat,
+                                         audio_channel_mask_t *pChannelMask,
+                                         uint32_t *pLatencyMs,
+                                         audio_output_flags_t flags)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) {
+        ALOGW("%s: could not get AudioFlinger", __func__);
+        return 0;
+    }
+
+    return af->openOutput((audio_module_handle_t)0, pDevices, pSamplingRate, pFormat, pChannelMask,
+                          pLatencyMs, flags);
+}
+
+audio_io_handle_t aps_open_output_on_module(void *service __unused,
+                                                   audio_module_handle_t module,
+                                                   audio_devices_t *pDevices,
+                                                   uint32_t *pSamplingRate,
+                                                   audio_format_t *pFormat,
+                                                   audio_channel_mask_t *pChannelMask,
+                                                   uint32_t *pLatencyMs,
+                                                   audio_output_flags_t flags,
+                                                   const audio_offload_info_t *offloadInfo)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) {
+        ALOGW("%s: could not get AudioFlinger", __func__);
+        return 0;
+    }
+    return af->openOutput(module, pDevices, pSamplingRate, pFormat, pChannelMask,
+                          pLatencyMs, flags, offloadInfo);
+}
+
+audio_io_handle_t aps_open_dup_output(void *service __unused,
+                                                 audio_io_handle_t output1,
+                                                 audio_io_handle_t output2)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) {
+        ALOGW("%s: could not get AudioFlinger", __func__);
+        return 0;
+    }
+    return af->openDuplicateOutput(output1, output2);
+}
+
+int aps_close_output(void *service __unused, audio_io_handle_t output)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) {
+        return PERMISSION_DENIED;
+    }
+
+    return af->closeOutput(output);
+}
+
+int aps_suspend_output(void *service __unused, audio_io_handle_t output)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) {
+        ALOGW("%s: could not get AudioFlinger", __func__);
+        return PERMISSION_DENIED;
+    }
+
+    return af->suspendOutput(output);
+}
+
+int aps_restore_output(void *service __unused, audio_io_handle_t output)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) {
+        ALOGW("%s: could not get AudioFlinger", __func__);
+        return PERMISSION_DENIED;
+    }
+
+    return af->restoreOutput(output);
+}
+
+// deprecated: replaced by aps_open_input_on_module(), and acoustics parameter is ignored
+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 __unused)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) {
+        ALOGW("%s: could not get AudioFlinger", __func__);
+        return 0;
+    }
+
+    return af->openInput((audio_module_handle_t)0, pDevices, pSamplingRate, pFormat, pChannelMask);
+}
+
+audio_io_handle_t aps_open_input_on_module(void *service __unused,
+                                                  audio_module_handle_t module,
+                                                  audio_devices_t *pDevices,
+                                                  uint32_t *pSamplingRate,
+                                                  audio_format_t *pFormat,
+                                                  audio_channel_mask_t *pChannelMask)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) {
+        ALOGW("%s: could not get AudioFlinger", __func__);
+        return 0;
+    }
+
+    return af->openInput(module, pDevices, pSamplingRate, pFormat, pChannelMask);
+}
+
+int aps_close_input(void *service __unused, audio_io_handle_t input)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) {
+        return PERMISSION_DENIED;
+    }
+
+    return af->closeInput(input);
+}
+
+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->invalidateStream(stream);
+}
+
+int aps_move_effects(void *service __unused, int session,
+                                audio_io_handle_t src_output,
+                                audio_io_handle_t dst_output)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) {
+        return PERMISSION_DENIED;
+    }
+
+    return af->moveEffects(session, src_output, dst_output);
+}
+
+char * aps_get_parameters(void *service __unused, audio_io_handle_t io_handle,
+                                     const char *keys)
+{
+    String8 result = AudioSystem::getParameters(io_handle, String8(keys));
+    return strdup(result.string());
+}
+
+void aps_set_parameters(void *service, audio_io_handle_t io_handle,
+                                   const char *kv_pairs, int delay_ms)
+{
+    AudioPolicyService *audioPolicyService = (AudioPolicyService *)service;
+
+    audioPolicyService->setParameters(io_handle, kv_pairs, delay_ms);
+}
+
+int aps_set_stream_volume(void *service, audio_stream_type_t stream,
+                                     float volume, audio_io_handle_t output,
+                                     int delay_ms)
+{
+    AudioPolicyService *audioPolicyService = (AudioPolicyService *)service;
+
+    return audioPolicyService->setStreamVolume(stream, volume, output,
+                                               delay_ms);
+}
+
+int aps_start_tone(void *service, audio_policy_tone_t tone,
+                              audio_stream_type_t stream)
+{
+    AudioPolicyService *audioPolicyService = (AudioPolicyService *)service;
+
+    return audioPolicyService->startTone(tone, stream);
+}
+
+int aps_stop_tone(void *service)
+{
+    AudioPolicyService *audioPolicyService = (AudioPolicyService *)service;
+
+    return audioPolicyService->stopTone();
+}
+
+int aps_set_voice_volume(void *service, float volume, int delay_ms)
+{
+    AudioPolicyService *audioPolicyService = (AudioPolicyService *)service;
+
+    return audioPolicyService->setVoiceVolume(volume, delay_ms);
+}
+
+}; // extern "C"
+
+}; // namespace android
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
new file mode 100644
index 0000000..768b13e
--- /dev/null
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -0,0 +1,257 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIOPOLICYINTERFACE_H
+#define ANDROID_AUDIOPOLICYINTERFACE_H
+
+#include <media/AudioSystem.h>
+#include <utils/String8.h>
+
+#include <hardware/audio_policy.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+// The AudioPolicyInterface and AudioPolicyClientInterface classes define the communication interfaces
+// between the platform specific audio policy manager and Android generic audio policy manager.
+// The platform specific audio policy manager must implement methods of the AudioPolicyInterface class.
+// This implementation makes use of the AudioPolicyClientInterface to control the activity and
+// configuration of audio input and output streams.
+//
+// The platform specific audio policy manager is in charge of the audio routing and volume control
+// policies for a given platform.
+// The main roles of this module are:
+//   - keep track of current system state (removable device connections, phone state, user requests...).
+//   System state changes and user actions are notified to audio policy manager with methods of the AudioPolicyInterface.
+//   - process getOutput() queries received when AudioTrack objects are created: Those queries
+//   return a handler on an output that has been selected, configured and opened by the audio policy manager and that
+//   must be used by the AudioTrack when registering to the AudioFlinger with the createTrack() method.
+//   When the AudioTrack object is released, a putOutput() query is received and the audio policy manager can decide
+//   to close or reconfigure the output depending on other streams using this output and current system state.
+//   - similarly process getInput() and putInput() queries received from AudioRecord objects and configure audio inputs.
+//   - process volume control requests: the stream volume is converted from an index value (received from UI) to a float value
+//   applicable to each output as a function of platform specific settings and current output route (destination device). It
+//   also make sure that streams are not muted if not allowed (e.g. camera shutter sound in some countries).
+//
+// The platform specific audio policy manager is provided as a shared library by platform vendors (as for libaudio.so)
+// and is linked with libaudioflinger.so
+
+
+//    Audio Policy Manager Interface
+class AudioPolicyInterface
+{
+
+public:
+    virtual ~AudioPolicyInterface() {}
+    //
+    // configuration functions
+    //
+
+    // indicate a change in device connection status
+    virtual status_t setDeviceConnectionState(audio_devices_t device,
+                                              audio_policy_dev_state_t state,
+                                          const char *device_address) = 0;
+    // retrieve a device connection status
+    virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
+                                                                          const char *device_address) = 0;
+    // indicate a change in phone state. Valid phones states are defined by audio_mode_t
+    virtual void setPhoneState(audio_mode_t state) = 0;
+    // force using a specific device category for the specified usage
+    virtual void setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config) = 0;
+    // retrieve current device category forced for a given usage
+    virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) = 0;
+    // set a system property (e.g. camera sound always audible)
+    virtual void setSystemProperty(const char* property, const char* value) = 0;
+    // check proper initialization
+    virtual status_t initCheck() = 0;
+
+    //
+    // Audio routing query functions
+    //
+
+    // request an output appropriate for playback of the supplied stream type and parameters
+    virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
+                                        uint32_t samplingRate,
+                                        audio_format_t format,
+                                        audio_channel_mask_t channelMask,
+                                        audio_output_flags_t flags,
+                                        const audio_offload_info_t *offloadInfo) = 0;
+    // indicates to the audio policy manager that the output starts being used by corresponding stream.
+    virtual status_t startOutput(audio_io_handle_t output,
+                                 audio_stream_type_t stream,
+                                 int session = 0) = 0;
+    // indicates to the audio policy manager that the output stops being used by corresponding stream.
+    virtual status_t stopOutput(audio_io_handle_t output,
+                                audio_stream_type_t stream,
+                                int session = 0) = 0;
+    // releases the output.
+    virtual void releaseOutput(audio_io_handle_t output) = 0;
+
+    // request an input appropriate for record from the supplied device with supplied parameters.
+    virtual audio_io_handle_t getInput(audio_source_t inputSource,
+                                    uint32_t samplingRate,
+                                    audio_format_t format,
+                                    audio_channel_mask_t channelMask,
+                                    audio_in_acoustics_t acoustics) = 0;
+    // indicates to the audio policy manager that the input starts being used.
+    virtual status_t startInput(audio_io_handle_t input) = 0;
+    // indicates to the audio policy manager that the input stops being used.
+    virtual status_t stopInput(audio_io_handle_t input) = 0;
+    // releases the input.
+    virtual void releaseInput(audio_io_handle_t input) = 0;
+
+    //
+    // volume control functions
+    //
+
+    // initialises stream volume conversion parameters by specifying volume index range.
+    virtual void initStreamVolume(audio_stream_type_t stream,
+                                      int indexMin,
+                                      int indexMax) = 0;
+
+    // sets the new stream volume at a level corresponding to the supplied index for the
+    // supplied device. By convention, specifying AUDIO_DEVICE_OUT_DEFAULT means
+    // setting volume for all devices
+    virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
+                                          int index,
+                                          audio_devices_t device) = 0;
+
+    // retrieve current volume index for the specified stream and the
+    // specified device. By convention, specifying AUDIO_DEVICE_OUT_DEFAULT means
+    // querying the volume of the active device.
+    virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
+                                          int *index,
+                                          audio_devices_t device) = 0;
+
+    // return the strategy corresponding to a given stream type
+    virtual uint32_t getStrategyForStream(audio_stream_type_t stream) = 0;
+
+    // return the enabled output devices for the given stream type
+    virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream) = 0;
+
+    // Audio effect management
+    virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc) = 0;
+    virtual status_t registerEffect(const effect_descriptor_t *desc,
+                                    audio_io_handle_t io,
+                                    uint32_t strategy,
+                                    int session,
+                                    int id) = 0;
+    virtual status_t unregisterEffect(int id) = 0;
+    virtual status_t setEffectEnabled(int id, bool enabled) = 0;
+
+    virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const = 0;
+    virtual bool isStreamActiveRemotely(audio_stream_type_t stream,
+                                        uint32_t inPastMs = 0) const = 0;
+    virtual bool isSourceActive(audio_source_t source) const = 0;
+
+    //dump state
+    virtual status_t    dump(int fd) = 0;
+
+    virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo) = 0;
+};
+
+
+// Audio Policy client Interface
+class AudioPolicyClientInterface
+{
+public:
+    virtual ~AudioPolicyClientInterface() {}
+
+    //
+    // Audio HW module functions
+    //
+
+    // loads a HW module.
+    virtual audio_module_handle_t loadHwModule(const char *name) = 0;
+
+    //
+    // Audio output Control functions
+    //
+
+    // opens an audio output with the requested parameters. The parameter values can indicate to use the default values
+    // in case the audio policy manager has no specific requirements for the output being opened.
+    // When the function returns, the parameter values reflect the actual values used by the audio hardware output stream.
+    // The audio policy manager can check if the proposed parameters are suitable or not and act accordingly.
+    virtual audio_io_handle_t openOutput(audio_module_handle_t module,
+                                         audio_devices_t *pDevices,
+                                         uint32_t *pSamplingRate,
+                                         audio_format_t *pFormat,
+                                         audio_channel_mask_t *pChannelMask,
+                                         uint32_t *pLatencyMs,
+                                         audio_output_flags_t flags,
+                                         const audio_offload_info_t *offloadInfo = NULL) = 0;
+    // creates a special output that is duplicated to the two outputs passed as arguments. The duplication is performed by
+    // a special mixer thread in the AudioFlinger.
+    virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2) = 0;
+    // closes the output stream
+    virtual status_t closeOutput(audio_io_handle_t output) = 0;
+    // suspends the output. When an output is suspended, the corresponding audio hardware output stream is placed in
+    // standby and the AudioTracks attached to the mixer thread are still processed but the output mix is discarded.
+    virtual status_t suspendOutput(audio_io_handle_t output) = 0;
+    // restores a suspended output.
+    virtual status_t restoreOutput(audio_io_handle_t output) = 0;
+
+    //
+    // Audio input Control functions
+    //
+
+    // opens an audio input
+    virtual audio_io_handle_t openInput(audio_module_handle_t module,
+                                        audio_devices_t *pDevices,
+                                        uint32_t *pSamplingRate,
+                                        audio_format_t *pFormat,
+                                        audio_channel_mask_t *pChannelMask) = 0;
+    // closes an audio input
+    virtual status_t closeInput(audio_io_handle_t input) = 0;
+    //
+    // misc control functions
+    //
+
+    // set a stream volume for a particular output. For the same user setting, a given stream type can have different volumes
+    // for each output (destination device) it is attached to.
+    virtual status_t setStreamVolume(audio_stream_type_t stream, float volume, audio_io_handle_t output, int delayMs = 0) = 0;
+
+    // invalidate a stream type, causing a reroute to an unspecified new output
+    virtual status_t invalidateStream(audio_stream_type_t stream) = 0;
+
+    // function enabling to send proprietary informations directly from audio policy manager to audio hardware interface.
+    virtual void setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs, int delayMs = 0) = 0;
+    // function enabling to receive proprietary informations directly from audio hardware interface to audio policy manager.
+    virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) = 0;
+
+    // request the playback of a tone on the specified stream: used for instance to replace notification sounds when playing
+    // over a telephony device during a phone call.
+    virtual status_t startTone(audio_policy_tone_t tone, audio_stream_type_t stream) = 0;
+    virtual status_t stopTone() = 0;
+
+    // set down link audio volume.
+    virtual status_t setVoiceVolume(float volume, int delayMs = 0) = 0;
+
+    // move effect to the specified output
+    virtual status_t moveEffects(int session,
+                                     audio_io_handle_t srcOutput,
+                                     audio_io_handle_t dstOutput) = 0;
+
+};
+
+extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface);
+extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface);
+
+
+}; // namespace android
+
+#endif // ANDROID_AUDIOPOLICYINTERFACE_H
diff --git a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/AudioPolicyInterfaceImpl.cpp
new file mode 100644
index 0000000..bb62ab3
--- /dev/null
+++ b/services/audiopolicy/AudioPolicyInterfaceImpl.cpp
@@ -0,0 +1,489 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioPolicyService"
+//#define LOG_NDEBUG 0
+
+#include <utils/Log.h>
+#include "AudioPolicyService.h"
+#include "ServiceUtilities.h"
+
+#include <system/audio.h>
+#include <system/audio_policy.h>
+#include <hardware/audio_policy.h>
+
+namespace android {
+
+
+// ----------------------------------------------------------------------------
+
+status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device,
+                                                  audio_policy_dev_state_t state,
+                                                  const char *device_address)
+{
+    if (mpAudioPolicy == NULL) {
+        return NO_INIT;
+    }
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+    if (!audio_is_output_device(device) && !audio_is_input_device(device)) {
+        return BAD_VALUE;
+    }
+    if (state != AUDIO_POLICY_DEVICE_STATE_AVAILABLE &&
+            state != AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
+        return BAD_VALUE;
+    }
+
+    ALOGV("setDeviceConnectionState()");
+    Mutex::Autolock _l(mLock);
+    return mpAudioPolicy->set_device_connection_state(mpAudioPolicy, device,
+                                                      state, device_address);
+}
+
+audio_policy_dev_state_t AudioPolicyService::getDeviceConnectionState(
+                                                              audio_devices_t device,
+                                                              const char *device_address)
+{
+    if (mpAudioPolicy == NULL) {
+        return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
+    }
+    return mpAudioPolicy->get_device_connection_state(mpAudioPolicy, device,
+                                                      device_address);
+}
+
+status_t AudioPolicyService::setPhoneState(audio_mode_t state)
+{
+    if (mpAudioPolicy == NULL) {
+        return NO_INIT;
+    }
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+    if (uint32_t(state) >= AUDIO_MODE_CNT) {
+        return BAD_VALUE;
+    }
+
+    ALOGV("setPhoneState()");
+
+    // TODO: check if it is more appropriate to do it in platform specific policy manager
+    AudioSystem::setMode(state);
+
+    Mutex::Autolock _l(mLock);
+    mpAudioPolicy->set_phone_state(mpAudioPolicy, state);
+    return NO_ERROR;
+}
+
+status_t AudioPolicyService::setForceUse(audio_policy_force_use_t usage,
+                                         audio_policy_forced_cfg_t config)
+{
+    if (mpAudioPolicy == NULL) {
+        return NO_INIT;
+    }
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+    if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) {
+        return BAD_VALUE;
+    }
+    if (config < 0 || config >= AUDIO_POLICY_FORCE_CFG_CNT) {
+        return BAD_VALUE;
+    }
+    ALOGV("setForceUse()");
+    Mutex::Autolock _l(mLock);
+    mpAudioPolicy->set_force_use(mpAudioPolicy, usage, config);
+    return NO_ERROR;
+}
+
+audio_policy_forced_cfg_t AudioPolicyService::getForceUse(audio_policy_force_use_t usage)
+{
+    if (mpAudioPolicy == NULL) {
+        return AUDIO_POLICY_FORCE_NONE;
+    }
+    if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) {
+        return AUDIO_POLICY_FORCE_NONE;
+    }
+    return mpAudioPolicy->get_force_use(mpAudioPolicy, usage);
+}
+
+audio_io_handle_t AudioPolicyService::getOutput(audio_stream_type_t stream,
+                                    uint32_t samplingRate,
+                                    audio_format_t format,
+                                    audio_channel_mask_t channelMask,
+                                    audio_output_flags_t flags,
+                                    const audio_offload_info_t *offloadInfo)
+{
+    if (mpAudioPolicy == NULL) {
+        return 0;
+    }
+    ALOGV("getOutput()");
+    Mutex::Autolock _l(mLock);
+    return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate,
+                                    format, channelMask, flags, offloadInfo);
+}
+
+status_t AudioPolicyService::startOutput(audio_io_handle_t output,
+                                         audio_stream_type_t stream,
+                                         int session)
+{
+    if (mpAudioPolicy == NULL) {
+        return NO_INIT;
+    }
+    ALOGV("startOutput()");
+    Mutex::Autolock _l(mLock);
+    return mpAudioPolicy->start_output(mpAudioPolicy, output, stream, session);
+}
+
+status_t AudioPolicyService::stopOutput(audio_io_handle_t output,
+                                        audio_stream_type_t stream,
+                                        int session)
+{
+    if (mpAudioPolicy == NULL) {
+        return NO_INIT;
+    }
+    ALOGV("stopOutput()");
+    mOutputCommandThread->stopOutputCommand(output, stream, session);
+    return NO_ERROR;
+}
+
+status_t  AudioPolicyService::doStopOutput(audio_io_handle_t output,
+                                      audio_stream_type_t stream,
+                                      int session)
+{
+    ALOGV("doStopOutput from tid %d", gettid());
+    Mutex::Autolock _l(mLock);
+    return mpAudioPolicy->stop_output(mpAudioPolicy, output, stream, session);
+}
+
+void AudioPolicyService::releaseOutput(audio_io_handle_t output)
+{
+    if (mpAudioPolicy == NULL) {
+        return;
+    }
+    ALOGV("releaseOutput()");
+    mOutputCommandThread->releaseOutputCommand(output);
+}
+
+void AudioPolicyService::doReleaseOutput(audio_io_handle_t output)
+{
+    ALOGV("doReleaseOutput from tid %d", gettid());
+    Mutex::Autolock _l(mLock);
+    mpAudioPolicy->release_output(mpAudioPolicy, output);
+}
+
+audio_io_handle_t AudioPolicyService::getInput(audio_source_t inputSource,
+                                    uint32_t samplingRate,
+                                    audio_format_t format,
+                                    audio_channel_mask_t channelMask,
+                                    int audioSession)
+{
+    if (mpAudioPolicy == NULL) {
+        return 0;
+    }
+    // already checked by client, but double-check in case the client wrapper is bypassed
+    if (inputSource >= AUDIO_SOURCE_CNT && inputSource != AUDIO_SOURCE_HOTWORD) {
+        return 0;
+    }
+
+    if ((inputSource == AUDIO_SOURCE_HOTWORD) && !captureHotwordAllowed()) {
+        return 0;
+    }
+
+    Mutex::Autolock _l(mLock);
+    // the audio_in_acoustics_t parameter is ignored by get_input()
+    audio_io_handle_t input = mpAudioPolicy->get_input(mpAudioPolicy, inputSource, samplingRate,
+                                                   format, channelMask, (audio_in_acoustics_t) 0);
+
+    if (input == 0) {
+        return input;
+    }
+    // create audio pre processors according to input source
+    audio_source_t aliasSource = (inputSource == AUDIO_SOURCE_HOTWORD) ?
+                                    AUDIO_SOURCE_VOICE_RECOGNITION : inputSource;
+
+    ssize_t index = mInputSources.indexOfKey(aliasSource);
+    if (index < 0) {
+        return input;
+    }
+    ssize_t idx = mInputs.indexOfKey(input);
+    InputDesc *inputDesc;
+    if (idx < 0) {
+        inputDesc = new InputDesc(audioSession);
+        mInputs.add(input, inputDesc);
+    } else {
+        inputDesc = mInputs.valueAt(idx);
+    }
+
+    Vector <EffectDesc *> effects = mInputSources.valueAt(index)->mEffects;
+    for (size_t i = 0; i < effects.size(); i++) {
+        EffectDesc *effect = effects[i];
+        sp<AudioEffect> fx = new AudioEffect(NULL, &effect->mUuid, -1, 0, 0, audioSession, input);
+        status_t status = fx->initCheck();
+        if (status != NO_ERROR && status != ALREADY_EXISTS) {
+            ALOGW("Failed to create Fx %s on input %d", effect->mName, input);
+            // fx goes out of scope and strong ref on AudioEffect is released
+            continue;
+        }
+        for (size_t j = 0; j < effect->mParams.size(); j++) {
+            fx->setParameter(effect->mParams[j]);
+        }
+        inputDesc->mEffects.add(fx);
+    }
+    setPreProcessorEnabled(inputDesc, true);
+    return input;
+}
+
+status_t AudioPolicyService::startInput(audio_io_handle_t input)
+{
+    if (mpAudioPolicy == NULL) {
+        return NO_INIT;
+    }
+    Mutex::Autolock _l(mLock);
+
+    return mpAudioPolicy->start_input(mpAudioPolicy, input);
+}
+
+status_t AudioPolicyService::stopInput(audio_io_handle_t input)
+{
+    if (mpAudioPolicy == NULL) {
+        return NO_INIT;
+    }
+    Mutex::Autolock _l(mLock);
+
+    return mpAudioPolicy->stop_input(mpAudioPolicy, input);
+}
+
+void AudioPolicyService::releaseInput(audio_io_handle_t input)
+{
+    if (mpAudioPolicy == NULL) {
+        return;
+    }
+    Mutex::Autolock _l(mLock);
+    mpAudioPolicy->release_input(mpAudioPolicy, input);
+
+    ssize_t index = mInputs.indexOfKey(input);
+    if (index < 0) {
+        return;
+    }
+    InputDesc *inputDesc = mInputs.valueAt(index);
+    setPreProcessorEnabled(inputDesc, false);
+    delete inputDesc;
+    mInputs.removeItemsAt(index);
+}
+
+status_t AudioPolicyService::initStreamVolume(audio_stream_type_t stream,
+                                            int indexMin,
+                                            int indexMax)
+{
+    if (mpAudioPolicy == NULL) {
+        return NO_INIT;
+    }
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
+        return BAD_VALUE;
+    }
+    Mutex::Autolock _l(mLock);
+    mpAudioPolicy->init_stream_volume(mpAudioPolicy, stream, indexMin, indexMax);
+    return NO_ERROR;
+}
+
+status_t AudioPolicyService::setStreamVolumeIndex(audio_stream_type_t stream,
+                                                  int index,
+                                                  audio_devices_t device)
+{
+    if (mpAudioPolicy == NULL) {
+        return NO_INIT;
+    }
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
+        return BAD_VALUE;
+    }
+    Mutex::Autolock _l(mLock);
+    if (mpAudioPolicy->set_stream_volume_index_for_device) {
+        return mpAudioPolicy->set_stream_volume_index_for_device(mpAudioPolicy,
+                                                                stream,
+                                                                index,
+                                                                device);
+    } else {
+        return mpAudioPolicy->set_stream_volume_index(mpAudioPolicy, stream, index);
+    }
+}
+
+status_t AudioPolicyService::getStreamVolumeIndex(audio_stream_type_t stream,
+                                                  int *index,
+                                                  audio_devices_t device)
+{
+    if (mpAudioPolicy == NULL) {
+        return NO_INIT;
+    }
+    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
+        return BAD_VALUE;
+    }
+    Mutex::Autolock _l(mLock);
+    if (mpAudioPolicy->get_stream_volume_index_for_device) {
+        return mpAudioPolicy->get_stream_volume_index_for_device(mpAudioPolicy,
+                                                                stream,
+                                                                index,
+                                                                device);
+    } else {
+        return mpAudioPolicy->get_stream_volume_index(mpAudioPolicy, stream, index);
+    }
+}
+
+uint32_t AudioPolicyService::getStrategyForStream(audio_stream_type_t stream)
+{
+    if (mpAudioPolicy == NULL) {
+        return 0;
+    }
+    return mpAudioPolicy->get_strategy_for_stream(mpAudioPolicy, stream);
+}
+
+//audio policy: use audio_device_t appropriately
+
+audio_devices_t AudioPolicyService::getDevicesForStream(audio_stream_type_t stream)
+{
+    if (mpAudioPolicy == NULL) {
+        return (audio_devices_t)0;
+    }
+    return mpAudioPolicy->get_devices_for_stream(mpAudioPolicy, stream);
+}
+
+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 0;
+    }
+    Mutex::Autolock _l(mLock);
+    return mpAudioPolicy->get_output_for_effect(mpAudioPolicy, desc);
+}
+
+status_t AudioPolicyService::registerEffect(const effect_descriptor_t *desc,
+                                audio_io_handle_t io,
+                                uint32_t strategy,
+                                int session,
+                                int id)
+{
+    if (mpAudioPolicy == NULL) {
+        return NO_INIT;
+    }
+    return mpAudioPolicy->register_effect(mpAudioPolicy, desc, io, strategy, session, id);
+}
+
+status_t AudioPolicyService::unregisterEffect(int id)
+{
+    if (mpAudioPolicy == NULL) {
+        return NO_INIT;
+    }
+    return mpAudioPolicy->unregister_effect(mpAudioPolicy, id);
+}
+
+status_t AudioPolicyService::setEffectEnabled(int id, bool enabled)
+{
+    if (mpAudioPolicy == NULL) {
+        return NO_INIT;
+    }
+    return mpAudioPolicy->set_effect_enabled(mpAudioPolicy, id, enabled);
+}
+
+bool AudioPolicyService::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
+{
+    if (mpAudioPolicy == NULL) {
+        return 0;
+    }
+    Mutex::Autolock _l(mLock);
+    return mpAudioPolicy->is_stream_active(mpAudioPolicy, stream, inPastMs);
+}
+
+bool AudioPolicyService::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const
+{
+    if (mpAudioPolicy == NULL) {
+        return 0;
+    }
+    Mutex::Autolock _l(mLock);
+    return mpAudioPolicy->is_stream_active_remotely(mpAudioPolicy, stream, inPastMs);
+}
+
+bool AudioPolicyService::isSourceActive(audio_source_t source) const
+{
+    if (mpAudioPolicy == NULL) {
+        return false;
+    }
+    if (mpAudioPolicy->is_source_active == 0) {
+        return false;
+    }
+    Mutex::Autolock _l(mLock);
+    return mpAudioPolicy->is_source_active(mpAudioPolicy, source);
+}
+
+status_t AudioPolicyService::queryDefaultPreProcessing(int audioSession,
+                                                       effect_descriptor_t *descriptors,
+                                                       uint32_t *count)
+{
+
+    if (mpAudioPolicy == NULL) {
+        *count = 0;
+        return NO_INIT;
+    }
+    Mutex::Autolock _l(mLock);
+    status_t status = NO_ERROR;
+
+    size_t index;
+    for (index = 0; index < mInputs.size(); index++) {
+        if (mInputs.valueAt(index)->mSessionId == audioSession) {
+            break;
+        }
+    }
+    if (index == mInputs.size()) {
+        *count = 0;
+        return BAD_VALUE;
+    }
+    Vector< sp<AudioEffect> > effects = mInputs.valueAt(index)->mEffects;
+
+    for (size_t i = 0; i < effects.size(); i++) {
+        effect_descriptor_t desc = effects[i]->descriptor();
+        if (i < *count) {
+            descriptors[i] = desc;
+        }
+    }
+    if (effects.size() > *count) {
+        status = NO_MEMORY;
+    }
+    *count = effects.size();
+    return status;
+}
+
+bool AudioPolicyService::isOffloadSupported(const audio_offload_info_t& info)
+{
+    if (mpAudioPolicy == NULL) {
+        ALOGV("mpAudioPolicy == NULL");
+        return false;
+    }
+
+    if (mpAudioPolicy->is_offload_supported == NULL) {
+        ALOGV("HAL does not implement is_offload_supported");
+        return false;
+    }
+
+    return mpAudioPolicy->is_offload_supported(mpAudioPolicy, &info);
+}
+
+
+}; // namespace android
diff --git a/services/audiopolicy/AudioPolicyManager.cpp b/services/audiopolicy/AudioPolicyManager.cpp
new file mode 100644
index 0000000..5ac9d9e
--- /dev/null
+++ b/services/audiopolicy/AudioPolicyManager.cpp
@@ -0,0 +1,4104 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioPolicyManager"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+// A device mask for all audio input devices that are considered "virtual" when evaluating
+// active inputs in getActiveInput()
+#define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL  AUDIO_DEVICE_IN_REMOTE_SUBMIX
+// A device mask for all audio output devices that are considered "remote" when evaluating
+// active output devices in isStreamActiveRemotely()
+#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL  AUDIO_DEVICE_OUT_REMOTE_SUBMIX
+
+#include <utils/Log.h>
+#include "AudioPolicyManager.h"
+#include <hardware/audio_effect.h>
+#include <hardware/audio.h>
+#include <math.h>
+#include <hardware_legacy/audio_policy_conf.h>
+#include <cutils/properties.h>
+#include <media/AudioParameter.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+// AudioPolicyInterface implementation
+// ----------------------------------------------------------------------------
+
+
+status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device,
+                                                          audio_policy_dev_state_t state,
+                                                  const char *device_address)
+{
+    SortedVector <audio_io_handle_t> outputs;
+
+    ALOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address);
+
+    // connect/disconnect only 1 device at a time
+    if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
+
+    if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) {
+        ALOGE("setDeviceConnectionState() invalid address: %s", device_address);
+        return BAD_VALUE;
+    }
+
+    // handle output devices
+    if (audio_is_output_device(device)) {
+
+        if (!mHasA2dp && audio_is_a2dp_device(device)) {
+            ALOGE("setDeviceConnectionState() invalid A2DP device: %x", device);
+            return BAD_VALUE;
+        }
+        if (!mHasUsb && audio_is_usb_device(device)) {
+            ALOGE("setDeviceConnectionState() invalid USB audio device: %x", device);
+            return BAD_VALUE;
+        }
+        if (!mHasRemoteSubmix && audio_is_remote_submix_device((audio_devices_t)device)) {
+            ALOGE("setDeviceConnectionState() invalid remote submix audio device: %x", device);
+            return BAD_VALUE;
+        }
+
+        // save a copy of the opened output descriptors before any output is opened or closed
+        // by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
+        mPreviousOutputs = mOutputs;
+        String8 paramStr;
+        switch (state)
+        {
+        // handle output device connection
+        case AUDIO_POLICY_DEVICE_STATE_AVAILABLE:
+            if (mAvailableOutputDevices & device) {
+                ALOGW("setDeviceConnectionState() device already connected: %x", device);
+                return INVALID_OPERATION;
+            }
+            ALOGV("setDeviceConnectionState() connecting device %x", device);
+
+            if (mHasA2dp && audio_is_a2dp_device(device)) {
+                // handle A2DP device connection
+                AudioParameter param;
+                param.add(String8(AUDIO_PARAMETER_A2DP_SINK_ADDRESS), String8(device_address));
+                paramStr = param.toString();
+            } else if (mHasUsb && audio_is_usb_device(device)) {
+                // handle USB device connection
+                paramStr = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+            }
+
+            if (checkOutputsForDevice(device, state, outputs, paramStr) != NO_ERROR) {
+                return INVALID_OPERATION;
+            }
+            ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %d outputs",
+                  outputs.size());
+            // register new device as available
+            mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices | device);
+
+            if (mHasA2dp && audio_is_a2dp_device(device)) {
+                // handle A2DP device connection
+                mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+                mA2dpSuspended = false;
+            } else if (audio_is_bluetooth_sco_device(device)) {
+                // handle SCO device connection
+                mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+            } else if (mHasUsb && audio_is_usb_device(device)) {
+                // handle USB device connection
+                mUsbCardAndDevice = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+            }
+
+            break;
+        // handle output device disconnection
+        case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
+            if (!(mAvailableOutputDevices & device)) {
+                ALOGW("setDeviceConnectionState() device not connected: %x", device);
+                return INVALID_OPERATION;
+            }
+
+            ALOGV("setDeviceConnectionState() disconnecting device %x", device);
+            // remove device from available output devices
+            mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices & ~device);
+
+            checkOutputsForDevice(device, state, outputs, paramStr);
+            if (mHasA2dp && audio_is_a2dp_device(device)) {
+                // handle A2DP device disconnection
+                mA2dpDeviceAddress = "";
+                mA2dpSuspended = false;
+            } else if (audio_is_bluetooth_sco_device(device)) {
+                // handle SCO device disconnection
+                mScoDeviceAddress = "";
+            } else if (mHasUsb && audio_is_usb_device(device)) {
+                // handle USB device disconnection
+                mUsbCardAndDevice = "";
+            }
+            // not currently handling multiple simultaneous submixes: ignoring remote submix
+            //   case and address
+            } break;
+
+        default:
+            ALOGE("setDeviceConnectionState() invalid state: %x", state);
+            return BAD_VALUE;
+        }
+
+        checkA2dpSuspend();
+        checkOutputForAllStrategies();
+        // outputs must be closed after checkOutputForAllStrategies() is executed
+        if (!outputs.isEmpty()) {
+            for (size_t i = 0; i < outputs.size(); i++) {
+                AudioOutputDescriptor *desc = mOutputs.valueFor(outputs[i]);
+                // close unused outputs after device disconnection or direct outputs that have been
+                // opened by checkOutputsForDevice() to query dynamic parameters
+                if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
+                        (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
+                         (desc->mDirectOpenCount == 0))) {
+                    closeOutput(outputs[i]);
+                }
+            }
+        }
+
+        updateDevicesAndOutputs();
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            // do not force device change on duplicated output because if device is 0, it will
+            // also force a device 0 for the two outputs it is duplicated to which may override
+            // a valid device selection on those outputs.
+            setOutputDevice(mOutputs.keyAt(i),
+                            getNewDevice(mOutputs.keyAt(i), true /*fromCache*/),
+                            !mOutputs.valueAt(i)->isDuplicated(),
+                            0);
+        }
+
+        if (device == AUDIO_DEVICE_OUT_WIRED_HEADSET) {
+            device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+        } else if (device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO ||
+                   device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET ||
+                   device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT) {
+            device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+        } else {
+            return NO_ERROR;
+        }
+    }
+    // handle input devices
+    if (audio_is_input_device(device)) {
+
+        switch (state)
+        {
+        // handle input device connection
+        case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
+            if (mAvailableInputDevices & device) {
+                ALOGW("setDeviceConnectionState() device already connected: %d", device);
+                return INVALID_OPERATION;
+            }
+            mAvailableInputDevices = mAvailableInputDevices | (device & ~AUDIO_DEVICE_BIT_IN);
+            }
+            break;
+
+        // handle input device disconnection
+        case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
+            if (!(mAvailableInputDevices & device)) {
+                ALOGW("setDeviceConnectionState() device not connected: %d", device);
+                return INVALID_OPERATION;
+            }
+            mAvailableInputDevices = (audio_devices_t) (mAvailableInputDevices & ~device);
+            } break;
+
+        default:
+            ALOGE("setDeviceConnectionState() invalid state: %x", state);
+            return BAD_VALUE;
+        }
+
+        audio_io_handle_t activeInput = getActiveInput();
+        if (activeInput != 0) {
+            AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
+            audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
+            if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
+                ALOGV("setDeviceConnectionState() changing device from %x to %x for input %d",
+                        inputDesc->mDevice, newDevice, activeInput);
+                inputDesc->mDevice = newDevice;
+                AudioParameter param = AudioParameter();
+                param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
+                mpClientInterface->setParameters(activeInput, param.toString());
+            }
+        }
+
+        return NO_ERROR;
+    }
+
+    ALOGW("setDeviceConnectionState() invalid device: %x", device);
+    return BAD_VALUE;
+}
+
+audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device,
+                                                  const char *device_address)
+{
+    audio_policy_dev_state_t state = AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
+    String8 address = String8(device_address);
+    if (audio_is_output_device(device)) {
+        if (device & mAvailableOutputDevices) {
+            if (audio_is_a2dp_device(device) &&
+                (!mHasA2dp || (address != "" && mA2dpDeviceAddress != address))) {
+                return state;
+            }
+            if (audio_is_bluetooth_sco_device(device) &&
+                address != "" && mScoDeviceAddress != address) {
+                return state;
+            }
+            if (audio_is_usb_device(device) &&
+                (!mHasUsb || (address != "" && mUsbCardAndDevice != address))) {
+                ALOGE("getDeviceConnectionState() invalid device: %x", device);
+                return state;
+            }
+            if (audio_is_remote_submix_device((audio_devices_t)device) && !mHasRemoteSubmix) {
+                return state;
+            }
+            state = AUDIO_POLICY_DEVICE_STATE_AVAILABLE;
+        }
+    } else if (audio_is_input_device(device)) {
+        if (device & mAvailableInputDevices) {
+            state = AUDIO_POLICY_DEVICE_STATE_AVAILABLE;
+        }
+    }
+
+    return state;
+}
+
+void AudioPolicyManager::setPhoneState(audio_mode_t state)
+{
+    ALOGV("setPhoneState() state %d", state);
+    audio_devices_t newDevice = AUDIO_DEVICE_NONE;
+    if (state < 0 || state >= AUDIO_MODE_CNT) {
+        ALOGW("setPhoneState() invalid state %d", state);
+        return;
+    }
+
+    if (state == mPhoneState ) {
+        ALOGW("setPhoneState() setting same state %d", state);
+        return;
+    }
+
+    // if leaving call state, handle special case of active streams
+    // pertaining to sonification strategy see handleIncallSonification()
+    if (isInCall()) {
+        ALOGV("setPhoneState() in call state management: new state is %d", state);
+        for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+            handleIncallSonification((audio_stream_type_t)stream, false, true);
+        }
+    }
+
+    // store previous phone state for management of sonification strategy below
+    int oldState = mPhoneState;
+    mPhoneState = state;
+    bool force = false;
+
+    // are we entering or starting a call
+    if (!isStateInCall(oldState) && isStateInCall(state)) {
+        ALOGV("  Entering call in setPhoneState()");
+        // force routing command to audio hardware when starting a call
+        // even if no device change is needed
+        force = true;
+        for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+            mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
+                    sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j];
+        }
+    } else if (isStateInCall(oldState) && !isStateInCall(state)) {
+        ALOGV("  Exiting call in setPhoneState()");
+        // force routing command to audio hardware when exiting a call
+        // even if no device change is needed
+        force = true;
+        for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+            mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
+                    sVolumeProfiles[AUDIO_STREAM_DTMF][j];
+        }
+    } else if (isStateInCall(state) && (state != oldState)) {
+        ALOGV("  Switching between telephony and VoIP in setPhoneState()");
+        // force routing command to audio hardware when switching between telephony and VoIP
+        // even if no device change is needed
+        force = true;
+    }
+
+    // check for device and output changes triggered by new phone state
+    newDevice = getNewDevice(mPrimaryOutput, false /*fromCache*/);
+    checkA2dpSuspend();
+    checkOutputForAllStrategies();
+    updateDevicesAndOutputs();
+
+    AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
+
+    // force routing command to audio hardware when ending call
+    // even if no device change is needed
+    if (isStateInCall(oldState) && newDevice == AUDIO_DEVICE_NONE) {
+        newDevice = hwOutputDesc->device();
+    }
+
+    int delayMs = 0;
+    if (isStateInCall(state)) {
+        nsecs_t sysTime = systemTime();
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+            // mute media and sonification strategies and delay device switch by the largest
+            // latency of any output where either strategy is active.
+            // This avoid sending the ring tone or music tail into the earpiece or headset.
+            if ((desc->isStrategyActive(STRATEGY_MEDIA,
+                                     SONIFICATION_HEADSET_MUSIC_DELAY,
+                                     sysTime) ||
+                    desc->isStrategyActive(STRATEGY_SONIFICATION,
+                                         SONIFICATION_HEADSET_MUSIC_DELAY,
+                                         sysTime)) &&
+                    (delayMs < (int)desc->mLatency*2)) {
+                delayMs = desc->mLatency*2;
+            }
+            setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i));
+            setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+                getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
+            setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i));
+            setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+                getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
+        }
+    }
+
+    // change routing is necessary
+    setOutputDevice(mPrimaryOutput, newDevice, force, delayMs);
+
+    // if entering in call state, handle special case of active streams
+    // pertaining to sonification strategy see handleIncallSonification()
+    if (isStateInCall(state)) {
+        ALOGV("setPhoneState() in call state management: new state is %d", state);
+        for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+            handleIncallSonification((audio_stream_type_t)stream, true, true);
+        }
+    }
+
+    // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
+    if (state == AUDIO_MODE_RINGTONE &&
+        isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
+        mLimitRingtoneVolume = true;
+    } else {
+        mLimitRingtoneVolume = false;
+    }
+}
+
+void AudioPolicyManager::setForceUse(audio_policy_force_use_t usage,
+                                         audio_policy_forced_cfg_t config)
+{
+    ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
+
+    bool forceVolumeReeval = false;
+    switch(usage) {
+    case AUDIO_POLICY_FORCE_FOR_COMMUNICATION:
+        if (config != AUDIO_POLICY_FORCE_SPEAKER && config != AUDIO_POLICY_FORCE_BT_SCO &&
+            config != AUDIO_POLICY_FORCE_NONE) {
+            ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
+            return;
+        }
+        forceVolumeReeval = true;
+        mForceUse[usage] = config;
+        break;
+    case AUDIO_POLICY_FORCE_FOR_MEDIA:
+        if (config != AUDIO_POLICY_FORCE_HEADPHONES && config != AUDIO_POLICY_FORCE_BT_A2DP &&
+            config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
+            config != AUDIO_POLICY_FORCE_ANALOG_DOCK &&
+            config != AUDIO_POLICY_FORCE_DIGITAL_DOCK && config != AUDIO_POLICY_FORCE_NONE &&
+            config != AUDIO_POLICY_FORCE_NO_BT_A2DP) {
+            ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config);
+            return;
+        }
+        mForceUse[usage] = config;
+        break;
+    case AUDIO_POLICY_FORCE_FOR_RECORD:
+        if (config != AUDIO_POLICY_FORCE_BT_SCO && config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
+            config != AUDIO_POLICY_FORCE_NONE) {
+            ALOGW("setForceUse() invalid config %d for FOR_RECORD", config);
+            return;
+        }
+        mForceUse[usage] = config;
+        break;
+    case AUDIO_POLICY_FORCE_FOR_DOCK:
+        if (config != AUDIO_POLICY_FORCE_NONE && config != AUDIO_POLICY_FORCE_BT_CAR_DOCK &&
+            config != AUDIO_POLICY_FORCE_BT_DESK_DOCK &&
+            config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
+            config != AUDIO_POLICY_FORCE_ANALOG_DOCK &&
+            config != AUDIO_POLICY_FORCE_DIGITAL_DOCK) {
+            ALOGW("setForceUse() invalid config %d for FOR_DOCK", config);
+        }
+        forceVolumeReeval = true;
+        mForceUse[usage] = config;
+        break;
+    case AUDIO_POLICY_FORCE_FOR_SYSTEM:
+        if (config != AUDIO_POLICY_FORCE_NONE &&
+            config != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
+            ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config);
+        }
+        forceVolumeReeval = true;
+        mForceUse[usage] = config;
+        break;
+    default:
+        ALOGW("setForceUse() invalid usage %d", usage);
+        break;
+    }
+
+    // check for device and output changes triggered by new force usage
+    checkA2dpSuspend();
+    checkOutputForAllStrategies();
+    updateDevicesAndOutputs();
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        audio_io_handle_t output = mOutputs.keyAt(i);
+        audio_devices_t newDevice = getNewDevice(output, true /*fromCache*/);
+        setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE));
+        if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
+            applyStreamVolumes(output, newDevice, 0, true);
+        }
+    }
+
+    audio_io_handle_t activeInput = getActiveInput();
+    if (activeInput != 0) {
+        AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
+        audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
+        if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
+            ALOGV("setForceUse() changing device from %x to %x for input %d",
+                    inputDesc->mDevice, newDevice, activeInput);
+            inputDesc->mDevice = newDevice;
+            AudioParameter param = AudioParameter();
+            param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
+            mpClientInterface->setParameters(activeInput, param.toString());
+        }
+    }
+
+}
+
+audio_policy_forced_cfg_t AudioPolicyManager::getForceUse(audio_policy_force_use_t usage)
+{
+    return mForceUse[usage];
+}
+
+void AudioPolicyManager::setSystemProperty(const char* property, const char* value)
+{
+    ALOGV("setSystemProperty() property %s, value %s", property, value);
+}
+
+// Find a direct output profile compatible with the parameters passed, even if the input flags do
+// not explicitly request a direct output
+AudioPolicyManager::IOProfile *AudioPolicyManager::getProfileForDirectOutput(
+                                                               audio_devices_t device,
+                                                               uint32_t samplingRate,
+                                                               audio_format_t format,
+                                                               audio_channel_mask_t channelMask,
+                                                               audio_output_flags_t flags)
+{
+    for (size_t i = 0; i < mHwModules.size(); i++) {
+        if (mHwModules[i]->mHandle == 0) {
+            continue;
+        }
+        for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
+            IOProfile *profile = mHwModules[i]->mOutputProfiles[j];
+            if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+                if (profile->isCompatibleProfile(device, samplingRate, format,
+                                           channelMask,
+                                           AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
+                    if (mAvailableOutputDevices & profile->mSupportedDevices) {
+                        return mHwModules[i]->mOutputProfiles[j];
+                    }
+                }
+            } else {
+                if (profile->isCompatibleProfile(device, samplingRate, format,
+                                           channelMask,
+                                           AUDIO_OUTPUT_FLAG_DIRECT)) {
+                    if (mAvailableOutputDevices & profile->mSupportedDevices) {
+                        return mHwModules[i]->mOutputProfiles[j];
+                    }
+                }
+            }
+        }
+    }
+    return 0;
+}
+
+audio_io_handle_t AudioPolicyManager::getOutput(audio_stream_type_t stream,
+                                    uint32_t samplingRate,
+                                    audio_format_t format,
+                                    audio_channel_mask_t channelMask,
+                                    audio_output_flags_t flags,
+                                    const audio_offload_info_t *offloadInfo)
+{
+    audio_io_handle_t output = 0;
+    uint32_t latency = 0;
+    routing_strategy strategy = getStrategy(stream);
+    audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+    ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x",
+          device, stream, samplingRate, format, channelMask, flags);
+
+#ifdef AUDIO_POLICY_TEST
+    if (mCurOutput != 0) {
+        ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
+                mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
+
+        if (mTestOutputs[mCurOutput] == 0) {
+            ALOGV("getOutput() opening test output");
+            AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
+            outputDesc->mDevice = mTestDevice;
+            outputDesc->mSamplingRate = mTestSamplingRate;
+            outputDesc->mFormat = mTestFormat;
+            outputDesc->mChannelMask = mTestChannels;
+            outputDesc->mLatency = mTestLatencyMs;
+            outputDesc->mFlags =
+                    (audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0);
+            outputDesc->mRefCount[stream] = 0;
+            mTestOutputs[mCurOutput] = mpClientInterface->openOutput(0, &outputDesc->mDevice,
+                                            &outputDesc->mSamplingRate,
+                                            &outputDesc->mFormat,
+                                            &outputDesc->mChannelMask,
+                                            &outputDesc->mLatency,
+                                            outputDesc->mFlags,
+                                            offloadInfo);
+            if (mTestOutputs[mCurOutput]) {
+                AudioParameter outputCmd = AudioParameter();
+                outputCmd.addInt(String8("set_id"),mCurOutput);
+                mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
+                addOutput(mTestOutputs[mCurOutput], outputDesc);
+            }
+        }
+        return mTestOutputs[mCurOutput];
+    }
+#endif //AUDIO_POLICY_TEST
+
+    // open a direct output if required by specified parameters
+    //force direct flag if offload flag is set: offloading implies a direct output stream
+    // and all common behaviors are driven by checking only the direct flag
+    // this should normally be set appropriately in the policy configuration file
+    if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+        flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
+    }
+
+    // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+    // creating an offloaded track and tearing it down immediately after start when audioflinger
+    // detects there is an active non offloadable effect.
+    // FIXME: We should check the audio session here but we do not have it in this context.
+    // This may prevent offloading in rare situations where effects are left active by apps
+    // in the background.
+    IOProfile *profile = NULL;
+    if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
+            !isNonOffloadableEffectEnabled()) {
+        profile = getProfileForDirectOutput(device,
+                                           samplingRate,
+                                           format,
+                                           channelMask,
+                                           (audio_output_flags_t)flags);
+    }
+
+    if (profile != NULL) {
+        AudioOutputDescriptor *outputDesc = NULL;
+
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+            if (!desc->isDuplicated() && (profile == desc->mProfile)) {
+                outputDesc = desc;
+                // reuse direct output if currently open and configured with same parameters
+                if ((samplingRate == outputDesc->mSamplingRate) &&
+                        (format == outputDesc->mFormat) &&
+                        (channelMask == outputDesc->mChannelMask)) {
+                    outputDesc->mDirectOpenCount++;
+                    ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
+                    return mOutputs.keyAt(i);
+                }
+            }
+        }
+        // close direct output if currently open and configured with different parameters
+        if (outputDesc != NULL) {
+            closeOutput(outputDesc->mId);
+        }
+        outputDesc = new AudioOutputDescriptor(profile);
+        outputDesc->mDevice = device;
+        outputDesc->mSamplingRate = samplingRate;
+        outputDesc->mFormat = format;
+        outputDesc->mChannelMask = channelMask;
+        outputDesc->mLatency = 0;
+        outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
+        outputDesc->mRefCount[stream] = 0;
+        outputDesc->mStopTime[stream] = 0;
+        outputDesc->mDirectOpenCount = 1;
+        output = mpClientInterface->openOutput(profile->mModule->mHandle,
+                                        &outputDesc->mDevice,
+                                        &outputDesc->mSamplingRate,
+                                        &outputDesc->mFormat,
+                                        &outputDesc->mChannelMask,
+                                        &outputDesc->mLatency,
+                                        outputDesc->mFlags,
+                                        offloadInfo);
+
+        // only accept an output with the requested parameters
+        if (output == 0 ||
+            (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
+            (format != AUDIO_FORMAT_DEFAULT && format != outputDesc->mFormat) ||
+            (channelMask != 0 && channelMask != outputDesc->mChannelMask)) {
+            ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
+                    "format %d %d, channelMask %04x %04x", output, samplingRate,
+                    outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
+                    outputDesc->mChannelMask);
+            if (output != 0) {
+                mpClientInterface->closeOutput(output);
+            }
+            delete outputDesc;
+            return 0;
+        }
+        audio_io_handle_t srcOutput = getOutputForEffect();
+        addOutput(output, outputDesc);
+        audio_io_handle_t dstOutput = getOutputForEffect();
+        if (dstOutput == output) {
+            mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
+        }
+        mPreviousOutputs = mOutputs;
+        ALOGV("getOutput() returns new direct output %d", output);
+        return output;
+    }
+
+    // ignoring channel mask due to downmix capability in mixer
+
+    // open a non direct output
+
+    // for non direct outputs, only PCM is supported
+    if (audio_is_linear_pcm(format)) {
+        // get which output is suitable for the specified stream. The actual
+        // routing change will happen when startOutput() will be called
+        SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+
+        output = selectOutput(outputs, flags);
+    }
+    ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
+            "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
+
+    ALOGV("getOutput() returns output %d", output);
+
+    return output;
+}
+
+audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector<audio_io_handle_t>& outputs,
+                                                       audio_output_flags_t flags)
+{
+    // select one output among several that provide a path to a particular device or set of
+    // devices (the list was previously build by getOutputsForDevice()).
+    // The priority is as follows:
+    // 1: the output with the highest number of requested policy flags
+    // 2: the primary output
+    // 3: the first output in the list
+
+    if (outputs.size() == 0) {
+        return 0;
+    }
+    if (outputs.size() == 1) {
+        return outputs[0];
+    }
+
+    int maxCommonFlags = 0;
+    audio_io_handle_t outputFlags = 0;
+    audio_io_handle_t outputPrimary = 0;
+
+    for (size_t i = 0; i < outputs.size(); i++) {
+        AudioOutputDescriptor *outputDesc = mOutputs.valueFor(outputs[i]);
+        if (!outputDesc->isDuplicated()) {
+            int commonFlags = popcount(outputDesc->mProfile->mFlags & flags);
+            if (commonFlags > maxCommonFlags) {
+                outputFlags = outputs[i];
+                maxCommonFlags = commonFlags;
+                ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags);
+            }
+            if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
+                outputPrimary = outputs[i];
+            }
+        }
+    }
+
+    if (outputFlags != 0) {
+        return outputFlags;
+    }
+    if (outputPrimary != 0) {
+        return outputPrimary;
+    }
+
+    return outputs[0];
+}
+
+status_t AudioPolicyManager::startOutput(audio_io_handle_t output,
+                                             audio_stream_type_t stream,
+                                             int session)
+{
+    ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
+    ssize_t index = mOutputs.indexOfKey(output);
+    if (index < 0) {
+        ALOGW("startOutput() unknown output %d", output);
+        return BAD_VALUE;
+    }
+
+    AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+
+    // increment usage count for this stream on the requested output:
+    // NOTE that the usage count is the same for duplicated output and hardware output which is
+    // necessary for a correct control of hardware output routing by startOutput() and stopOutput()
+    outputDesc->changeRefCount(stream, 1);
+
+    if (outputDesc->mRefCount[stream] == 1) {
+        audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/);
+        routing_strategy strategy = getStrategy(stream);
+        bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
+                            (strategy == STRATEGY_SONIFICATION_RESPECTFUL);
+        uint32_t waitMs = 0;
+        bool force = false;
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+            if (desc != outputDesc) {
+                // force a device change if any other output is managed by the same hw
+                // module and has a current device selection that differs from selected device.
+                // In this case, the audio HAL must receive the new device selection so that it can
+                // change the device currently selected by the other active output.
+                if (outputDesc->sharesHwModuleWith(desc) &&
+                    desc->device() != newDevice) {
+                    force = true;
+                }
+                // wait for audio on other active outputs to be presented when starting
+                // a notification so that audio focus effect can propagate.
+                uint32_t latency = desc->latency();
+                if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
+                    waitMs = latency;
+                }
+            }
+        }
+        uint32_t muteWaitMs = setOutputDevice(output, newDevice, force);
+
+        // handle special case for sonification while in call
+        if (isInCall()) {
+            handleIncallSonification(stream, true, false);
+        }
+
+        // apply volume rules for current stream and device if necessary
+        checkAndSetVolume(stream,
+                          mStreams[stream].getVolumeIndex(newDevice),
+                          output,
+                          newDevice);
+
+        // update the outputs if starting an output with a stream that can affect notification
+        // routing
+        handleNotificationRoutingForStream(stream);
+        if (waitMs > muteWaitMs) {
+            usleep((waitMs - muteWaitMs) * 2 * 1000);
+        }
+    }
+    return NO_ERROR;
+}
+
+
+status_t AudioPolicyManager::stopOutput(audio_io_handle_t output,
+                                            audio_stream_type_t stream,
+                                            int session)
+{
+    ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
+    ssize_t index = mOutputs.indexOfKey(output);
+    if (index < 0) {
+        ALOGW("stopOutput() unknown output %d", output);
+        return BAD_VALUE;
+    }
+
+    AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+
+    // handle special case for sonification while in call
+    if (isInCall()) {
+        handleIncallSonification(stream, false, false);
+    }
+
+    if (outputDesc->mRefCount[stream] > 0) {
+        // decrement usage count of this stream on the output
+        outputDesc->changeRefCount(stream, -1);
+        // store time at which the stream was stopped - see isStreamActive()
+        if (outputDesc->mRefCount[stream] == 0) {
+            outputDesc->mStopTime[stream] = systemTime();
+            audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/);
+            // delay the device switch by twice the latency because stopOutput() is executed when
+            // the track stop() command is received and at that time the audio track buffer can
+            // still contain data that needs to be drained. The latency only covers the audio HAL
+            // and kernel buffers. Also the latency does not always include additional delay in the
+            // audio path (audio DSP, CODEC ...)
+            setOutputDevice(output, newDevice, false, outputDesc->mLatency*2);
+
+            // force restoring the device selection on other active outputs if it differs from the
+            // one being selected for this output
+            for (size_t i = 0; i < mOutputs.size(); i++) {
+                audio_io_handle_t curOutput = mOutputs.keyAt(i);
+                AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+                if (curOutput != output &&
+                        desc->isActive() &&
+                        outputDesc->sharesHwModuleWith(desc) &&
+                        (newDevice != desc->device())) {
+                    setOutputDevice(curOutput,
+                                    getNewDevice(curOutput, false /*fromCache*/),
+                                    true,
+                                    outputDesc->mLatency*2);
+                }
+            }
+            // update the outputs if stopping one with a stream that can affect notification routing
+            handleNotificationRoutingForStream(stream);
+        }
+        return NO_ERROR;
+    } else {
+        ALOGW("stopOutput() refcount is already 0 for output %d", output);
+        return INVALID_OPERATION;
+    }
+}
+
+void AudioPolicyManager::releaseOutput(audio_io_handle_t output)
+{
+    ALOGV("releaseOutput() %d", output);
+    ssize_t index = mOutputs.indexOfKey(output);
+    if (index < 0) {
+        ALOGW("releaseOutput() releasing unknown output %d", output);
+        return;
+    }
+
+#ifdef AUDIO_POLICY_TEST
+    int testIndex = testOutputIndex(output);
+    if (testIndex != 0) {
+        AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+        if (outputDesc->isActive()) {
+            mpClientInterface->closeOutput(output);
+            delete mOutputs.valueAt(index);
+            mOutputs.removeItem(output);
+            mTestOutputs[testIndex] = 0;
+        }
+        return;
+    }
+#endif //AUDIO_POLICY_TEST
+
+    AudioOutputDescriptor *desc = mOutputs.valueAt(index);
+    if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+        if (desc->mDirectOpenCount <= 0) {
+            ALOGW("releaseOutput() invalid open count %d for output %d",
+                                                              desc->mDirectOpenCount, output);
+            return;
+        }
+        if (--desc->mDirectOpenCount == 0) {
+            closeOutput(output);
+            // If effects where present on the output, audioflinger moved them to the primary
+            // output by default: move them back to the appropriate output.
+            audio_io_handle_t dstOutput = getOutputForEffect();
+            if (dstOutput != mPrimaryOutput) {
+                mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput, dstOutput);
+            }
+        }
+    }
+}
+
+
+audio_io_handle_t AudioPolicyManager::getInput(audio_source_t inputSource,
+                                    uint32_t samplingRate,
+                                    audio_format_t format,
+                                    audio_channel_mask_t channelMask,
+                                    audio_in_acoustics_t acoustics)
+{
+    audio_io_handle_t input = 0;
+    audio_devices_t device = getDeviceForInputSource(inputSource);
+
+    ALOGV("getInput() inputSource %d, samplingRate %d, format %d, channelMask %x, acoustics %x",
+          inputSource, samplingRate, format, channelMask, acoustics);
+
+    if (device == AUDIO_DEVICE_NONE) {
+        ALOGW("getInput() could not find device for inputSource %d", inputSource);
+        return 0;
+    }
+
+    // adapt channel selection to input source
+    switch(inputSource) {
+    case AUDIO_SOURCE_VOICE_UPLINK:
+        channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK;
+        break;
+    case AUDIO_SOURCE_VOICE_DOWNLINK:
+        channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK;
+        break;
+    case AUDIO_SOURCE_VOICE_CALL:
+        channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK;
+        break;
+    default:
+        break;
+    }
+
+    IOProfile *profile = getInputProfile(device,
+                                         samplingRate,
+                                         format,
+                                         channelMask);
+    if (profile == NULL) {
+        ALOGW("getInput() could not find profile for device %04x, samplingRate %d, format %d, "
+                "channelMask %04x",
+                device, samplingRate, format, channelMask);
+        return 0;
+    }
+
+    if (profile->mModule->mHandle == 0) {
+        ALOGE("getInput(): HW module %s not opened", profile->mModule->mName);
+        return 0;
+    }
+
+    AudioInputDescriptor *inputDesc = new AudioInputDescriptor(profile);
+
+    inputDesc->mInputSource = inputSource;
+    inputDesc->mDevice = device;
+    inputDesc->mSamplingRate = samplingRate;
+    inputDesc->mFormat = format;
+    inputDesc->mChannelMask = channelMask;
+    inputDesc->mRefCount = 0;
+    input = mpClientInterface->openInput(profile->mModule->mHandle,
+                                    &inputDesc->mDevice,
+                                    &inputDesc->mSamplingRate,
+                                    &inputDesc->mFormat,
+                                    &inputDesc->mChannelMask);
+
+    // only accept input with the exact requested set of parameters
+    if (input == 0 ||
+        (samplingRate != inputDesc->mSamplingRate) ||
+        (format != inputDesc->mFormat) ||
+        (channelMask != inputDesc->mChannelMask)) {
+        ALOGI("getInput() failed opening input: samplingRate %d, format %d, channelMask %x",
+                samplingRate, format, channelMask);
+        if (input != 0) {
+            mpClientInterface->closeInput(input);
+        }
+        delete inputDesc;
+        return 0;
+    }
+    mInputs.add(input, inputDesc);
+    return input;
+}
+
+status_t AudioPolicyManager::startInput(audio_io_handle_t input)
+{
+    ALOGV("startInput() input %d", input);
+    ssize_t index = mInputs.indexOfKey(input);
+    if (index < 0) {
+        ALOGW("startInput() unknown input %d", input);
+        return BAD_VALUE;
+    }
+    AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
+
+#ifdef AUDIO_POLICY_TEST
+    if (mTestInput == 0)
+#endif //AUDIO_POLICY_TEST
+    {
+        // refuse 2 active AudioRecord clients at the same time except if the active input
+        // uses AUDIO_SOURCE_HOTWORD in which case it is closed.
+        audio_io_handle_t activeInput = getActiveInput();
+        if (!isVirtualInputDevice(inputDesc->mDevice) && activeInput != 0) {
+            AudioInputDescriptor *activeDesc = mInputs.valueFor(activeInput);
+            if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) {
+                ALOGW("startInput() preempting already started low-priority input %d", activeInput);
+                stopInput(activeInput);
+                releaseInput(activeInput);
+            } else {
+                ALOGW("startInput() input %d failed: other input already started", input);
+                return INVALID_OPERATION;
+            }
+        }
+    }
+
+    audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
+    if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
+        inputDesc->mDevice = newDevice;
+    }
+
+    // automatically enable the remote submix output when input is started
+    if (audio_is_remote_submix_device(inputDesc->mDevice)) {
+        setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+                AUDIO_POLICY_DEVICE_STATE_AVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS);
+    }
+
+    AudioParameter param = AudioParameter();
+    param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice);
+
+    int aliasSource = (inputDesc->mInputSource == AUDIO_SOURCE_HOTWORD) ?
+                                        AUDIO_SOURCE_VOICE_RECOGNITION : inputDesc->mInputSource;
+
+    param.addInt(String8(AudioParameter::keyInputSource), aliasSource);
+    ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
+
+    mpClientInterface->setParameters(input, param.toString());
+
+    inputDesc->mRefCount = 1;
+    return NO_ERROR;
+}
+
+status_t AudioPolicyManager::stopInput(audio_io_handle_t input)
+{
+    ALOGV("stopInput() input %d", input);
+    ssize_t index = mInputs.indexOfKey(input);
+    if (index < 0) {
+        ALOGW("stopInput() unknown input %d", input);
+        return BAD_VALUE;
+    }
+    AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
+
+    if (inputDesc->mRefCount == 0) {
+        ALOGW("stopInput() input %d already stopped", input);
+        return INVALID_OPERATION;
+    } else {
+        // automatically disable the remote submix output when input is stopped
+        if (audio_is_remote_submix_device(inputDesc->mDevice)) {
+            setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+                    AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS);
+        }
+
+        AudioParameter param = AudioParameter();
+        param.addInt(String8(AudioParameter::keyRouting), 0);
+        mpClientInterface->setParameters(input, param.toString());
+        inputDesc->mRefCount = 0;
+        return NO_ERROR;
+    }
+}
+
+void AudioPolicyManager::releaseInput(audio_io_handle_t input)
+{
+    ALOGV("releaseInput() %d", input);
+    ssize_t index = mInputs.indexOfKey(input);
+    if (index < 0) {
+        ALOGW("releaseInput() releasing unknown input %d", input);
+        return;
+    }
+    mpClientInterface->closeInput(input);
+    delete mInputs.valueAt(index);
+    mInputs.removeItem(input);
+    ALOGV("releaseInput() exit");
+}
+
+void AudioPolicyManager::initStreamVolume(audio_stream_type_t stream,
+                                            int indexMin,
+                                            int indexMax)
+{
+    ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
+    if (indexMin < 0 || indexMin >= indexMax) {
+        ALOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax);
+        return;
+    }
+    mStreams[stream].mIndexMin = indexMin;
+    mStreams[stream].mIndexMax = indexMax;
+}
+
+status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream,
+                                                      int index,
+                                                      audio_devices_t device)
+{
+
+    if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
+        return BAD_VALUE;
+    }
+    if (!audio_is_output_device(device)) {
+        return BAD_VALUE;
+    }
+
+    // Force max volume if stream cannot be muted
+    if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax;
+
+    ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d",
+          stream, device, index);
+
+    // if device is AUDIO_DEVICE_OUT_DEFAULT set default value and
+    // clear all device specific values
+    if (device == AUDIO_DEVICE_OUT_DEFAULT) {
+        mStreams[stream].mIndexCur.clear();
+    }
+    mStreams[stream].mIndexCur.add(device, index);
+
+    // compute and apply stream volume on all outputs according to connected device
+    status_t status = NO_ERROR;
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        audio_devices_t curDevice =
+                getDeviceForVolume(mOutputs.valueAt(i)->device());
+        if ((device == AUDIO_DEVICE_OUT_DEFAULT) || (device == curDevice)) {
+            status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice);
+            if (volStatus != NO_ERROR) {
+                status = volStatus;
+            }
+        }
+    }
+    return status;
+}
+
+status_t AudioPolicyManager::getStreamVolumeIndex(audio_stream_type_t stream,
+                                                      int *index,
+                                                      audio_devices_t device)
+{
+    if (index == NULL) {
+        return BAD_VALUE;
+    }
+    if (!audio_is_output_device(device)) {
+        return BAD_VALUE;
+    }
+    // if device is AUDIO_DEVICE_OUT_DEFAULT, return volume for device corresponding to
+    // the strategy the stream belongs to.
+    if (device == AUDIO_DEVICE_OUT_DEFAULT) {
+        device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
+    }
+    device = getDeviceForVolume(device);
+
+    *index =  mStreams[stream].getVolumeIndex(device);
+    ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index);
+    return NO_ERROR;
+}
+
+audio_io_handle_t AudioPolicyManager::selectOutputForEffects(
+                                            const SortedVector<audio_io_handle_t>& outputs)
+{
+    // select one output among several suitable for global effects.
+    // The priority is as follows:
+    // 1: An offloaded output. If the effect ends up not being offloadable,
+    //    AudioFlinger will invalidate the track and the offloaded output
+    //    will be closed causing the effect to be moved to a PCM output.
+    // 2: A deep buffer output
+    // 3: the first output in the list
+
+    if (outputs.size() == 0) {
+        return 0;
+    }
+
+    audio_io_handle_t outputOffloaded = 0;
+    audio_io_handle_t outputDeepBuffer = 0;
+
+    for (size_t i = 0; i < outputs.size(); i++) {
+        AudioOutputDescriptor *desc = mOutputs.valueFor(outputs[i]);
+        ALOGV("selectOutputForEffects outputs[%d] flags %x", i, desc->mFlags);
+        if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+            outputOffloaded = outputs[i];
+        }
+        if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) {
+            outputDeepBuffer = outputs[i];
+        }
+    }
+
+    ALOGV("selectOutputForEffects outputOffloaded %d outputDeepBuffer %d",
+          outputOffloaded, outputDeepBuffer);
+    if (outputOffloaded != 0) {
+        return outputOffloaded;
+    }
+    if (outputDeepBuffer != 0) {
+        return outputDeepBuffer;
+    }
+
+    return outputs[0];
+}
+
+audio_io_handle_t AudioPolicyManager::getOutputForEffect(const effect_descriptor_t *desc)
+{
+    // apply simple rule where global effects are attached to the same output as MUSIC streams
+
+    routing_strategy strategy = getStrategy(AUDIO_STREAM_MUSIC);
+    audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+    SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(device, mOutputs);
+
+    audio_io_handle_t output = selectOutputForEffects(dstOutputs);
+    ALOGV("getOutputForEffect() got output %d for fx %s flags %x",
+          output, (desc == NULL) ? "unspecified" : desc->name,  (desc == NULL) ? 0 : desc->flags);
+
+    return output;
+}
+
+status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc,
+                                audio_io_handle_t io,
+                                uint32_t strategy,
+                                int session,
+                                int id)
+{
+    ssize_t index = mOutputs.indexOfKey(io);
+    if (index < 0) {
+        index = mInputs.indexOfKey(io);
+        if (index < 0) {
+            ALOGW("registerEffect() unknown io %d", io);
+            return INVALID_OPERATION;
+        }
+    }
+
+    if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) {
+        ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB",
+                desc->name, desc->memoryUsage);
+        return INVALID_OPERATION;
+    }
+    mTotalEffectsMemory += desc->memoryUsage;
+    ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d",
+            desc->name, io, strategy, session, id);
+    ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory);
+
+    EffectDescriptor *pDesc = new EffectDescriptor();
+    memcpy (&pDesc->mDesc, desc, sizeof(effect_descriptor_t));
+    pDesc->mIo = io;
+    pDesc->mStrategy = (routing_strategy)strategy;
+    pDesc->mSession = session;
+    pDesc->mEnabled = false;
+
+    mEffects.add(id, pDesc);
+
+    return NO_ERROR;
+}
+
+status_t AudioPolicyManager::unregisterEffect(int id)
+{
+    ssize_t index = mEffects.indexOfKey(id);
+    if (index < 0) {
+        ALOGW("unregisterEffect() unknown effect ID %d", id);
+        return INVALID_OPERATION;
+    }
+
+    EffectDescriptor *pDesc = mEffects.valueAt(index);
+
+    setEffectEnabled(pDesc, false);
+
+    if (mTotalEffectsMemory < pDesc->mDesc.memoryUsage) {
+        ALOGW("unregisterEffect() memory %d too big for total %d",
+                pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
+        pDesc->mDesc.memoryUsage = mTotalEffectsMemory;
+    }
+    mTotalEffectsMemory -= pDesc->mDesc.memoryUsage;
+    ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d",
+            pDesc->mDesc.name, id, pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
+
+    mEffects.removeItem(id);
+    delete pDesc;
+
+    return NO_ERROR;
+}
+
+status_t AudioPolicyManager::setEffectEnabled(int id, bool enabled)
+{
+    ssize_t index = mEffects.indexOfKey(id);
+    if (index < 0) {
+        ALOGW("unregisterEffect() unknown effect ID %d", id);
+        return INVALID_OPERATION;
+    }
+
+    return setEffectEnabled(mEffects.valueAt(index), enabled);
+}
+
+status_t AudioPolicyManager::setEffectEnabled(EffectDescriptor *pDesc, bool enabled)
+{
+    if (enabled == pDesc->mEnabled) {
+        ALOGV("setEffectEnabled(%s) effect already %s",
+             enabled?"true":"false", enabled?"enabled":"disabled");
+        return INVALID_OPERATION;
+    }
+
+    if (enabled) {
+        if (mTotalEffectsCpuLoad + pDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) {
+            ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS",
+                 pDesc->mDesc.name, (float)pDesc->mDesc.cpuLoad/10);
+            return INVALID_OPERATION;
+        }
+        mTotalEffectsCpuLoad += pDesc->mDesc.cpuLoad;
+        ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad);
+    } else {
+        if (mTotalEffectsCpuLoad < pDesc->mDesc.cpuLoad) {
+            ALOGW("setEffectEnabled(false) CPU load %d too high for total %d",
+                    pDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad);
+            pDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad;
+        }
+        mTotalEffectsCpuLoad -= pDesc->mDesc.cpuLoad;
+        ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad);
+    }
+    pDesc->mEnabled = enabled;
+    return NO_ERROR;
+}
+
+bool AudioPolicyManager::isNonOffloadableEffectEnabled()
+{
+    for (size_t i = 0; i < mEffects.size(); i++) {
+        const EffectDescriptor * const pDesc = mEffects.valueAt(i);
+        if (pDesc->mEnabled && (pDesc->mStrategy == STRATEGY_MEDIA) &&
+                ((pDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) {
+            ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d",
+                  pDesc->mDesc.name, pDesc->mSession);
+            return true;
+        }
+    }
+    return false;
+}
+
+bool AudioPolicyManager::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
+{
+    nsecs_t sysTime = systemTime();
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
+        if (outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
+            return true;
+        }
+    }
+    return false;
+}
+
+bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream,
+                                                    uint32_t inPastMs) const
+{
+    nsecs_t sysTime = systemTime();
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
+        if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
+                outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
+            return true;
+        }
+    }
+    return false;
+}
+
+bool AudioPolicyManager::isSourceActive(audio_source_t source) const
+{
+    for (size_t i = 0; i < mInputs.size(); i++) {
+        const AudioInputDescriptor * inputDescriptor = mInputs.valueAt(i);
+        if ((inputDescriptor->mInputSource == (int)source ||
+                (source == AUDIO_SOURCE_VOICE_RECOGNITION &&
+                 inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD))
+             && (inputDescriptor->mRefCount > 0)) {
+            return true;
+        }
+    }
+    return false;
+}
+
+
+status_t AudioPolicyManager::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this);
+    result.append(buffer);
+
+    snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " A2DP device address: %s\n", mA2dpDeviceAddress.string());
+    result.append(buffer);
+    snprintf(buffer, SIZE, " SCO device address: %s\n", mScoDeviceAddress.string());
+    result.append(buffer);
+    snprintf(buffer, SIZE, " USB audio ALSA %s\n", mUsbCardAndDevice.string());
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Output devices: %08x\n", mAvailableOutputDevices);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Input devices: %08x\n", mAvailableInputDevices);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Force use for communications %d\n",
+             mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA]);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD]);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK]);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Force use for system %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM]);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+
+
+    snprintf(buffer, SIZE, "\nHW Modules dump:\n");
+    write(fd, buffer, strlen(buffer));
+    for (size_t i = 0; i < mHwModules.size(); i++) {
+        snprintf(buffer, SIZE, "- HW Module %d:\n", i + 1);
+        write(fd, buffer, strlen(buffer));
+        mHwModules[i]->dump(fd);
+    }
+
+    snprintf(buffer, SIZE, "\nOutputs dump:\n");
+    write(fd, buffer, strlen(buffer));
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i));
+        write(fd, buffer, strlen(buffer));
+        mOutputs.valueAt(i)->dump(fd);
+    }
+
+    snprintf(buffer, SIZE, "\nInputs dump:\n");
+    write(fd, buffer, strlen(buffer));
+    for (size_t i = 0; i < mInputs.size(); i++) {
+        snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i));
+        write(fd, buffer, strlen(buffer));
+        mInputs.valueAt(i)->dump(fd);
+    }
+
+    snprintf(buffer, SIZE, "\nStreams dump:\n");
+    write(fd, buffer, strlen(buffer));
+    snprintf(buffer, SIZE,
+             " Stream  Can be muted  Index Min  Index Max  Index Cur [device : index]...\n");
+    write(fd, buffer, strlen(buffer));
+    for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+        snprintf(buffer, SIZE, " %02d      ", i);
+        write(fd, buffer, strlen(buffer));
+        mStreams[i].dump(fd);
+    }
+
+    snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n",
+            (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory);
+    write(fd, buffer, strlen(buffer));
+
+    snprintf(buffer, SIZE, "Registered effects:\n");
+    write(fd, buffer, strlen(buffer));
+    for (size_t i = 0; i < mEffects.size(); i++) {
+        snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i));
+        write(fd, buffer, strlen(buffer));
+        mEffects.valueAt(i)->dump(fd);
+    }
+
+
+    return NO_ERROR;
+}
+
+// This function checks for the parameters which can be offloaded.
+// This can be enhanced depending on the capability of the DSP and policy
+// of the system.
+bool AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadInfo)
+{
+    ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
+     " BitRate=%u, duration=%lld us, has_video=%d",
+     offloadInfo.sample_rate, offloadInfo.channel_mask,
+     offloadInfo.format,
+     offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
+     offloadInfo.has_video);
+
+    // Check if offload has been disabled
+    char propValue[PROPERTY_VALUE_MAX];
+    if (property_get("audio.offload.disable", propValue, "0")) {
+        if (atoi(propValue) != 0) {
+            ALOGV("offload disabled by audio.offload.disable=%s", propValue );
+            return false;
+        }
+    }
+
+    // Check if stream type is music, then only allow offload as of now.
+    if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
+    {
+        ALOGV("isOffloadSupported: stream_type != MUSIC, returning false");
+        return false;
+    }
+
+    //TODO: enable audio offloading with video when ready
+    if (offloadInfo.has_video)
+    {
+        ALOGV("isOffloadSupported: has_video == true, returning false");
+        return false;
+    }
+
+    //If duration is less than minimum value defined in property, return false
+    if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
+        if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
+            ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
+            return false;
+        }
+    } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
+        ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
+        return false;
+    }
+
+    // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+    // creating an offloaded track and tearing it down immediately after start when audioflinger
+    // detects there is an active non offloadable effect.
+    // FIXME: We should check the audio session here but we do not have it in this context.
+    // This may prevent offloading in rare situations where effects are left active by apps
+    // in the background.
+    if (isNonOffloadableEffectEnabled()) {
+        return false;
+    }
+
+    // See if there is a profile to support this.
+    // AUDIO_DEVICE_NONE
+    IOProfile *profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
+                                            offloadInfo.sample_rate,
+                                            offloadInfo.format,
+                                            offloadInfo.channel_mask,
+                                            AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+    ALOGV("isOffloadSupported() profile %sfound", profile != NULL ? "" : "NOT ");
+    return (profile != NULL);
+}
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManager
+// ----------------------------------------------------------------------------
+
+AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
+    :
+#ifdef AUDIO_POLICY_TEST
+    Thread(false),
+#endif //AUDIO_POLICY_TEST
+    mPrimaryOutput((audio_io_handle_t)0),
+    mAvailableOutputDevices(AUDIO_DEVICE_NONE),
+    mPhoneState(AUDIO_MODE_NORMAL),
+    mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
+    mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
+    mA2dpSuspended(false), mHasA2dp(false), mHasUsb(false), mHasRemoteSubmix(false),
+    mSpeakerDrcEnabled(false)
+{
+    mpClientInterface = clientInterface;
+
+    for (int i = 0; i < AUDIO_POLICY_FORCE_USE_CNT; i++) {
+        mForceUse[i] = AUDIO_POLICY_FORCE_NONE;
+    }
+
+    mA2dpDeviceAddress = String8("");
+    mScoDeviceAddress = String8("");
+    mUsbCardAndDevice = String8("");
+
+    if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) {
+        if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) {
+            ALOGE("could not load audio policy configuration file, setting defaults");
+            defaultAudioPolicyConfig();
+        }
+    }
+
+    // must be done after reading the policy
+    initializeVolumeCurves();
+
+    // open all output streams needed to access attached devices
+    for (size_t i = 0; i < mHwModules.size(); i++) {
+        mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName);
+        if (mHwModules[i]->mHandle == 0) {
+            ALOGW("could not open HW module %s", mHwModules[i]->mName);
+            continue;
+        }
+        // open all output streams needed to access attached devices
+        // except for direct output streams that are only opened when they are actually
+        // required by an app.
+        for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+        {
+            const IOProfile *outProfile = mHwModules[i]->mOutputProfiles[j];
+
+            if ((outProfile->mSupportedDevices & mAttachedOutputDevices) &&
+                    ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0)) {
+                AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(outProfile);
+                outputDesc->mDevice = (audio_devices_t)(mDefaultOutputDevice &
+                                                            outProfile->mSupportedDevices);
+                audio_io_handle_t output = mpClientInterface->openOutput(
+                                                outProfile->mModule->mHandle,
+                                                &outputDesc->mDevice,
+                                                &outputDesc->mSamplingRate,
+                                                &outputDesc->mFormat,
+                                                &outputDesc->mChannelMask,
+                                                &outputDesc->mLatency,
+                                                outputDesc->mFlags);
+                if (output == 0) {
+                    delete outputDesc;
+                } else {
+                    mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices |
+                                            (outProfile->mSupportedDevices & mAttachedOutputDevices));
+                    if (mPrimaryOutput == 0 &&
+                            outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
+                        mPrimaryOutput = output;
+                    }
+                    addOutput(output, outputDesc);
+                    setOutputDevice(output,
+                                    (audio_devices_t)(mDefaultOutputDevice &
+                                                        outProfile->mSupportedDevices),
+                                    true);
+                }
+            }
+        }
+    }
+
+    ALOGE_IF((mAttachedOutputDevices & ~mAvailableOutputDevices),
+             "Not output found for attached devices %08x",
+             (mAttachedOutputDevices & ~mAvailableOutputDevices));
+
+    ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output");
+
+    updateDevicesAndOutputs();
+
+#ifdef AUDIO_POLICY_TEST
+    if (mPrimaryOutput != 0) {
+        AudioParameter outputCmd = AudioParameter();
+        outputCmd.addInt(String8("set_id"), 0);
+        mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
+
+        mTestDevice = AUDIO_DEVICE_OUT_SPEAKER;
+        mTestSamplingRate = 44100;
+        mTestFormat = AUDIO_FORMAT_PCM_16_BIT;
+        mTestChannels =  AUDIO_CHANNEL_OUT_STEREO;
+        mTestLatencyMs = 0;
+        mCurOutput = 0;
+        mDirectOutput = false;
+        for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
+            mTestOutputs[i] = 0;
+        }
+
+        const size_t SIZE = 256;
+        char buffer[SIZE];
+        snprintf(buffer, SIZE, "AudioPolicyManagerTest");
+        run(buffer, ANDROID_PRIORITY_AUDIO);
+    }
+#endif //AUDIO_POLICY_TEST
+}
+
+AudioPolicyManager::~AudioPolicyManager()
+{
+#ifdef AUDIO_POLICY_TEST
+    exit();
+#endif //AUDIO_POLICY_TEST
+   for (size_t i = 0; i < mOutputs.size(); i++) {
+        mpClientInterface->closeOutput(mOutputs.keyAt(i));
+        delete mOutputs.valueAt(i);
+   }
+   for (size_t i = 0; i < mInputs.size(); i++) {
+        mpClientInterface->closeInput(mInputs.keyAt(i));
+        delete mInputs.valueAt(i);
+   }
+   for (size_t i = 0; i < mHwModules.size(); i++) {
+        delete mHwModules[i];
+   }
+}
+
+status_t AudioPolicyManager::initCheck()
+{
+    return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR;
+}
+
+#ifdef AUDIO_POLICY_TEST
+bool AudioPolicyManager::threadLoop()
+{
+    ALOGV("entering threadLoop()");
+    while (!exitPending())
+    {
+        String8 command;
+        int valueInt;
+        String8 value;
+
+        Mutex::Autolock _l(mLock);
+        mWaitWorkCV.waitRelative(mLock, milliseconds(50));
+
+        command = mpClientInterface->getParameters(0, String8("test_cmd_policy"));
+        AudioParameter param = AudioParameter(command);
+
+        if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR &&
+            valueInt != 0) {
+            ALOGV("Test command %s received", command.string());
+            String8 target;
+            if (param.get(String8("target"), target) != NO_ERROR) {
+                target = "Manager";
+            }
+            if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) {
+                param.remove(String8("test_cmd_policy_output"));
+                mCurOutput = valueInt;
+            }
+            if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) {
+                param.remove(String8("test_cmd_policy_direct"));
+                if (value == "false") {
+                    mDirectOutput = false;
+                } else if (value == "true") {
+                    mDirectOutput = true;
+                }
+            }
+            if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) {
+                param.remove(String8("test_cmd_policy_input"));
+                mTestInput = valueInt;
+            }
+
+            if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) {
+                param.remove(String8("test_cmd_policy_format"));
+                int format = AUDIO_FORMAT_INVALID;
+                if (value == "PCM 16 bits") {
+                    format = AUDIO_FORMAT_PCM_16_BIT;
+                } else if (value == "PCM 8 bits") {
+                    format = AUDIO_FORMAT_PCM_8_BIT;
+                } else if (value == "Compressed MP3") {
+                    format = AUDIO_FORMAT_MP3;
+                }
+                if (format != AUDIO_FORMAT_INVALID) {
+                    if (target == "Manager") {
+                        mTestFormat = format;
+                    } else if (mTestOutputs[mCurOutput] != 0) {
+                        AudioParameter outputParam = AudioParameter();
+                        outputParam.addInt(String8("format"), format);
+                        mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
+                    }
+                }
+            }
+            if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) {
+                param.remove(String8("test_cmd_policy_channels"));
+                int channels = 0;
+
+                if (value == "Channels Stereo") {
+                    channels =  AUDIO_CHANNEL_OUT_STEREO;
+                } else if (value == "Channels Mono") {
+                    channels =  AUDIO_CHANNEL_OUT_MONO;
+                }
+                if (channels != 0) {
+                    if (target == "Manager") {
+                        mTestChannels = channels;
+                    } else if (mTestOutputs[mCurOutput] != 0) {
+                        AudioParameter outputParam = AudioParameter();
+                        outputParam.addInt(String8("channels"), channels);
+                        mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
+                    }
+                }
+            }
+            if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) {
+                param.remove(String8("test_cmd_policy_sampleRate"));
+                if (valueInt >= 0 && valueInt <= 96000) {
+                    int samplingRate = valueInt;
+                    if (target == "Manager") {
+                        mTestSamplingRate = samplingRate;
+                    } else if (mTestOutputs[mCurOutput] != 0) {
+                        AudioParameter outputParam = AudioParameter();
+                        outputParam.addInt(String8("sampling_rate"), samplingRate);
+                        mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
+                    }
+                }
+            }
+
+            if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) {
+                param.remove(String8("test_cmd_policy_reopen"));
+
+                AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput);
+                mpClientInterface->closeOutput(mPrimaryOutput);
+
+                audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle;
+
+                delete mOutputs.valueFor(mPrimaryOutput);
+                mOutputs.removeItem(mPrimaryOutput);
+
+                AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
+                outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER;
+                mPrimaryOutput = mpClientInterface->openOutput(moduleHandle,
+                                                &outputDesc->mDevice,
+                                                &outputDesc->mSamplingRate,
+                                                &outputDesc->mFormat,
+                                                &outputDesc->mChannelMask,
+                                                &outputDesc->mLatency,
+                                                outputDesc->mFlags);
+                if (mPrimaryOutput == 0) {
+                    ALOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d",
+                            outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask);
+                } else {
+                    AudioParameter outputCmd = AudioParameter();
+                    outputCmd.addInt(String8("set_id"), 0);
+                    mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
+                    addOutput(mPrimaryOutput, outputDesc);
+                }
+            }
+
+
+            mpClientInterface->setParameters(0, String8("test_cmd_policy="));
+        }
+    }
+    return false;
+}
+
+void AudioPolicyManager::exit()
+{
+    {
+        AutoMutex _l(mLock);
+        requestExit();
+        mWaitWorkCV.signal();
+    }
+    requestExitAndWait();
+}
+
+int AudioPolicyManager::testOutputIndex(audio_io_handle_t output)
+{
+    for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
+        if (output == mTestOutputs[i]) return i;
+    }
+    return 0;
+}
+#endif //AUDIO_POLICY_TEST
+
+// ---
+
+void AudioPolicyManager::addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc)
+{
+    outputDesc->mId = id;
+    mOutputs.add(id, outputDesc);
+}
+
+
+status_t AudioPolicyManager::checkOutputsForDevice(audio_devices_t device,
+                                                       audio_policy_dev_state_t state,
+                                                       SortedVector<audio_io_handle_t>& outputs,
+                                                       const String8 paramStr)
+{
+    AudioOutputDescriptor *desc;
+
+    if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
+        // first list already open outputs that can be routed to this device
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            desc = mOutputs.valueAt(i);
+            if (!desc->isDuplicated() && (desc->mProfile->mSupportedDevices & device)) {
+                ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i));
+                outputs.add(mOutputs.keyAt(i));
+            }
+        }
+        // then look for output profiles that can be routed to this device
+        SortedVector<IOProfile *> profiles;
+        for (size_t i = 0; i < mHwModules.size(); i++)
+        {
+            if (mHwModules[i]->mHandle == 0) {
+                continue;
+            }
+            for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+            {
+                if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices & device) {
+                    ALOGV("checkOutputsForDevice(): adding profile %d from module %d", j, i);
+                    profiles.add(mHwModules[i]->mOutputProfiles[j]);
+                }
+            }
+        }
+
+        if (profiles.isEmpty() && outputs.isEmpty()) {
+            ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
+            return BAD_VALUE;
+        }
+
+        // open outputs for matching profiles if needed. Direct outputs are also opened to
+        // query for dynamic parameters and will be closed later by setDeviceConnectionState()
+        for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
+            IOProfile *profile = profiles[profile_index];
+
+            // nothing to do if one output is already opened for this profile
+            size_t j;
+            for (j = 0; j < mOutputs.size(); j++) {
+                desc = mOutputs.valueAt(j);
+                if (!desc->isDuplicated() && desc->mProfile == profile) {
+                    break;
+                }
+            }
+            if (j != mOutputs.size()) {
+                continue;
+            }
+
+            ALOGV("opening output for device %08x with params %s", device, paramStr.string());
+            desc = new AudioOutputDescriptor(profile);
+            desc->mDevice = device;
+            audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER;
+            offloadInfo.sample_rate = desc->mSamplingRate;
+            offloadInfo.format = desc->mFormat;
+            offloadInfo.channel_mask = desc->mChannelMask;
+
+            audio_io_handle_t output = mpClientInterface->openOutput(profile->mModule->mHandle,
+                                                                       &desc->mDevice,
+                                                                       &desc->mSamplingRate,
+                                                                       &desc->mFormat,
+                                                                       &desc->mChannelMask,
+                                                                       &desc->mLatency,
+                                                                       desc->mFlags,
+                                                                       &offloadInfo);
+            if (output != 0) {
+                if (!paramStr.isEmpty()) {
+                    mpClientInterface->setParameters(output, paramStr);
+                }
+
+                if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+                    String8 reply;
+                    char *value;
+                    if (profile->mSamplingRates[0] == 0) {
+                        reply = mpClientInterface->getParameters(output,
+                                                String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
+                        ALOGV("checkOutputsForDevice() direct output sup sampling rates %s",
+                                  reply.string());
+                        value = strpbrk((char *)reply.string(), "=");
+                        if (value != NULL) {
+                            loadSamplingRates(value + 1, profile);
+                        }
+                    }
+                    if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
+                        reply = mpClientInterface->getParameters(output,
+                                                       String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
+                        ALOGV("checkOutputsForDevice() direct output sup formats %s",
+                                  reply.string());
+                        value = strpbrk((char *)reply.string(), "=");
+                        if (value != NULL) {
+                            loadFormats(value + 1, profile);
+                        }
+                    }
+                    if (profile->mChannelMasks[0] == 0) {
+                        reply = mpClientInterface->getParameters(output,
+                                                      String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
+                        ALOGV("checkOutputsForDevice() direct output sup channel masks %s",
+                                  reply.string());
+                        value = strpbrk((char *)reply.string(), "=");
+                        if (value != NULL) {
+                            loadOutChannels(value + 1, profile);
+                        }
+                    }
+                    if (((profile->mSamplingRates[0] == 0) &&
+                             (profile->mSamplingRates.size() < 2)) ||
+                         ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) &&
+                             (profile->mFormats.size() < 2)) ||
+                         ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) &&
+                             (profile->mChannelMasks.size() < 2))) {
+                        ALOGW("checkOutputsForDevice() direct output missing param");
+                        mpClientInterface->closeOutput(output);
+                        output = 0;
+                    } else {
+                        addOutput(output, desc);
+                    }
+                } else {
+                    audio_io_handle_t duplicatedOutput = 0;
+                    // add output descriptor
+                    addOutput(output, desc);
+                    // set initial stream volume for device
+                    applyStreamVolumes(output, device, 0, true);
+
+                    //TODO: configure audio effect output stage here
+
+                    // open a duplicating output thread for the new output and the primary output
+                    duplicatedOutput = mpClientInterface->openDuplicateOutput(output,
+                                                                              mPrimaryOutput);
+                    if (duplicatedOutput != 0) {
+                        // add duplicated output descriptor
+                        AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor(NULL);
+                        dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput);
+                        dupOutputDesc->mOutput2 = mOutputs.valueFor(output);
+                        dupOutputDesc->mSamplingRate = desc->mSamplingRate;
+                        dupOutputDesc->mFormat = desc->mFormat;
+                        dupOutputDesc->mChannelMask = desc->mChannelMask;
+                        dupOutputDesc->mLatency = desc->mLatency;
+                        addOutput(duplicatedOutput, dupOutputDesc);
+                        applyStreamVolumes(duplicatedOutput, device, 0, true);
+                    } else {
+                        ALOGW("checkOutputsForDevice() could not open dup output for %d and %d",
+                                mPrimaryOutput, output);
+                        mpClientInterface->closeOutput(output);
+                        mOutputs.removeItem(output);
+                        output = 0;
+                    }
+                }
+            }
+            if (output == 0) {
+                ALOGW("checkOutputsForDevice() could not open output for device %x", device);
+                delete desc;
+                profiles.removeAt(profile_index);
+                profile_index--;
+            } else {
+                outputs.add(output);
+                ALOGV("checkOutputsForDevice(): adding output %d", output);
+            }
+        }
+
+        if (profiles.isEmpty()) {
+            ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
+            return BAD_VALUE;
+        }
+    } else {
+        // check if one opened output is not needed any more after disconnecting one device
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            desc = mOutputs.valueAt(i);
+            if (!desc->isDuplicated() &&
+                    !(desc->mProfile->mSupportedDevices & mAvailableOutputDevices)) {
+                ALOGV("checkOutputsForDevice(): disconnecting adding output %d", mOutputs.keyAt(i));
+                outputs.add(mOutputs.keyAt(i));
+            }
+        }
+        for (size_t i = 0; i < mHwModules.size(); i++)
+        {
+            if (mHwModules[i]->mHandle == 0) {
+                continue;
+            }
+            for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+            {
+                IOProfile *profile = mHwModules[i]->mOutputProfiles[j];
+                if ((profile->mSupportedDevices & device) &&
+                        (profile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
+                    ALOGV("checkOutputsForDevice(): clearing direct output profile %d on module %d",
+                          j, i);
+                    if (profile->mSamplingRates[0] == 0) {
+                        profile->mSamplingRates.clear();
+                        profile->mSamplingRates.add(0);
+                    }
+                    if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
+                        profile->mFormats.clear();
+                        profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
+                    }
+                    if (profile->mChannelMasks[0] == 0) {
+                        profile->mChannelMasks.clear();
+                        profile->mChannelMasks.add(0);
+                    }
+                }
+            }
+        }
+    }
+    return NO_ERROR;
+}
+
+void AudioPolicyManager::closeOutput(audio_io_handle_t output)
+{
+    ALOGV("closeOutput(%d)", output);
+
+    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+    if (outputDesc == NULL) {
+        ALOGW("closeOutput() unknown output %d", output);
+        return;
+    }
+
+    // look for duplicated outputs connected to the output being removed.
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        AudioOutputDescriptor *dupOutputDesc = mOutputs.valueAt(i);
+        if (dupOutputDesc->isDuplicated() &&
+                (dupOutputDesc->mOutput1 == outputDesc ||
+                dupOutputDesc->mOutput2 == outputDesc)) {
+            AudioOutputDescriptor *outputDesc2;
+            if (dupOutputDesc->mOutput1 == outputDesc) {
+                outputDesc2 = dupOutputDesc->mOutput2;
+            } else {
+                outputDesc2 = dupOutputDesc->mOutput1;
+            }
+            // As all active tracks on duplicated output will be deleted,
+            // and as they were also referenced on the other output, the reference
+            // count for their stream type must be adjusted accordingly on
+            // the other output.
+            for (int j = 0; j < AUDIO_STREAM_CNT; j++) {
+                int refCount = dupOutputDesc->mRefCount[j];
+                outputDesc2->changeRefCount((audio_stream_type_t)j,-refCount);
+            }
+            audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i);
+            ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput);
+
+            mpClientInterface->closeOutput(duplicatedOutput);
+            delete mOutputs.valueFor(duplicatedOutput);
+            mOutputs.removeItem(duplicatedOutput);
+        }
+    }
+
+    AudioParameter param;
+    param.add(String8("closing"), String8("true"));
+    mpClientInterface->setParameters(output, param.toString());
+
+    mpClientInterface->closeOutput(output);
+    delete outputDesc;
+    mOutputs.removeItem(output);
+    mPreviousOutputs = mOutputs;
+}
+
+SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(audio_devices_t device,
+                        DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> openOutputs)
+{
+    SortedVector<audio_io_handle_t> outputs;
+
+    ALOGVV("getOutputsForDevice() device %04x", device);
+    for (size_t i = 0; i < openOutputs.size(); i++) {
+        ALOGVV("output %d isDuplicated=%d device=%04x",
+                i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices());
+        if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) {
+            ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i));
+            outputs.add(openOutputs.keyAt(i));
+        }
+    }
+    return outputs;
+}
+
+bool AudioPolicyManager::vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
+                                   SortedVector<audio_io_handle_t>& outputs2)
+{
+    if (outputs1.size() != outputs2.size()) {
+        return false;
+    }
+    for (size_t i = 0; i < outputs1.size(); i++) {
+        if (outputs1[i] != outputs2[i]) {
+            return false;
+        }
+    }
+    return true;
+}
+
+void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy)
+{
+    audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
+    audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
+    SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs);
+    SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
+
+    if (!vectorsEqual(srcOutputs,dstOutputs)) {
+        ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d",
+              strategy, srcOutputs[0], dstOutputs[0]);
+        // mute strategy while moving tracks from one output to another
+        for (size_t i = 0; i < srcOutputs.size(); i++) {
+            AudioOutputDescriptor *desc = mOutputs.valueFor(srcOutputs[i]);
+            if (desc->isStrategyActive(strategy)) {
+                setStrategyMute(strategy, true, srcOutputs[i]);
+                setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice);
+            }
+        }
+
+        // Move effects associated to this strategy from previous output to new output
+        if (strategy == STRATEGY_MEDIA) {
+            audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs);
+            SortedVector<audio_io_handle_t> moved;
+            for (size_t i = 0; i < mEffects.size(); i++) {
+                EffectDescriptor *desc = mEffects.valueAt(i);
+                if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX &&
+                        desc->mIo != fxOutput) {
+                    if (moved.indexOf(desc->mIo) < 0) {
+                        ALOGV("checkOutputForStrategy() moving effect %d to output %d",
+                              mEffects.keyAt(i), fxOutput);
+                        mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, desc->mIo,
+                                                       fxOutput);
+                        moved.add(desc->mIo);
+                    }
+                    desc->mIo = fxOutput;
+                }
+            }
+        }
+        // Move tracks associated to this strategy from previous output to new output
+        for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+            if (getStrategy((audio_stream_type_t)i) == strategy) {
+                mpClientInterface->invalidateStream((audio_stream_type_t)i);
+            }
+        }
+    }
+}
+
+void AudioPolicyManager::checkOutputForAllStrategies()
+{
+    checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
+    checkOutputForStrategy(STRATEGY_PHONE);
+    checkOutputForStrategy(STRATEGY_SONIFICATION);
+    checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
+    checkOutputForStrategy(STRATEGY_MEDIA);
+    checkOutputForStrategy(STRATEGY_DTMF);
+}
+
+audio_io_handle_t AudioPolicyManager::getA2dpOutput()
+{
+    if (!mHasA2dp) {
+        return 0;
+    }
+
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
+        if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) {
+            return mOutputs.keyAt(i);
+        }
+    }
+
+    return 0;
+}
+
+void AudioPolicyManager::checkA2dpSuspend()
+{
+    if (!mHasA2dp) {
+        return;
+    }
+    audio_io_handle_t a2dpOutput = getA2dpOutput();
+    if (a2dpOutput == 0) {
+        return;
+    }
+
+    // suspend A2DP output if:
+    //      (NOT already suspended) &&
+    //      ((SCO device is connected &&
+    //       (forced usage for communication || for record is SCO))) ||
+    //      (phone state is ringing || in call)
+    //
+    // restore A2DP output if:
+    //      (Already suspended) &&
+    //      ((SCO device is NOT connected ||
+    //       (forced usage NOT for communication && NOT for record is SCO))) &&
+    //      (phone state is NOT ringing && NOT in call)
+    //
+    if (mA2dpSuspended) {
+        if (((mScoDeviceAddress == "") ||
+             ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO) &&
+              (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] != AUDIO_POLICY_FORCE_BT_SCO))) &&
+             ((mPhoneState != AUDIO_MODE_IN_CALL) &&
+              (mPhoneState != AUDIO_MODE_RINGTONE))) {
+
+            mpClientInterface->restoreOutput(a2dpOutput);
+            mA2dpSuspended = false;
+        }
+    } else {
+        if (((mScoDeviceAddress != "") &&
+             ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) ||
+              (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO))) ||
+             ((mPhoneState == AUDIO_MODE_IN_CALL) ||
+              (mPhoneState == AUDIO_MODE_RINGTONE))) {
+
+            mpClientInterface->suspendOutput(a2dpOutput);
+            mA2dpSuspended = true;
+        }
+    }
+}
+
+audio_devices_t AudioPolicyManager::getNewDevice(audio_io_handle_t output, bool fromCache)
+{
+    audio_devices_t device = AUDIO_DEVICE_NONE;
+
+    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+    // check the following by order of priority to request a routing change if necessary:
+    // 1: the strategy enforced audible is active on the output:
+    //      use device for strategy enforced audible
+    // 2: we are in call or the strategy phone is active on the output:
+    //      use device for strategy phone
+    // 3: the strategy sonification is active on the output:
+    //      use device for strategy sonification
+    // 4: the strategy "respectful" sonification is active on the output:
+    //      use device for strategy "respectful" sonification
+    // 5: the strategy media is active on the output:
+    //      use device for strategy media
+    // 6: the strategy DTMF is active on the output:
+    //      use device for strategy DTMF
+    if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) {
+        device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
+    } else if (isInCall() ||
+                    outputDesc->isStrategyActive(STRATEGY_PHONE)) {
+        device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
+    } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)) {
+        device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
+    } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) {
+        device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
+    } else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) {
+        device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
+    } else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) {
+        device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
+    }
+
+    ALOGV("getNewDevice() selected device %x", device);
+    return device;
+}
+
+uint32_t AudioPolicyManager::getStrategyForStream(audio_stream_type_t stream) {
+    return (uint32_t)getStrategy(stream);
+}
+
+audio_devices_t AudioPolicyManager::getDevicesForStream(audio_stream_type_t stream) {
+    audio_devices_t devices;
+    // By checking the range of stream before calling getStrategy, we avoid
+    // getStrategy's behavior for invalid streams.  getStrategy would do a ALOGE
+    // and then return STRATEGY_MEDIA, but we want to return the empty set.
+    if (stream < (audio_stream_type_t) 0 || stream >= AUDIO_STREAM_CNT) {
+        devices = AUDIO_DEVICE_NONE;
+    } else {
+        AudioPolicyManager::routing_strategy strategy = getStrategy(stream);
+        devices = getDeviceForStrategy(strategy, true /*fromCache*/);
+    }
+    return devices;
+}
+
+AudioPolicyManager::routing_strategy AudioPolicyManager::getStrategy(
+        audio_stream_type_t stream) {
+    // stream to strategy mapping
+    switch (stream) {
+    case AUDIO_STREAM_VOICE_CALL:
+    case AUDIO_STREAM_BLUETOOTH_SCO:
+        return STRATEGY_PHONE;
+    case AUDIO_STREAM_RING:
+    case AUDIO_STREAM_ALARM:
+        return STRATEGY_SONIFICATION;
+    case AUDIO_STREAM_NOTIFICATION:
+        return STRATEGY_SONIFICATION_RESPECTFUL;
+    case AUDIO_STREAM_DTMF:
+        return STRATEGY_DTMF;
+    default:
+        ALOGE("unknown stream type");
+    case AUDIO_STREAM_SYSTEM:
+        // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
+        // while key clicks are played produces a poor result
+    case AUDIO_STREAM_TTS:
+    case AUDIO_STREAM_MUSIC:
+        return STRATEGY_MEDIA;
+    case AUDIO_STREAM_ENFORCED_AUDIBLE:
+        return STRATEGY_ENFORCED_AUDIBLE;
+    }
+}
+
+void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) {
+    switch(stream) {
+    case AUDIO_STREAM_MUSIC:
+        checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
+        updateDevicesAndOutputs();
+        break;
+    default:
+        break;
+    }
+}
+
+audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy,
+                                                             bool fromCache)
+{
+    uint32_t device = AUDIO_DEVICE_NONE;
+
+    if (fromCache) {
+        ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x",
+              strategy, mDeviceForStrategy[strategy]);
+        return mDeviceForStrategy[strategy];
+    }
+
+    switch (strategy) {
+
+    case STRATEGY_SONIFICATION_RESPECTFUL:
+        if (isInCall()) {
+            device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+        } else if (isStreamActiveRemotely(AUDIO_STREAM_MUSIC,
+                SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
+            // while media is playing on a remote device, use the the sonification behavior.
+            // Note that we test this usecase before testing if media is playing because
+            //   the isStreamActive() method only informs about the activity of a stream, not
+            //   if it's for local playback. Note also that we use the same delay between both tests
+            device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+        } else if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
+            // while media is playing (or has recently played), use the same device
+            device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+        } else {
+            // when media is not playing anymore, fall back on the sonification behavior
+            device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+        }
+
+        break;
+
+    case STRATEGY_DTMF:
+        if (!isInCall()) {
+            // when off call, DTMF strategy follows the same rules as MEDIA strategy
+            device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+            break;
+        }
+        // when in call, DTMF and PHONE strategies follow the same rules
+        // FALL THROUGH
+
+    case STRATEGY_PHONE:
+        // for phone strategy, we first consider the forced use and then the available devices by order
+        // of priority
+        switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) {
+        case AUDIO_POLICY_FORCE_BT_SCO:
+            if (!isInCall() || strategy != STRATEGY_DTMF) {
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
+                if (device) break;
+            }
+            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
+            if (device) break;
+            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
+            if (device) break;
+            // if SCO device is requested but no SCO device is available, fall back to default case
+            // FALL THROUGH
+
+        default:    // FORCE_NONE
+            // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
+            if (mHasA2dp && !isInCall() &&
+                    (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
+                    (getA2dpOutput() != 0) && !mA2dpSuspended) {
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+                if (device) break;
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+                if (device) break;
+            }
+            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+            if (device) break;
+            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
+            if (device) break;
+            if (mPhoneState != AUDIO_MODE_IN_CALL) {
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+                if (device) break;
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
+                if (device) break;
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+                if (device) break;
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+                if (device) break;
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+                if (device) break;
+            }
+            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_EARPIECE;
+            if (device) break;
+            device = mDefaultOutputDevice;
+            if (device == AUDIO_DEVICE_NONE) {
+                ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE");
+            }
+            break;
+
+        case AUDIO_POLICY_FORCE_SPEAKER:
+            // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
+            // A2DP speaker when forcing to speaker output
+            if (mHasA2dp && !isInCall() &&
+                    (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
+                    (getA2dpOutput() != 0) && !mA2dpSuspended) {
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+                if (device) break;
+            }
+            if (mPhoneState != AUDIO_MODE_IN_CALL) {
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+                if (device) break;
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
+                if (device) break;
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+                if (device) break;
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+                if (device) break;
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+                if (device) break;
+            }
+            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
+            if (device) break;
+            device = mDefaultOutputDevice;
+            if (device == AUDIO_DEVICE_NONE) {
+                ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER");
+            }
+            break;
+        }
+    break;
+
+    case STRATEGY_SONIFICATION:
+
+        // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
+        // handleIncallSonification().
+        if (isInCall()) {
+            device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/);
+            break;
+        }
+        // FALL THROUGH
+
+    case STRATEGY_ENFORCED_AUDIBLE:
+        // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION
+        // except:
+        //   - when in call where it doesn't default to STRATEGY_PHONE behavior
+        //   - in countries where not enforced in which case it follows STRATEGY_MEDIA
+
+        if ((strategy == STRATEGY_SONIFICATION) ||
+                (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)) {
+            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
+            if (device == AUDIO_DEVICE_NONE) {
+                ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION");
+            }
+        }
+        // The second device used for sonification is the same as the device used by media strategy
+        // FALL THROUGH
+
+    case STRATEGY_MEDIA: {
+        uint32_t device2 = AUDIO_DEVICE_NONE;
+        if (strategy != STRATEGY_SONIFICATION) {
+            // no sonification on remote submix (e.g. WFD)
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+        }
+        if ((device2 == AUDIO_DEVICE_NONE) &&
+                mHasA2dp &&
+                (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
+                (getA2dpOutput() != 0) && !mA2dpSuspended) {
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+            if (device2 == AUDIO_DEVICE_NONE) {
+                device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+            }
+            if (device2 == AUDIO_DEVICE_NONE) {
+                device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+            }
+        }
+        if (device2 == AUDIO_DEVICE_NONE) {
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+        }
+        if (device2 == AUDIO_DEVICE_NONE) {
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
+        }
+        if (device2 == AUDIO_DEVICE_NONE) {
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+        }
+        if (device2 == AUDIO_DEVICE_NONE) {
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
+        }
+        if (device2 == AUDIO_DEVICE_NONE) {
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+        }
+        if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
+            // no sonification on aux digital (e.g. HDMI)
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+        }
+        if ((device2 == AUDIO_DEVICE_NONE) &&
+                (mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK] == AUDIO_POLICY_FORCE_ANALOG_DOCK)) {
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+        }
+        if (device2 == AUDIO_DEVICE_NONE) {
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
+        }
+
+        // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
+        // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
+        device |= device2;
+        if (device) break;
+        device = mDefaultOutputDevice;
+        if (device == AUDIO_DEVICE_NONE) {
+            ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA");
+        }
+        } break;
+
+    default:
+        ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
+        break;
+    }
+
+    ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
+    return device;
+}
+
+void AudioPolicyManager::updateDevicesAndOutputs()
+{
+    for (int i = 0; i < NUM_STRATEGIES; i++) {
+        mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
+    }
+    mPreviousOutputs = mOutputs;
+}
+
+uint32_t AudioPolicyManager::checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc,
+                                                       audio_devices_t prevDevice,
+                                                       uint32_t delayMs)
+{
+    // mute/unmute strategies using an incompatible device combination
+    // if muting, wait for the audio in pcm buffer to be drained before proceeding
+    // if unmuting, unmute only after the specified delay
+    if (outputDesc->isDuplicated()) {
+        return 0;
+    }
+
+    uint32_t muteWaitMs = 0;
+    audio_devices_t device = outputDesc->device();
+    bool shouldMute = outputDesc->isActive() && (popcount(device) >= 2);
+    // temporary mute output if device selection changes to avoid volume bursts due to
+    // different per device volumes
+    bool tempMute = outputDesc->isActive() && (device != prevDevice);
+
+    for (size_t i = 0; i < NUM_STRATEGIES; i++) {
+        audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
+        bool mute = shouldMute && (curDevice & device) && (curDevice != device);
+        bool doMute = false;
+
+        if (mute && !outputDesc->mStrategyMutedByDevice[i]) {
+            doMute = true;
+            outputDesc->mStrategyMutedByDevice[i] = true;
+        } else if (!mute && outputDesc->mStrategyMutedByDevice[i]){
+            doMute = true;
+            outputDesc->mStrategyMutedByDevice[i] = false;
+        }
+        if (doMute || tempMute) {
+            for (size_t j = 0; j < mOutputs.size(); j++) {
+                AudioOutputDescriptor *desc = mOutputs.valueAt(j);
+                // skip output if it does not share any device with current output
+                if ((desc->supportedDevices() & outputDesc->supportedDevices())
+                        == AUDIO_DEVICE_NONE) {
+                    continue;
+                }
+                audio_io_handle_t curOutput = mOutputs.keyAt(j);
+                ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d",
+                      mute ? "muting" : "unmuting", i, curDevice, curOutput);
+                setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs);
+                if (desc->isStrategyActive((routing_strategy)i)) {
+                    // do tempMute only for current output
+                    if (tempMute && (desc == outputDesc)) {
+                        setStrategyMute((routing_strategy)i, true, curOutput);
+                        setStrategyMute((routing_strategy)i, false, curOutput,
+                                            desc->latency() * 2, device);
+                    }
+                    if ((tempMute && (desc == outputDesc)) || mute) {
+                        if (muteWaitMs < desc->latency()) {
+                            muteWaitMs = desc->latency();
+                        }
+                    }
+                }
+            }
+        }
+    }
+
+    // FIXME: should not need to double latency if volume could be applied immediately by the
+    // audioflinger mixer. We must account for the delay between now and the next time
+    // the audioflinger thread for this output will process a buffer (which corresponds to
+    // one buffer size, usually 1/2 or 1/4 of the latency).
+    muteWaitMs *= 2;
+    // wait for the PCM output buffers to empty before proceeding with the rest of the command
+    if (muteWaitMs > delayMs) {
+        muteWaitMs -= delayMs;
+        usleep(muteWaitMs * 1000);
+        return muteWaitMs;
+    }
+    return 0;
+}
+
+uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output,
+                                             audio_devices_t device,
+                                             bool force,
+                                             int delayMs)
+{
+    ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs);
+    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+    AudioParameter param;
+    uint32_t muteWaitMs;
+
+    if (outputDesc->isDuplicated()) {
+        muteWaitMs = setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs);
+        muteWaitMs += setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs);
+        return muteWaitMs;
+    }
+    // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
+    // output profile
+    if ((device != AUDIO_DEVICE_NONE) &&
+            ((device & outputDesc->mProfile->mSupportedDevices) == 0)) {
+        return 0;
+    }
+
+    // filter devices according to output selected
+    device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices);
+
+    audio_devices_t prevDevice = outputDesc->mDevice;
+
+    ALOGV("setOutputDevice() prevDevice %04x", prevDevice);
+
+    if (device != AUDIO_DEVICE_NONE) {
+        outputDesc->mDevice = device;
+    }
+    muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs);
+
+    // Do not change the routing if:
+    //  - the requested device is AUDIO_DEVICE_NONE
+    //  - the requested device is the same as current device and force is not specified.
+    // Doing this check here allows the caller to call setOutputDevice() without conditions
+    if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force) {
+        ALOGV("setOutputDevice() setting same device %04x or null device for output %d", device, output);
+        return muteWaitMs;
+    }
+
+    ALOGV("setOutputDevice() changing device");
+    // do the routing
+    param.addInt(String8(AudioParameter::keyRouting), (int)device);
+    mpClientInterface->setParameters(output, param.toString(), delayMs);
+
+    // update stream volumes according to new device
+    applyStreamVolumes(output, device, delayMs);
+
+    return muteWaitMs;
+}
+
+AudioPolicyManager::IOProfile *AudioPolicyManager::getInputProfile(audio_devices_t device,
+                                                   uint32_t samplingRate,
+                                                   audio_format_t format,
+                                                   audio_channel_mask_t channelMask)
+{
+    // Choose an input profile based on the requested capture parameters: select the first available
+    // profile supporting all requested parameters.
+
+    for (size_t i = 0; i < mHwModules.size(); i++)
+    {
+        if (mHwModules[i]->mHandle == 0) {
+            continue;
+        }
+        for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
+        {
+            IOProfile *profile = mHwModules[i]->mInputProfiles[j];
+            if (profile->isCompatibleProfile(device, samplingRate, format,
+                                             channelMask, AUDIO_OUTPUT_FLAG_NONE)) {
+                return profile;
+            }
+        }
+    }
+    return NULL;
+}
+
+audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource)
+{
+    uint32_t device = AUDIO_DEVICE_NONE;
+
+    switch (inputSource) {
+    case AUDIO_SOURCE_VOICE_UPLINK:
+      if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) {
+          device = AUDIO_DEVICE_IN_VOICE_CALL;
+          break;
+      }
+      // FALL THROUGH
+
+    case AUDIO_SOURCE_DEFAULT:
+    case AUDIO_SOURCE_MIC:
+    case AUDIO_SOURCE_VOICE_RECOGNITION:
+    case AUDIO_SOURCE_HOTWORD:
+    case AUDIO_SOURCE_VOICE_COMMUNICATION:
+        if (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO &&
+            mAvailableInputDevices & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
+            device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+        } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+            device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+        } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+            device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+        }
+        break;
+    case AUDIO_SOURCE_CAMCORDER:
+        if (mAvailableInputDevices & AUDIO_DEVICE_IN_BACK_MIC) {
+            device = AUDIO_DEVICE_IN_BACK_MIC;
+        } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+            device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+        }
+        break;
+    case AUDIO_SOURCE_VOICE_DOWNLINK:
+    case AUDIO_SOURCE_VOICE_CALL:
+        if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) {
+            device = AUDIO_DEVICE_IN_VOICE_CALL;
+        }
+        break;
+    case AUDIO_SOURCE_REMOTE_SUBMIX:
+        if (mAvailableInputDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
+            device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+        }
+        break;
+    default:
+        ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
+        break;
+    }
+    ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
+    return device;
+}
+
+bool AudioPolicyManager::isVirtualInputDevice(audio_devices_t device)
+{
+    if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
+        device &= ~AUDIO_DEVICE_BIT_IN;
+        if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0))
+            return true;
+    }
+    return false;
+}
+
+audio_io_handle_t AudioPolicyManager::getActiveInput(bool ignoreVirtualInputs)
+{
+    for (size_t i = 0; i < mInputs.size(); i++) {
+        const AudioInputDescriptor * input_descriptor = mInputs.valueAt(i);
+        if ((input_descriptor->mRefCount > 0)
+                && (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) {
+            return mInputs.keyAt(i);
+        }
+    }
+    return 0;
+}
+
+
+audio_devices_t AudioPolicyManager::getDeviceForVolume(audio_devices_t device)
+{
+    if (device == AUDIO_DEVICE_NONE) {
+        // this happens when forcing a route update and no track is active on an output.
+        // In this case the returned category is not important.
+        device =  AUDIO_DEVICE_OUT_SPEAKER;
+    } else if (popcount(device) > 1) {
+        // Multiple device selection is either:
+        //  - speaker + one other device: give priority to speaker in this case.
+        //  - one A2DP device + another device: happens with duplicated output. In this case
+        // retain the device on the A2DP output as the other must not correspond to an active
+        // selection if not the speaker.
+        if (device & AUDIO_DEVICE_OUT_SPEAKER) {
+            device = AUDIO_DEVICE_OUT_SPEAKER;
+        } else {
+            device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
+        }
+    }
+
+    ALOGW_IF(popcount(device) != 1,
+            "getDeviceForVolume() invalid device combination: %08x",
+            device);
+
+    return device;
+}
+
+AudioPolicyManager::device_category AudioPolicyManager::getDeviceCategory(audio_devices_t device)
+{
+    switch(getDeviceForVolume(device)) {
+        case AUDIO_DEVICE_OUT_EARPIECE:
+            return DEVICE_CATEGORY_EARPIECE;
+        case AUDIO_DEVICE_OUT_WIRED_HEADSET:
+        case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
+            return DEVICE_CATEGORY_HEADSET;
+        case AUDIO_DEVICE_OUT_SPEAKER:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
+        case AUDIO_DEVICE_OUT_AUX_DIGITAL:
+        case AUDIO_DEVICE_OUT_USB_ACCESSORY:
+        case AUDIO_DEVICE_OUT_USB_DEVICE:
+        case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
+        default:
+            return DEVICE_CATEGORY_SPEAKER;
+    }
+}
+
+float AudioPolicyManager::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+        int indexInUi)
+{
+    device_category deviceCategory = getDeviceCategory(device);
+    const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory];
+
+    // the volume index in the UI is relative to the min and max volume indices for this stream type
+    int nbSteps = 1 + curve[VOLMAX].mIndex -
+            curve[VOLMIN].mIndex;
+    int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) /
+            (streamDesc.mIndexMax - streamDesc.mIndexMin);
+
+    // find what part of the curve this index volume belongs to, or if it's out of bounds
+    int segment = 0;
+    if (volIdx < curve[VOLMIN].mIndex) {         // out of bounds
+        return 0.0f;
+    } else if (volIdx < curve[VOLKNEE1].mIndex) {
+        segment = 0;
+    } else if (volIdx < curve[VOLKNEE2].mIndex) {
+        segment = 1;
+    } else if (volIdx <= curve[VOLMAX].mIndex) {
+        segment = 2;
+    } else {                                                               // out of bounds
+        return 1.0f;
+    }
+
+    // linear interpolation in the attenuation table in dB
+    float decibels = curve[segment].mDBAttenuation +
+            ((float)(volIdx - curve[segment].mIndex)) *
+                ( (curve[segment+1].mDBAttenuation -
+                        curve[segment].mDBAttenuation) /
+                    ((float)(curve[segment+1].mIndex -
+                            curve[segment].mIndex)) );
+
+    float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
+
+    ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
+            curve[segment].mIndex, volIdx,
+            curve[segment+1].mIndex,
+            curve[segment].mDBAttenuation,
+            decibels,
+            curve[segment+1].mDBAttenuation,
+            amplification);
+
+    return amplification;
+}
+
+const AudioPolicyManager::VolumeCurvePoint
+    AudioPolicyManager::sDefaultVolumeCurve[AudioPolicyManager::VOLCNT] = {
+    {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
+};
+
+const AudioPolicyManager::VolumeCurvePoint
+    AudioPolicyManager::sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT] = {
+    {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
+};
+
+const AudioPolicyManager::VolumeCurvePoint
+    AudioPolicyManager::sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT] = {
+    {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
+};
+
+const AudioPolicyManager::VolumeCurvePoint
+    AudioPolicyManager::sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT] = {
+    {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
+};
+
+const AudioPolicyManager::VolumeCurvePoint
+    AudioPolicyManager::sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
+    {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f}
+};
+
+// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks
+// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets.
+// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java).
+// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset.
+
+const AudioPolicyManager::VolumeCurvePoint
+    AudioPolicyManager::sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
+    {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
+};
+
+const AudioPolicyManager::VolumeCurvePoint
+    AudioPolicyManager::sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
+    {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f}
+};
+
+const AudioPolicyManager::VolumeCurvePoint
+    AudioPolicyManager::sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
+    {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
+};
+
+const AudioPolicyManager::VolumeCurvePoint
+    AudioPolicyManager::sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = {
+    {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
+};
+
+const AudioPolicyManager::VolumeCurvePoint
+    AudioPolicyManager::sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = {
+    {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
+};
+
+const AudioPolicyManager::VolumeCurvePoint
+            *AudioPolicyManager::sVolumeProfiles[AUDIO_STREAM_CNT]
+                                                   [AudioPolicyManager::DEVICE_CATEGORY_CNT] = {
+    { // AUDIO_STREAM_VOICE_CALL
+        sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultVoiceVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+    { // AUDIO_STREAM_SYSTEM
+        sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultSystemVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+    { // AUDIO_STREAM_RING
+        sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+    { // AUDIO_STREAM_MUSIC
+        sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+    { // AUDIO_STREAM_ALARM
+        sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+    { // AUDIO_STREAM_NOTIFICATION
+        sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+    { // AUDIO_STREAM_BLUETOOTH_SCO
+        sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultVoiceVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+    { // AUDIO_STREAM_ENFORCED_AUDIBLE
+        sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultSystemVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+    {  // AUDIO_STREAM_DTMF
+        sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultSystemVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+    { // AUDIO_STREAM_TTS
+        sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+};
+
+void AudioPolicyManager::initializeVolumeCurves()
+{
+    for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+        for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+            mStreams[i].mVolumeCurve[j] =
+                    sVolumeProfiles[i][j];
+        }
+    }
+
+    // Check availability of DRC on speaker path: if available, override some of the speaker curves
+    if (mSpeakerDrcEnabled) {
+        mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
+                sDefaultSystemVolumeCurveDrc;
+        mStreams[AUDIO_STREAM_RING].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
+                sSpeakerSonificationVolumeCurveDrc;
+        mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
+                sSpeakerSonificationVolumeCurveDrc;
+        mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
+                sSpeakerSonificationVolumeCurveDrc;
+    }
+}
+
+float AudioPolicyManager::computeVolume(audio_stream_type_t stream,
+                                            int index,
+                                            audio_io_handle_t output,
+                                            audio_devices_t device)
+{
+    float volume = 1.0;
+    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+    StreamDescriptor &streamDesc = mStreams[stream];
+
+    if (device == AUDIO_DEVICE_NONE) {
+        device = outputDesc->device();
+    }
+
+    // if volume is not 0 (not muted), force media volume to max on digital output
+    if (stream == AUDIO_STREAM_MUSIC &&
+        index != mStreams[stream].mIndexMin &&
+        (device == AUDIO_DEVICE_OUT_AUX_DIGITAL ||
+         device == AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET ||
+         device == AUDIO_DEVICE_OUT_USB_ACCESSORY ||
+         device == AUDIO_DEVICE_OUT_USB_DEVICE)) {
+        return 1.0;
+    }
+
+    volume = volIndexToAmpl(device, streamDesc, index);
+
+    // if a headset is connected, apply the following rules to ring tones and notifications
+    // to avoid sound level bursts in user's ears:
+    // - always attenuate ring tones and notifications volume by 6dB
+    // - if music is playing, always limit the volume to current music volume,
+    // with a minimum threshold at -36dB so that notification is always perceived.
+    const routing_strategy stream_strategy = getStrategy(stream);
+    if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP |
+            AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
+            AUDIO_DEVICE_OUT_WIRED_HEADSET |
+            AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) &&
+        ((stream_strategy == STRATEGY_SONIFICATION)
+                || (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL)
+                || (stream == AUDIO_STREAM_SYSTEM)
+                || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) &&
+                    (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) &&
+        streamDesc.mCanBeMuted) {
+        volume *= SONIFICATION_HEADSET_VOLUME_FACTOR;
+        // when the phone is ringing we must consider that music could have been paused just before
+        // by the music application and behave as if music was active if the last music track was
+        // just stopped
+        if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) ||
+                mLimitRingtoneVolume) {
+            audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/);
+            float musicVol = computeVolume(AUDIO_STREAM_MUSIC,
+                               mStreams[AUDIO_STREAM_MUSIC].getVolumeIndex(musicDevice),
+                               output,
+                               musicDevice);
+            float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ?
+                                musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
+            if (volume > minVol) {
+                volume = minVol;
+                ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
+            }
+        }
+    }
+
+    return volume;
+}
+
+status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream,
+                                                   int index,
+                                                   audio_io_handle_t output,
+                                                   audio_devices_t device,
+                                                   int delayMs,
+                                                   bool force)
+{
+
+    // do not change actual stream volume if the stream is muted
+    if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
+        ALOGVV("checkAndSetVolume() stream %d muted count %d",
+              stream, mOutputs.valueFor(output)->mMuteCount[stream]);
+        return NO_ERROR;
+    }
+
+    // do not change in call volume if bluetooth is connected and vice versa
+    if ((stream == AUDIO_STREAM_VOICE_CALL &&
+            mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) ||
+        (stream == AUDIO_STREAM_BLUETOOTH_SCO &&
+                mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO)) {
+        ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
+             stream, mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]);
+        return INVALID_OPERATION;
+    }
+
+    float volume = computeVolume(stream, index, output, device);
+    // We actually change the volume if:
+    // - the float value returned by computeVolume() changed
+    // - the force flag is set
+    if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
+            force) {
+        mOutputs.valueFor(output)->mCurVolume[stream] = volume;
+        ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
+        // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is
+        // enabled
+        if (stream == AUDIO_STREAM_BLUETOOTH_SCO) {
+            mpClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volume, output, delayMs);
+        }
+        mpClientInterface->setStreamVolume(stream, volume, output, delayMs);
+    }
+
+    if (stream == AUDIO_STREAM_VOICE_CALL ||
+        stream == AUDIO_STREAM_BLUETOOTH_SCO) {
+        float voiceVolume;
+        // Force voice volume to max for bluetooth SCO as volume is managed by the headset
+        if (stream == AUDIO_STREAM_VOICE_CALL) {
+            voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
+        } else {
+            voiceVolume = 1.0;
+        }
+
+        if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) {
+            mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
+            mLastVoiceVolume = voiceVolume;
+        }
+    }
+
+    return NO_ERROR;
+}
+
+void AudioPolicyManager::applyStreamVolumes(audio_io_handle_t output,
+                                                audio_devices_t device,
+                                                int delayMs,
+                                                bool force)
+{
+    ALOGVV("applyStreamVolumes() for output %d and device %x", output, device);
+
+    for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+        checkAndSetVolume((audio_stream_type_t)stream,
+                          mStreams[stream].getVolumeIndex(device),
+                          output,
+                          device,
+                          delayMs,
+                          force);
+    }
+}
+
+void AudioPolicyManager::setStrategyMute(routing_strategy strategy,
+                                             bool on,
+                                             audio_io_handle_t output,
+                                             int delayMs,
+                                             audio_devices_t device)
+{
+    ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
+    for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+        if (getStrategy((audio_stream_type_t)stream) == strategy) {
+            setStreamMute((audio_stream_type_t)stream, on, output, delayMs, device);
+        }
+    }
+}
+
+void AudioPolicyManager::setStreamMute(audio_stream_type_t stream,
+                                           bool on,
+                                           audio_io_handle_t output,
+                                           int delayMs,
+                                           audio_devices_t device)
+{
+    StreamDescriptor &streamDesc = mStreams[stream];
+    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+    if (device == AUDIO_DEVICE_NONE) {
+        device = outputDesc->device();
+    }
+
+    ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x",
+          stream, on, output, outputDesc->mMuteCount[stream], device);
+
+    if (on) {
+        if (outputDesc->mMuteCount[stream] == 0) {
+            if (streamDesc.mCanBeMuted &&
+                    ((stream != AUDIO_STREAM_ENFORCED_AUDIBLE) ||
+                     (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) {
+                checkAndSetVolume(stream, 0, output, device, delayMs);
+            }
+        }
+        // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
+        outputDesc->mMuteCount[stream]++;
+    } else {
+        if (outputDesc->mMuteCount[stream] == 0) {
+            ALOGV("setStreamMute() unmuting non muted stream!");
+            return;
+        }
+        if (--outputDesc->mMuteCount[stream] == 0) {
+            checkAndSetVolume(stream,
+                              streamDesc.getVolumeIndex(device),
+                              output,
+                              device,
+                              delayMs);
+        }
+    }
+}
+
+void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream,
+                                                      bool starting, bool stateChange)
+{
+    // if the stream pertains to sonification strategy and we are in call we must
+    // mute the stream if it is low visibility. If it is high visibility, we must play a tone
+    // in the device used for phone strategy and play the tone if the selected device does not
+    // interfere with the device used for phone strategy
+    // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
+    // many times as there are active tracks on the output
+    const routing_strategy stream_strategy = getStrategy(stream);
+    if ((stream_strategy == STRATEGY_SONIFICATION) ||
+            ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
+        AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput);
+        ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
+                stream, starting, outputDesc->mDevice, stateChange);
+        if (outputDesc->mRefCount[stream]) {
+            int muteCount = 1;
+            if (stateChange) {
+                muteCount = outputDesc->mRefCount[stream];
+            }
+            if (audio_is_low_visibility(stream)) {
+                ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
+                for (int i = 0; i < muteCount; i++) {
+                    setStreamMute(stream, starting, mPrimaryOutput);
+                }
+            } else {
+                ALOGV("handleIncallSonification() high visibility");
+                if (outputDesc->device() &
+                        getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) {
+                    ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
+                    for (int i = 0; i < muteCount; i++) {
+                        setStreamMute(stream, starting, mPrimaryOutput);
+                    }
+                }
+                if (starting) {
+                    mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
+                                                 AUDIO_STREAM_VOICE_CALL);
+                } else {
+                    mpClientInterface->stopTone();
+                }
+            }
+        }
+    }
+}
+
+bool AudioPolicyManager::isInCall()
+{
+    return isStateInCall(mPhoneState);
+}
+
+bool AudioPolicyManager::isStateInCall(int state) {
+    return ((state == AUDIO_MODE_IN_CALL) ||
+            (state == AUDIO_MODE_IN_COMMUNICATION));
+}
+
+uint32_t AudioPolicyManager::getMaxEffectsCpuLoad()
+{
+    return MAX_EFFECTS_CPU_LOAD;
+}
+
+uint32_t AudioPolicyManager::getMaxEffectsMemory()
+{
+    return MAX_EFFECTS_MEMORY;
+}
+
+// --- AudioOutputDescriptor class implementation
+
+AudioPolicyManager::AudioOutputDescriptor::AudioOutputDescriptor(
+        const IOProfile *profile)
+    : mId(0), mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT),
+      mChannelMask(0), mLatency(0),
+    mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE),
+    mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
+{
+    // clear usage count for all stream types
+    for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+        mRefCount[i] = 0;
+        mCurVolume[i] = -1.0;
+        mMuteCount[i] = 0;
+        mStopTime[i] = 0;
+    }
+    for (int i = 0; i < NUM_STRATEGIES; i++) {
+        mStrategyMutedByDevice[i] = false;
+    }
+    if (profile != NULL) {
+        mSamplingRate = profile->mSamplingRates[0];
+        mFormat = profile->mFormats[0];
+        mChannelMask = profile->mChannelMasks[0];
+        mFlags = profile->mFlags;
+    }
+}
+
+audio_devices_t AudioPolicyManager::AudioOutputDescriptor::device() const
+{
+    if (isDuplicated()) {
+        return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
+    } else {
+        return mDevice;
+    }
+}
+
+uint32_t AudioPolicyManager::AudioOutputDescriptor::latency()
+{
+    if (isDuplicated()) {
+        return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
+    } else {
+        return mLatency;
+    }
+}
+
+bool AudioPolicyManager::AudioOutputDescriptor::sharesHwModuleWith(
+        const AudioOutputDescriptor *outputDesc)
+{
+    if (isDuplicated()) {
+        return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
+    } else if (outputDesc->isDuplicated()){
+        return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
+    } else {
+        return (mProfile->mModule == outputDesc->mProfile->mModule);
+    }
+}
+
+void AudioPolicyManager::AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream,
+                                                                   int delta)
+{
+    // forward usage count change to attached outputs
+    if (isDuplicated()) {
+        mOutput1->changeRefCount(stream, delta);
+        mOutput2->changeRefCount(stream, delta);
+    }
+    if ((delta + (int)mRefCount[stream]) < 0) {
+        ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d",
+              delta, stream, mRefCount[stream]);
+        mRefCount[stream] = 0;
+        return;
+    }
+    mRefCount[stream] += delta;
+    ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
+}
+
+audio_devices_t AudioPolicyManager::AudioOutputDescriptor::supportedDevices()
+{
+    if (isDuplicated()) {
+        return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
+    } else {
+        return mProfile->mSupportedDevices ;
+    }
+}
+
+bool AudioPolicyManager::AudioOutputDescriptor::isActive(uint32_t inPastMs) const
+{
+    return isStrategyActive(NUM_STRATEGIES, inPastMs);
+}
+
+bool AudioPolicyManager::AudioOutputDescriptor::isStrategyActive(routing_strategy strategy,
+                                                                       uint32_t inPastMs,
+                                                                       nsecs_t sysTime) const
+{
+    if ((sysTime == 0) && (inPastMs != 0)) {
+        sysTime = systemTime();
+    }
+    for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
+        if (((getStrategy((audio_stream_type_t)i) == strategy) ||
+                (NUM_STRATEGIES == strategy)) &&
+                isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) {
+            return true;
+        }
+    }
+    return false;
+}
+
+bool AudioPolicyManager::AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream,
+                                                                       uint32_t inPastMs,
+                                                                       nsecs_t sysTime) const
+{
+    if (mRefCount[stream] != 0) {
+        return true;
+    }
+    if (inPastMs == 0) {
+        return false;
+    }
+    if (sysTime == 0) {
+        sysTime = systemTime();
+    }
+    if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) {
+        return true;
+    }
+    return false;
+}
+
+
+status_t AudioPolicyManager::AudioOutputDescriptor::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Format: %08x\n", mFormat);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Devices %08x\n", device());
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
+    result.append(buffer);
+    for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
+        snprintf(buffer, SIZE, " %02d     %.03f     %02d       %02d\n",
+                 i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
+        result.append(buffer);
+    }
+    write(fd, result.string(), result.size());
+
+    return NO_ERROR;
+}
+
+// --- AudioInputDescriptor class implementation
+
+AudioPolicyManager::AudioInputDescriptor::AudioInputDescriptor(const IOProfile *profile)
+    : mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT), mChannelMask(0),
+      mDevice(AUDIO_DEVICE_NONE), mRefCount(0),
+      mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile)
+{
+}
+
+status_t AudioPolicyManager::AudioInputDescriptor::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Format: %d\n", mFormat);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+
+    return NO_ERROR;
+}
+
+// --- StreamDescriptor class implementation
+
+AudioPolicyManager::StreamDescriptor::StreamDescriptor()
+    :   mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
+{
+    mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
+}
+
+int AudioPolicyManager::StreamDescriptor::getVolumeIndex(audio_devices_t device)
+{
+    device = AudioPolicyManager::getDeviceForVolume(device);
+    // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT
+    if (mIndexCur.indexOfKey(device) < 0) {
+        device = AUDIO_DEVICE_OUT_DEFAULT;
+    }
+    return mIndexCur.valueFor(device);
+}
+
+void AudioPolicyManager::StreamDescriptor::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "%s         %02d         %02d         ",
+             mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax);
+    result.append(buffer);
+    for (size_t i = 0; i < mIndexCur.size(); i++) {
+        snprintf(buffer, SIZE, "%04x : %02d, ",
+                 mIndexCur.keyAt(i),
+                 mIndexCur.valueAt(i));
+        result.append(buffer);
+    }
+    result.append("\n");
+
+    write(fd, result.string(), result.size());
+}
+
+// --- EffectDescriptor class implementation
+
+status_t AudioPolicyManager::EffectDescriptor::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, " I/O: %d\n", mIo);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Session: %d\n", mSession);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Name: %s\n",  mDesc.name);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " %s\n",  mEnabled ? "Enabled" : "Disabled");
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+
+    return NO_ERROR;
+}
+
+// --- IOProfile class implementation
+
+AudioPolicyManager::HwModule::HwModule(const char *name)
+    : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)), mHandle(0)
+{
+}
+
+AudioPolicyManager::HwModule::~HwModule()
+{
+    for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+         delete mOutputProfiles[i];
+    }
+    for (size_t i = 0; i < mInputProfiles.size(); i++) {
+         delete mInputProfiles[i];
+    }
+    free((void *)mName);
+}
+
+void AudioPolicyManager::HwModule::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "  - name: %s\n", mName);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "  - handle: %d\n", mHandle);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+    if (mOutputProfiles.size()) {
+        write(fd, "  - outputs:\n", strlen("  - outputs:\n"));
+        for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+            snprintf(buffer, SIZE, "    output %d:\n", i);
+            write(fd, buffer, strlen(buffer));
+            mOutputProfiles[i]->dump(fd);
+        }
+    }
+    if (mInputProfiles.size()) {
+        write(fd, "  - inputs:\n", strlen("  - inputs:\n"));
+        for (size_t i = 0; i < mInputProfiles.size(); i++) {
+            snprintf(buffer, SIZE, "    input %d:\n", i);
+            write(fd, buffer, strlen(buffer));
+            mInputProfiles[i]->dump(fd);
+        }
+    }
+}
+
+AudioPolicyManager::IOProfile::IOProfile(HwModule *module)
+    : mFlags((audio_output_flags_t)0), mModule(module)
+{
+}
+
+AudioPolicyManager::IOProfile::~IOProfile()
+{
+}
+
+// checks if the IO profile is compatible with specified parameters.
+// Sampling rate, format and channel mask must be specified in order to
+// get a valid a match
+bool AudioPolicyManager::IOProfile::isCompatibleProfile(audio_devices_t device,
+                                                            uint32_t samplingRate,
+                                                            audio_format_t format,
+                                                            audio_channel_mask_t channelMask,
+                                                            audio_output_flags_t flags) const
+{
+    if (samplingRate == 0 || !audio_is_valid_format(format) || channelMask == 0) {
+         return false;
+     }
+
+     if ((mSupportedDevices & device) != device) {
+         return false;
+     }
+     if ((mFlags & flags) != flags) {
+         return false;
+     }
+     size_t i;
+     for (i = 0; i < mSamplingRates.size(); i++)
+     {
+         if (mSamplingRates[i] == samplingRate) {
+             break;
+         }
+     }
+     if (i == mSamplingRates.size()) {
+         return false;
+     }
+     for (i = 0; i < mFormats.size(); i++)
+     {
+         if (mFormats[i] == format) {
+             break;
+         }
+     }
+     if (i == mFormats.size()) {
+         return false;
+     }
+     for (i = 0; i < mChannelMasks.size(); i++)
+     {
+         if (mChannelMasks[i] == channelMask) {
+             break;
+         }
+     }
+     if (i == mChannelMasks.size()) {
+         return false;
+     }
+     return true;
+}
+
+void AudioPolicyManager::IOProfile::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "    - sampling rates: ");
+    result.append(buffer);
+    for (size_t i = 0; i < mSamplingRates.size(); i++) {
+        snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
+        result.append(buffer);
+        result.append(i == (mSamplingRates.size() - 1) ? "\n" : ", ");
+    }
+
+    snprintf(buffer, SIZE, "    - channel masks: ");
+    result.append(buffer);
+    for (size_t i = 0; i < mChannelMasks.size(); i++) {
+        snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
+        result.append(buffer);
+        result.append(i == (mChannelMasks.size() - 1) ? "\n" : ", ");
+    }
+
+    snprintf(buffer, SIZE, "    - formats: ");
+    result.append(buffer);
+    for (size_t i = 0; i < mFormats.size(); i++) {
+        snprintf(buffer, SIZE, "0x%08x", mFormats[i]);
+        result.append(buffer);
+        result.append(i == (mFormats.size() - 1) ? "\n" : ", ");
+    }
+
+    snprintf(buffer, SIZE, "    - devices: 0x%04x\n", mSupportedDevices);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "    - flags: 0x%04x\n", mFlags);
+    result.append(buffer);
+
+    write(fd, result.string(), result.size());
+}
+
+// --- audio_policy.conf file parsing
+
+struct StringToEnum {
+    const char *name;
+    uint32_t value;
+};
+
+#define STRING_TO_ENUM(string) { #string, string }
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+
+const struct StringToEnum sDeviceNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
+};
+
+const struct StringToEnum sFlagNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING),
+};
+
+const struct StringToEnum sFormatNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
+    STRING_TO_ENUM(AUDIO_FORMAT_MP3),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC),
+    STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
+};
+
+const struct StringToEnum sOutChannelsNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
+};
+
+const struct StringToEnum sInChannelsNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
+};
+
+
+uint32_t AudioPolicyManager::stringToEnum(const struct StringToEnum *table,
+                                              size_t size,
+                                              const char *name)
+{
+    for (size_t i = 0; i < size; i++) {
+        if (strcmp(table[i].name, name) == 0) {
+            ALOGV("stringToEnum() found %s", table[i].name);
+            return table[i].value;
+        }
+    }
+    return 0;
+}
+
+bool AudioPolicyManager::stringToBool(const char *value)
+{
+    return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0));
+}
+
+audio_output_flags_t AudioPolicyManager::parseFlagNames(char *name)
+{
+    uint32_t flag = 0;
+
+    // it is OK to cast name to non const here as we are not going to use it after
+    // strtok() modifies it
+    char *flagName = strtok(name, "|");
+    while (flagName != NULL) {
+        if (strlen(flagName) != 0) {
+            flag |= stringToEnum(sFlagNameToEnumTable,
+                               ARRAY_SIZE(sFlagNameToEnumTable),
+                               flagName);
+        }
+        flagName = strtok(NULL, "|");
+    }
+    //force direct flag if offload flag is set: offloading implies a direct output stream
+    // and all common behaviors are driven by checking only the direct flag
+    // this should normally be set appropriately in the policy configuration file
+    if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+        flag |= AUDIO_OUTPUT_FLAG_DIRECT;
+    }
+
+    return (audio_output_flags_t)flag;
+}
+
+audio_devices_t AudioPolicyManager::parseDeviceNames(char *name)
+{
+    uint32_t device = 0;
+
+    char *devName = strtok(name, "|");
+    while (devName != NULL) {
+        if (strlen(devName) != 0) {
+            device |= stringToEnum(sDeviceNameToEnumTable,
+                                 ARRAY_SIZE(sDeviceNameToEnumTable),
+                                 devName);
+        }
+        devName = strtok(NULL, "|");
+    }
+    return device;
+}
+
+void AudioPolicyManager::loadSamplingRates(char *name, IOProfile *profile)
+{
+    char *str = strtok(name, "|");
+
+    // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
+    // rates should be read from the output stream after it is opened for the first time
+    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+        profile->mSamplingRates.add(0);
+        return;
+    }
+
+    while (str != NULL) {
+        uint32_t rate = atoi(str);
+        if (rate != 0) {
+            ALOGV("loadSamplingRates() adding rate %d", rate);
+            profile->mSamplingRates.add(rate);
+        }
+        str = strtok(NULL, "|");
+    }
+    return;
+}
+
+void AudioPolicyManager::loadFormats(char *name, IOProfile *profile)
+{
+    char *str = strtok(name, "|");
+
+    // by convention, "0' in the first entry in mFormats indicates the supported formats
+    // should be read from the output stream after it is opened for the first time
+    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+        profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
+        return;
+    }
+
+    while (str != NULL) {
+        audio_format_t format = (audio_format_t)stringToEnum(sFormatNameToEnumTable,
+                                                             ARRAY_SIZE(sFormatNameToEnumTable),
+                                                             str);
+        if (format != AUDIO_FORMAT_DEFAULT) {
+            profile->mFormats.add(format);
+        }
+        str = strtok(NULL, "|");
+    }
+    return;
+}
+
+void AudioPolicyManager::loadInChannels(char *name, IOProfile *profile)
+{
+    const char *str = strtok(name, "|");
+
+    ALOGV("loadInChannels() %s", name);
+
+    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+        profile->mChannelMasks.add(0);
+        return;
+    }
+
+    while (str != NULL) {
+        audio_channel_mask_t channelMask =
+                (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
+                                                   ARRAY_SIZE(sInChannelsNameToEnumTable),
+                                                   str);
+        if (channelMask != 0) {
+            ALOGV("loadInChannels() adding channelMask %04x", channelMask);
+            profile->mChannelMasks.add(channelMask);
+        }
+        str = strtok(NULL, "|");
+    }
+    return;
+}
+
+void AudioPolicyManager::loadOutChannels(char *name, IOProfile *profile)
+{
+    const char *str = strtok(name, "|");
+
+    ALOGV("loadOutChannels() %s", name);
+
+    // by convention, "0' in the first entry in mChannelMasks indicates the supported channel
+    // masks should be read from the output stream after it is opened for the first time
+    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+        profile->mChannelMasks.add(0);
+        return;
+    }
+
+    while (str != NULL) {
+        audio_channel_mask_t channelMask =
+                (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
+                                                   ARRAY_SIZE(sOutChannelsNameToEnumTable),
+                                                   str);
+        if (channelMask != 0) {
+            profile->mChannelMasks.add(channelMask);
+        }
+        str = strtok(NULL, "|");
+    }
+    return;
+}
+
+status_t AudioPolicyManager::loadInput(cnode *root, HwModule *module)
+{
+    cnode *node = root->first_child;
+
+    IOProfile *profile = new IOProfile(module);
+
+    while (node) {
+        if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
+            loadSamplingRates((char *)node->value, profile);
+        } else if (strcmp(node->name, FORMATS_TAG) == 0) {
+            loadFormats((char *)node->value, profile);
+        } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+            loadInChannels((char *)node->value, profile);
+        } else if (strcmp(node->name, DEVICES_TAG) == 0) {
+            profile->mSupportedDevices = parseDeviceNames((char *)node->value);
+        }
+        node = node->next;
+    }
+    ALOGW_IF(profile->mSupportedDevices == AUDIO_DEVICE_NONE,
+            "loadInput() invalid supported devices");
+    ALOGW_IF(profile->mChannelMasks.size() == 0,
+            "loadInput() invalid supported channel masks");
+    ALOGW_IF(profile->mSamplingRates.size() == 0,
+            "loadInput() invalid supported sampling rates");
+    ALOGW_IF(profile->mFormats.size() == 0,
+            "loadInput() invalid supported formats");
+    if ((profile->mSupportedDevices != AUDIO_DEVICE_NONE) &&
+            (profile->mChannelMasks.size() != 0) &&
+            (profile->mSamplingRates.size() != 0) &&
+            (profile->mFormats.size() != 0)) {
+
+        ALOGV("loadInput() adding input mSupportedDevices %04x", profile->mSupportedDevices);
+
+        module->mInputProfiles.add(profile);
+        return NO_ERROR;
+    } else {
+        delete profile;
+        return BAD_VALUE;
+    }
+}
+
+status_t AudioPolicyManager::loadOutput(cnode *root, HwModule *module)
+{
+    cnode *node = root->first_child;
+
+    IOProfile *profile = new IOProfile(module);
+
+    while (node) {
+        if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
+            loadSamplingRates((char *)node->value, profile);
+        } else if (strcmp(node->name, FORMATS_TAG) == 0) {
+            loadFormats((char *)node->value, profile);
+        } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+            loadOutChannels((char *)node->value, profile);
+        } else if (strcmp(node->name, DEVICES_TAG) == 0) {
+            profile->mSupportedDevices = parseDeviceNames((char *)node->value);
+        } else if (strcmp(node->name, FLAGS_TAG) == 0) {
+            profile->mFlags = parseFlagNames((char *)node->value);
+        }
+        node = node->next;
+    }
+    ALOGW_IF(profile->mSupportedDevices == AUDIO_DEVICE_NONE,
+            "loadOutput() invalid supported devices");
+    ALOGW_IF(profile->mChannelMasks.size() == 0,
+            "loadOutput() invalid supported channel masks");
+    ALOGW_IF(profile->mSamplingRates.size() == 0,
+            "loadOutput() invalid supported sampling rates");
+    ALOGW_IF(profile->mFormats.size() == 0,
+            "loadOutput() invalid supported formats");
+    if ((profile->mSupportedDevices != AUDIO_DEVICE_NONE) &&
+            (profile->mChannelMasks.size() != 0) &&
+            (profile->mSamplingRates.size() != 0) &&
+            (profile->mFormats.size() != 0)) {
+
+        ALOGV("loadOutput() adding output mSupportedDevices %04x, mFlags %04x",
+              profile->mSupportedDevices, profile->mFlags);
+
+        module->mOutputProfiles.add(profile);
+        return NO_ERROR;
+    } else {
+        delete profile;
+        return BAD_VALUE;
+    }
+}
+
+void AudioPolicyManager::loadHwModule(cnode *root)
+{
+    cnode *node = config_find(root, OUTPUTS_TAG);
+    status_t status = NAME_NOT_FOUND;
+
+    HwModule *module = new HwModule(root->name);
+
+    if (node != NULL) {
+        if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_A2DP) == 0) {
+            mHasA2dp = true;
+        } else if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_USB) == 0) {
+            mHasUsb = true;
+        } else if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX) == 0) {
+            mHasRemoteSubmix = true;
+        }
+
+        node = node->first_child;
+        while (node) {
+            ALOGV("loadHwModule() loading output %s", node->name);
+            status_t tmpStatus = loadOutput(node, module);
+            if (status == NAME_NOT_FOUND || status == NO_ERROR) {
+                status = tmpStatus;
+            }
+            node = node->next;
+        }
+    }
+    node = config_find(root, INPUTS_TAG);
+    if (node != NULL) {
+        node = node->first_child;
+        while (node) {
+            ALOGV("loadHwModule() loading input %s", node->name);
+            status_t tmpStatus = loadInput(node, module);
+            if (status == NAME_NOT_FOUND || status == NO_ERROR) {
+                status = tmpStatus;
+            }
+            node = node->next;
+        }
+    }
+    if (status == NO_ERROR) {
+        mHwModules.add(module);
+    } else {
+        delete module;
+    }
+}
+
+void AudioPolicyManager::loadHwModules(cnode *root)
+{
+    cnode *node = config_find(root, AUDIO_HW_MODULE_TAG);
+    if (node == NULL) {
+        return;
+    }
+
+    node = node->first_child;
+    while (node) {
+        ALOGV("loadHwModules() loading module %s", node->name);
+        loadHwModule(node);
+        node = node->next;
+    }
+}
+
+void AudioPolicyManager::loadGlobalConfig(cnode *root)
+{
+    cnode *node = config_find(root, GLOBAL_CONFIG_TAG);
+    if (node == NULL) {
+        return;
+    }
+    node = node->first_child;
+    while (node) {
+        if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) {
+            mAttachedOutputDevices = parseDeviceNames((char *)node->value);
+            ALOGW_IF(mAttachedOutputDevices == AUDIO_DEVICE_NONE,
+                    "loadGlobalConfig() no attached output devices");
+            ALOGV("loadGlobalConfig() mAttachedOutputDevices %04x", mAttachedOutputDevices);
+        } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) {
+            mDefaultOutputDevice = (audio_devices_t)stringToEnum(sDeviceNameToEnumTable,
+                                              ARRAY_SIZE(sDeviceNameToEnumTable),
+                                              (char *)node->value);
+            ALOGW_IF(mDefaultOutputDevice == AUDIO_DEVICE_NONE,
+                    "loadGlobalConfig() default device not specified");
+            ALOGV("loadGlobalConfig() mDefaultOutputDevice %04x", mDefaultOutputDevice);
+        } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) {
+            mAvailableInputDevices = parseDeviceNames((char *)node->value) & ~AUDIO_DEVICE_BIT_IN;
+            ALOGV("loadGlobalConfig() mAvailableInputDevices %04x", mAvailableInputDevices);
+        } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) {
+            mSpeakerDrcEnabled = stringToBool((char *)node->value);
+            ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled);
+        }
+        node = node->next;
+    }
+}
+
+status_t AudioPolicyManager::loadAudioPolicyConfig(const char *path)
+{
+    cnode *root;
+    char *data;
+
+    data = (char *)load_file(path, NULL);
+    if (data == NULL) {
+        return -ENODEV;
+    }
+    root = config_node("", "");
+    config_load(root, data);
+
+    loadGlobalConfig(root);
+    loadHwModules(root);
+
+    config_free(root);
+    free(root);
+    free(data);
+
+    ALOGI("loadAudioPolicyConfig() loaded %s\n", path);
+
+    return NO_ERROR;
+}
+
+void AudioPolicyManager::defaultAudioPolicyConfig(void)
+{
+    HwModule *module;
+    IOProfile *profile;
+
+    mDefaultOutputDevice = AUDIO_DEVICE_OUT_SPEAKER;
+    mAttachedOutputDevices = AUDIO_DEVICE_OUT_SPEAKER;
+    mAvailableInputDevices = AUDIO_DEVICE_IN_BUILTIN_MIC & ~AUDIO_DEVICE_BIT_IN;
+
+    module = new HwModule("primary");
+
+    profile = new IOProfile(module);
+    profile->mSamplingRates.add(44100);
+    profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
+    profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO);
+    profile->mSupportedDevices = AUDIO_DEVICE_OUT_SPEAKER;
+    profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY;
+    module->mOutputProfiles.add(profile);
+
+    profile = new IOProfile(module);
+    profile->mSamplingRates.add(8000);
+    profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
+    profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO);
+    profile->mSupportedDevices = AUDIO_DEVICE_IN_BUILTIN_MIC;
+    module->mInputProfiles.add(profile);
+
+    mHwModules.add(module);
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/AudioPolicyManager.h b/services/audiopolicy/AudioPolicyManager.h
new file mode 100644
index 0000000..e00d8ab
--- /dev/null
+++ b/services/audiopolicy/AudioPolicyManager.h
@@ -0,0 +1,582 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <cutils/config_utils.h>
+#include <cutils/misc.h>
+#include <utils/Timers.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <utils/SortedVector.h>
+#include "AudioPolicyInterface.h"
+
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+#define MAX_DEVICE_ADDRESS_LEN 20
+// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
+#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5
+// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB
+#define SONIFICATION_HEADSET_VOLUME_MIN  0.016
+// Time in milliseconds during which we consider that music is still active after a music
+// track was stopped - see computeVolume()
+#define SONIFICATION_HEADSET_MUSIC_DELAY  5000
+// Time in milliseconds after media stopped playing during which we consider that the
+// sonification should be as unobtrusive as during the time media was playing.
+#define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000
+// Time in milliseconds during witch some streams are muted while the audio path
+// is switched
+#define MUTE_TIME_MS 2000
+
+#define NUM_TEST_OUTPUTS 5
+
+#define NUM_VOL_CURVE_KNEES 2
+
+// Default minimum length allowed for offloading a compressed track
+// Can be overridden by the audio.offload.min.duration.secs property
+#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManager implements audio policy manager behavior common to all platforms.
+// ----------------------------------------------------------------------------
+
+class AudioPolicyManager: public AudioPolicyInterface
+#ifdef AUDIO_POLICY_TEST
+    , public Thread
+#endif //AUDIO_POLICY_TEST
+{
+
+public:
+                AudioPolicyManager(AudioPolicyClientInterface *clientInterface);
+        virtual ~AudioPolicyManager();
+
+        // AudioPolicyInterface
+        virtual status_t setDeviceConnectionState(audio_devices_t device,
+                                                          audio_policy_dev_state_t state,
+                                                          const char *device_address);
+        virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
+                                                                              const char *device_address);
+        virtual void setPhoneState(audio_mode_t state);
+        virtual void setForceUse(audio_policy_force_use_t usage,
+                                 audio_policy_forced_cfg_t config);
+        virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
+        virtual void setSystemProperty(const char* property, const char* value);
+        virtual status_t initCheck();
+        virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
+                                            uint32_t samplingRate,
+                                            audio_format_t format,
+                                            audio_channel_mask_t channelMask,
+                                            audio_output_flags_t flags,
+                                            const audio_offload_info_t *offloadInfo);
+        virtual status_t startOutput(audio_io_handle_t output,
+                                     audio_stream_type_t stream,
+                                     int session = 0);
+        virtual status_t stopOutput(audio_io_handle_t output,
+                                    audio_stream_type_t stream,
+                                    int session = 0);
+        virtual void releaseOutput(audio_io_handle_t output);
+        virtual audio_io_handle_t getInput(audio_source_t inputSource,
+                                            uint32_t samplingRate,
+                                            audio_format_t format,
+                                            audio_channel_mask_t channelMask,
+                                            audio_in_acoustics_t acoustics);
+
+        // indicates to the audio policy manager that the input starts being used.
+        virtual status_t startInput(audio_io_handle_t input);
+
+        // indicates to the audio policy manager that the input stops being used.
+        virtual status_t stopInput(audio_io_handle_t input);
+        virtual void releaseInput(audio_io_handle_t input);
+        virtual void initStreamVolume(audio_stream_type_t stream,
+                                                    int indexMin,
+                                                    int indexMax);
+        virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
+                                              int index,
+                                              audio_devices_t device);
+        virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
+                                              int *index,
+                                              audio_devices_t device);
+
+        // return the strategy corresponding to a given stream type
+        virtual uint32_t getStrategyForStream(audio_stream_type_t stream);
+
+        // return the enabled output devices for the given stream type
+        virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream);
+
+        virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL);
+        virtual status_t registerEffect(const effect_descriptor_t *desc,
+                                        audio_io_handle_t io,
+                                        uint32_t strategy,
+                                        int session,
+                                        int id);
+        virtual status_t unregisterEffect(int id);
+        virtual status_t setEffectEnabled(int id, bool enabled);
+
+        virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
+        // return whether a stream is playing remotely, override to change the definition of
+        //   local/remote playback, used for instance by notification manager to not make
+        //   media players lose audio focus when not playing locally
+        virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
+        virtual bool isSourceActive(audio_source_t source) const;
+
+        virtual status_t dump(int fd);
+
+        virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
+
+protected:
+
+        enum routing_strategy {
+            STRATEGY_MEDIA,
+            STRATEGY_PHONE,
+            STRATEGY_SONIFICATION,
+            STRATEGY_SONIFICATION_RESPECTFUL,
+            STRATEGY_DTMF,
+            STRATEGY_ENFORCED_AUDIBLE,
+            NUM_STRATEGIES
+        };
+
+        // 4 points to define the volume attenuation curve, each characterized by the volume
+        // index (from 0 to 100) at which they apply, and the attenuation in dB at that index.
+        // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl()
+
+        enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4};
+
+        class VolumeCurvePoint
+        {
+        public:
+            int mIndex;
+            float mDBAttenuation;
+        };
+
+        // device categories used for volume curve management.
+        enum device_category {
+            DEVICE_CATEGORY_HEADSET,
+            DEVICE_CATEGORY_SPEAKER,
+            DEVICE_CATEGORY_EARPIECE,
+            DEVICE_CATEGORY_CNT
+        };
+
+        class IOProfile;
+
+        class HwModule {
+        public:
+                    HwModule(const char *name);
+                    ~HwModule();
+
+            void dump(int fd);
+
+            const char *const mName; // base name of the audio HW module (primary, a2dp ...)
+            audio_module_handle_t mHandle;
+            Vector <IOProfile *> mOutputProfiles; // output profiles exposed by this module
+            Vector <IOProfile *> mInputProfiles;  // input profiles exposed by this module
+        };
+
+        // the IOProfile class describes the capabilities of an output or input stream.
+        // It is currently assumed that all combination of listed parameters are supported.
+        // It is used by the policy manager to determine if an output or input is suitable for
+        // a given use case,  open/close it accordingly and connect/disconnect audio tracks
+        // to/from it.
+        class IOProfile
+        {
+        public:
+            IOProfile(HwModule *module);
+            ~IOProfile();
+
+            bool isCompatibleProfile(audio_devices_t device,
+                                     uint32_t samplingRate,
+                                     audio_format_t format,
+                                     audio_channel_mask_t channelMask,
+                                     audio_output_flags_t flags) const;
+
+            void dump(int fd);
+
+            // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats
+            // indicates the supported parameters should be read from the output stream
+            // after it is opened for the first time
+            Vector <uint32_t> mSamplingRates; // supported sampling rates
+            Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks
+            Vector <audio_format_t> mFormats; // supported audio formats
+            audio_devices_t mSupportedDevices; // supported devices (devices this output can be
+                                               // routed to)
+            audio_output_flags_t mFlags; // attribute flags (e.g primary output,
+                                                // direct output...). For outputs only.
+            HwModule *mModule;                     // audio HW module exposing this I/O stream
+        };
+
+        // default volume curve
+        static const VolumeCurvePoint sDefaultVolumeCurve[AudioPolicyManager::VOLCNT];
+        // default volume curve for media strategy
+        static const VolumeCurvePoint sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT];
+        // volume curve for media strategy on speakers
+        static const VolumeCurvePoint sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT];
+        // volume curve for sonification strategy on speakers
+        static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT];
+        static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT];
+        static const VolumeCurvePoint sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT];
+        static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT];
+        static const VolumeCurvePoint sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT];
+        static const VolumeCurvePoint sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT];
+        static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT];
+        // default volume curves per stream and device category. See initializeVolumeCurves()
+        static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][DEVICE_CATEGORY_CNT];
+
+        // descriptor for audio outputs. Used to maintain current configuration of each opened audio output
+        // and keep track of the usage of this output by each audio stream type.
+        class AudioOutputDescriptor
+        {
+        public:
+            AudioOutputDescriptor(const IOProfile *profile);
+
+            status_t    dump(int fd);
+
+            audio_devices_t device() const;
+            void changeRefCount(audio_stream_type_t stream, int delta);
+
+            bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
+            audio_devices_t supportedDevices();
+            uint32_t latency();
+            bool sharesHwModuleWith(const AudioOutputDescriptor *outputDesc);
+            bool isActive(uint32_t inPastMs = 0) const;
+            bool isStreamActive(audio_stream_type_t stream,
+                                uint32_t inPastMs = 0,
+                                nsecs_t sysTime = 0) const;
+            bool isStrategyActive(routing_strategy strategy,
+                             uint32_t inPastMs = 0,
+                             nsecs_t sysTime = 0) const;
+
+            audio_io_handle_t mId;              // output handle
+            uint32_t mSamplingRate;             //
+            audio_format_t mFormat;             //
+            audio_channel_mask_t mChannelMask;     // output configuration
+            uint32_t mLatency;                  //
+            audio_output_flags_t mFlags;   //
+            audio_devices_t mDevice;                   // current device this output is routed to
+            uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output
+            nsecs_t mStopTime[AUDIO_STREAM_CNT];
+            AudioOutputDescriptor *mOutput1;    // used by duplicated outputs: first output
+            AudioOutputDescriptor *mOutput2;    // used by duplicated outputs: second output
+            float mCurVolume[AUDIO_STREAM_CNT];   // current stream volume
+            int mMuteCount[AUDIO_STREAM_CNT];     // mute request counter
+            const IOProfile *mProfile;          // I/O profile this output derives from
+            bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible
+                                                // device selection. See checkDeviceMuteStrategies()
+            uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
+        };
+
+        // descriptor for audio inputs. Used to maintain current configuration of each opened audio input
+        // and keep track of the usage of this input.
+        class AudioInputDescriptor
+        {
+        public:
+            AudioInputDescriptor(const IOProfile *profile);
+
+            status_t    dump(int fd);
+
+            uint32_t mSamplingRate;                     //
+            audio_format_t mFormat;                     // input configuration
+            audio_channel_mask_t mChannelMask;             //
+            audio_devices_t mDevice;                    // current device this input is routed to
+            uint32_t mRefCount;                         // number of AudioRecord clients using this output
+            audio_source_t mInputSource;                // input source selected by application (mediarecorder.h)
+            const IOProfile *mProfile;                  // I/O profile this output derives from
+        };
+
+        // stream descriptor used for volume control
+        class StreamDescriptor
+        {
+        public:
+            StreamDescriptor();
+
+            int getVolumeIndex(audio_devices_t device);
+            void dump(int fd);
+
+            int mIndexMin;      // min volume index
+            int mIndexMax;      // max volume index
+            KeyedVector<audio_devices_t, int> mIndexCur;   // current volume index per device
+            bool mCanBeMuted;   // true is the stream can be muted
+
+            const VolumeCurvePoint *mVolumeCurve[DEVICE_CATEGORY_CNT];
+        };
+
+        // stream descriptor used for volume control
+        class EffectDescriptor
+        {
+        public:
+
+            status_t dump(int fd);
+
+            int mIo;                // io the effect is attached to
+            routing_strategy mStrategy; // routing strategy the effect is associated to
+            int mSession;               // audio session the effect is on
+            effect_descriptor_t mDesc;  // effect descriptor
+            bool mEnabled;              // enabled state: CPU load being used or not
+        };
+
+        void addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc);
+
+        // return the strategy corresponding to a given stream type
+        static routing_strategy getStrategy(audio_stream_type_t stream);
+
+        // return appropriate device for streams handled by the specified strategy according to current
+        // phone state, connected devices...
+        // if fromCache is true, the device is returned from mDeviceForStrategy[],
+        // otherwise it is determine by current state
+        // (device connected,phone state, force use, a2dp output...)
+        // This allows to:
+        //  1 speed up process when the state is stable (when starting or stopping an output)
+        //  2 access to either current device selection (fromCache == true) or
+        // "future" device selection (fromCache == false) when called from a context
+        //  where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
+        //  before updateDevicesAndOutputs() is called.
+        virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy,
+                                                     bool fromCache);
+
+        // change the route of the specified output. Returns the number of ms we have slept to
+        // allow new routing to take effect in certain cases.
+        uint32_t setOutputDevice(audio_io_handle_t output,
+                             audio_devices_t device,
+                             bool force = false,
+                             int delayMs = 0);
+
+        // select input device corresponding to requested audio source
+        virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource);
+
+        // return io handle of active input or 0 if no input is active
+        //    Only considers inputs from physical devices (e.g. main mic, headset mic) when
+        //    ignoreVirtualInputs is true.
+        audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true);
+
+        // initialize volume curves for each strategy and device category
+        void initializeVolumeCurves();
+
+        // compute the actual volume for a given stream according to the requested index and a particular
+        // device
+        virtual float computeVolume(audio_stream_type_t stream, int index,
+                                    audio_io_handle_t output, audio_devices_t device);
+
+        // check that volume change is permitted, compute and send new volume to audio hardware
+        status_t checkAndSetVolume(audio_stream_type_t stream, int index, audio_io_handle_t output,
+                                   audio_devices_t device, int delayMs = 0, bool force = false);
+
+        // apply all stream volumes to the specified output and device
+        void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
+
+        // Mute or unmute all streams handled by the specified strategy on the specified output
+        void setStrategyMute(routing_strategy strategy,
+                             bool on,
+                             audio_io_handle_t output,
+                             int delayMs = 0,
+                             audio_devices_t device = (audio_devices_t)0);
+
+        // Mute or unmute the stream on the specified output
+        void setStreamMute(audio_stream_type_t stream,
+                           bool on,
+                           audio_io_handle_t output,
+                           int delayMs = 0,
+                           audio_devices_t device = (audio_devices_t)0);
+
+        // handle special cases for sonification strategy while in call: mute streams or replace by
+        // a special tone in the device used for communication
+        void handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange);
+
+        // true if device is in a telephony or VoIP call
+        virtual bool isInCall();
+
+        // true if given state represents a device in a telephony or VoIP call
+        virtual bool isStateInCall(int state);
+
+        // when a device is connected, checks if an open output can be routed
+        // to this device. If none is open, tries to open one of the available outputs.
+        // Returns an output suitable to this device or 0.
+        // when a device is disconnected, checks if an output is not used any more and
+        // returns its handle if any.
+        // transfers the audio tracks and effects from one output thread to another accordingly.
+        status_t checkOutputsForDevice(audio_devices_t device,
+                                       audio_policy_dev_state_t state,
+                                       SortedVector<audio_io_handle_t>& outputs,
+                                       const String8 paramStr);
+
+        // close an output and its companion duplicating output.
+        void closeOutput(audio_io_handle_t output);
+
+        // checks and if necessary changes outputs used for all strategies.
+        // must be called every time a condition that affects the output choice for a given strategy
+        // changes: connected device, phone state, force use...
+        // Must be called before updateDevicesAndOutputs()
+        void checkOutputForStrategy(routing_strategy strategy);
+
+        // Same as checkOutputForStrategy() but for a all strategies in order of priority
+        void checkOutputForAllStrategies();
+
+        // manages A2DP output suspend/restore according to phone state and BT SCO usage
+        void checkA2dpSuspend();
+
+        // returns the A2DP output handle if it is open or 0 otherwise
+        audio_io_handle_t getA2dpOutput();
+
+        // selects the most appropriate device on output for current state
+        // must be called every time a condition that affects the device choice for a given output is
+        // changed: connected device, phone state, force use, output start, output stop..
+        // see getDeviceForStrategy() for the use of fromCache parameter
+
+        audio_devices_t getNewDevice(audio_io_handle_t output, bool fromCache);
+        // updates cache of device used by all strategies (mDeviceForStrategy[])
+        // must be called every time a condition that affects the device choice for a given strategy is
+        // changed: connected device, phone state, force use...
+        // cached values are used by getDeviceForStrategy() if parameter fromCache is true.
+         // Must be called after checkOutputForAllStrategies()
+
+        void updateDevicesAndOutputs();
+
+        virtual uint32_t getMaxEffectsCpuLoad();
+        virtual uint32_t getMaxEffectsMemory();
+#ifdef AUDIO_POLICY_TEST
+        virtual     bool        threadLoop();
+                    void        exit();
+        int testOutputIndex(audio_io_handle_t output);
+#endif //AUDIO_POLICY_TEST
+
+        status_t setEffectEnabled(EffectDescriptor *pDesc, bool enabled);
+
+        // returns the category the device belongs to with regard to volume curve management
+        static device_category getDeviceCategory(audio_devices_t device);
+
+        // extract one device relevant for volume control from multiple device selection
+        static audio_devices_t getDeviceForVolume(audio_devices_t device);
+
+        SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
+                        DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> openOutputs);
+        bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
+                                           SortedVector<audio_io_handle_t>& outputs2);
+
+        // mute/unmute strategies using an incompatible device combination
+        // if muting, wait for the audio in pcm buffer to be drained before proceeding
+        // if unmuting, unmute only after the specified delay
+        // Returns the number of ms waited
+        uint32_t  checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc,
+                                            audio_devices_t prevDevice,
+                                            uint32_t delayMs);
+
+        audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
+                                       audio_output_flags_t flags);
+        IOProfile *getInputProfile(audio_devices_t device,
+                                   uint32_t samplingRate,
+                                   audio_format_t format,
+                                   audio_channel_mask_t channelMask);
+        IOProfile *getProfileForDirectOutput(audio_devices_t device,
+                                                       uint32_t samplingRate,
+                                                       audio_format_t format,
+                                                       audio_channel_mask_t channelMask,
+                                                       audio_output_flags_t flags);
+
+        audio_io_handle_t selectOutputForEffects(const SortedVector<audio_io_handle_t>& outputs);
+
+        bool isNonOffloadableEffectEnabled();
+
+        //
+        // Audio policy configuration file parsing (audio_policy.conf)
+        //
+        static uint32_t stringToEnum(const struct StringToEnum *table,
+                                     size_t size,
+                                     const char *name);
+        static bool stringToBool(const char *value);
+        static audio_output_flags_t parseFlagNames(char *name);
+        static audio_devices_t parseDeviceNames(char *name);
+        void loadSamplingRates(char *name, IOProfile *profile);
+        void loadFormats(char *name, IOProfile *profile);
+        void loadOutChannels(char *name, IOProfile *profile);
+        void loadInChannels(char *name, IOProfile *profile);
+        status_t loadOutput(cnode *root,  HwModule *module);
+        status_t loadInput(cnode *root,  HwModule *module);
+        void loadHwModule(cnode *root);
+        void loadHwModules(cnode *root);
+        void loadGlobalConfig(cnode *root);
+        status_t loadAudioPolicyConfig(const char *path);
+        void defaultAudioPolicyConfig(void);
+
+
+        AudioPolicyClientInterface *mpClientInterface;  // audio policy client interface
+        audio_io_handle_t mPrimaryOutput;              // primary output handle
+        // list of descriptors for outputs currently opened
+        DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mOutputs;
+        // copy of mOutputs before setDeviceConnectionState() opens new outputs
+        // reset to mOutputs when updateDevicesAndOutputs() is called.
+        DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mPreviousOutputs;
+        DefaultKeyedVector<audio_io_handle_t, AudioInputDescriptor *> mInputs;     // list of input descriptors
+        audio_devices_t mAvailableOutputDevices; // bit field of all available output devices
+        audio_devices_t mAvailableInputDevices; // bit field of all available input devices
+                                                // without AUDIO_DEVICE_BIT_IN to allow direct bit
+                                                // field comparisons
+        int mPhoneState;                                                    // current phone state
+        audio_policy_forced_cfg_t mForceUse[AUDIO_POLICY_FORCE_USE_CNT];   // current forced use configuration
+
+        StreamDescriptor mStreams[AUDIO_STREAM_CNT];           // stream descriptors for volume control
+        String8 mA2dpDeviceAddress;                                         // A2DP device MAC address
+        String8 mScoDeviceAddress;                                          // SCO device MAC address
+        String8 mUsbCardAndDevice; // USB audio ALSA card and device numbers:
+                                   // card=<card_number>;device=<><device_number>
+        bool    mLimitRingtoneVolume;                                       // limit ringtone volume to music volume if headset connected
+        audio_devices_t mDeviceForStrategy[NUM_STRATEGIES];
+        float   mLastVoiceVolume;                                           // last voice volume value sent to audio HAL
+
+        // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units
+        static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000;
+        // Maximum memory allocated to audio effects in KB
+        static const uint32_t MAX_EFFECTS_MEMORY = 512;
+        uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects
+        uint32_t mTotalEffectsMemory;  // current memory used by effects
+        KeyedVector<int, EffectDescriptor *> mEffects;  // list of registered audio effects
+        bool    mA2dpSuspended;  // true if A2DP output is suspended
+        bool mHasA2dp; // true on platforms with support for bluetooth A2DP
+        bool mHasUsb; // true on platforms with support for USB audio
+        bool mHasRemoteSubmix; // true on platforms with support for remote presentation of a submix
+        audio_devices_t mAttachedOutputDevices; // output devices always available on the platform
+        audio_devices_t mDefaultOutputDevice; // output device selected by default at boot time
+                                              // (must be in mAttachedOutputDevices)
+        bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path
+                                // to boost soft sounds, used to adjust volume curves accordingly
+
+        Vector <HwModule *> mHwModules;
+
+#ifdef AUDIO_POLICY_TEST
+        Mutex   mLock;
+        Condition mWaitWorkCV;
+
+        int             mCurOutput;
+        bool            mDirectOutput;
+        audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS];
+        int             mTestInput;
+        uint32_t        mTestDevice;
+        uint32_t        mTestSamplingRate;
+        uint32_t        mTestFormat;
+        uint32_t        mTestChannels;
+        uint32_t        mTestLatencyMs;
+#endif //AUDIO_POLICY_TEST
+
+private:
+        static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+                int indexInUi);
+        // updates device caching and output for streams that can influence the
+        //    routing of notifications
+        void handleNotificationRoutingForStream(audio_stream_type_t stream);
+        static bool isVirtualInputDevice(audio_devices_t device);
+};
+
+};
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audiopolicy/AudioPolicyService.cpp
similarity index 65%
rename from services/audioflinger/AudioPolicyService.cpp
rename to services/audiopolicy/AudioPolicyService.cpp
index 41bd990..49145a5 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audiopolicy/AudioPolicyService.cpp
@@ -138,445 +138,6 @@
     }
 }
 
-status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device,
-                                                  audio_policy_dev_state_t state,
-                                                  const char *device_address)
-{
-    if (mpAudioPolicy == NULL) {
-        return NO_INIT;
-    }
-    if (!settingsAllowed()) {
-        return PERMISSION_DENIED;
-    }
-    if (!audio_is_output_device(device) && !audio_is_input_device(device)) {
-        return BAD_VALUE;
-    }
-    if (state != AUDIO_POLICY_DEVICE_STATE_AVAILABLE &&
-            state != AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
-        return BAD_VALUE;
-    }
-
-    ALOGV("setDeviceConnectionState()");
-    Mutex::Autolock _l(mLock);
-    return mpAudioPolicy->set_device_connection_state(mpAudioPolicy, device,
-                                                      state, device_address);
-}
-
-audio_policy_dev_state_t AudioPolicyService::getDeviceConnectionState(
-                                                              audio_devices_t device,
-                                                              const char *device_address)
-{
-    if (mpAudioPolicy == NULL) {
-        return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
-    }
-    return mpAudioPolicy->get_device_connection_state(mpAudioPolicy, device,
-                                                      device_address);
-}
-
-status_t AudioPolicyService::setPhoneState(audio_mode_t state)
-{
-    if (mpAudioPolicy == NULL) {
-        return NO_INIT;
-    }
-    if (!settingsAllowed()) {
-        return PERMISSION_DENIED;
-    }
-    if (uint32_t(state) >= AUDIO_MODE_CNT) {
-        return BAD_VALUE;
-    }
-
-    ALOGV("setPhoneState()");
-
-    // TODO: check if it is more appropriate to do it in platform specific policy manager
-    AudioSystem::setMode(state);
-
-    Mutex::Autolock _l(mLock);
-    mpAudioPolicy->set_phone_state(mpAudioPolicy, state);
-    return NO_ERROR;
-}
-
-status_t AudioPolicyService::setForceUse(audio_policy_force_use_t usage,
-                                         audio_policy_forced_cfg_t config)
-{
-    if (mpAudioPolicy == NULL) {
-        return NO_INIT;
-    }
-    if (!settingsAllowed()) {
-        return PERMISSION_DENIED;
-    }
-    if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) {
-        return BAD_VALUE;
-    }
-    if (config < 0 || config >= AUDIO_POLICY_FORCE_CFG_CNT) {
-        return BAD_VALUE;
-    }
-    ALOGV("setForceUse()");
-    Mutex::Autolock _l(mLock);
-    mpAudioPolicy->set_force_use(mpAudioPolicy, usage, config);
-    return NO_ERROR;
-}
-
-audio_policy_forced_cfg_t AudioPolicyService::getForceUse(audio_policy_force_use_t usage)
-{
-    if (mpAudioPolicy == NULL) {
-        return AUDIO_POLICY_FORCE_NONE;
-    }
-    if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) {
-        return AUDIO_POLICY_FORCE_NONE;
-    }
-    return mpAudioPolicy->get_force_use(mpAudioPolicy, usage);
-}
-
-audio_io_handle_t AudioPolicyService::getOutput(audio_stream_type_t stream,
-                                    uint32_t samplingRate,
-                                    audio_format_t format,
-                                    audio_channel_mask_t channelMask,
-                                    audio_output_flags_t flags,
-                                    const audio_offload_info_t *offloadInfo)
-{
-    if (mpAudioPolicy == NULL) {
-        return 0;
-    }
-    ALOGV("getOutput()");
-    Mutex::Autolock _l(mLock);
-    return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate,
-                                    format, channelMask, flags, offloadInfo);
-}
-
-status_t AudioPolicyService::startOutput(audio_io_handle_t output,
-                                         audio_stream_type_t stream,
-                                         int session)
-{
-    if (mpAudioPolicy == NULL) {
-        return NO_INIT;
-    }
-    ALOGV("startOutput()");
-    Mutex::Autolock _l(mLock);
-    return mpAudioPolicy->start_output(mpAudioPolicy, output, stream, session);
-}
-
-status_t AudioPolicyService::stopOutput(audio_io_handle_t output,
-                                        audio_stream_type_t stream,
-                                        int session)
-{
-    if (mpAudioPolicy == NULL) {
-        return NO_INIT;
-    }
-    ALOGV("stopOutput()");
-    mOutputCommandThread->stopOutputCommand(output, stream, session);
-    return NO_ERROR;
-}
-
-status_t  AudioPolicyService::doStopOutput(audio_io_handle_t output,
-                                      audio_stream_type_t stream,
-                                      int session)
-{
-    ALOGV("doStopOutput from tid %d", gettid());
-    Mutex::Autolock _l(mLock);
-    return mpAudioPolicy->stop_output(mpAudioPolicy, output, stream, session);
-}
-
-void AudioPolicyService::releaseOutput(audio_io_handle_t output)
-{
-    if (mpAudioPolicy == NULL) {
-        return;
-    }
-    ALOGV("releaseOutput()");
-    mOutputCommandThread->releaseOutputCommand(output);
-}
-
-void AudioPolicyService::doReleaseOutput(audio_io_handle_t output)
-{
-    ALOGV("doReleaseOutput from tid %d", gettid());
-    Mutex::Autolock _l(mLock);
-    mpAudioPolicy->release_output(mpAudioPolicy, output);
-}
-
-audio_io_handle_t AudioPolicyService::getInput(audio_source_t inputSource,
-                                    uint32_t samplingRate,
-                                    audio_format_t format,
-                                    audio_channel_mask_t channelMask,
-                                    int audioSession)
-{
-    if (mpAudioPolicy == NULL) {
-        return 0;
-    }
-    // already checked by client, but double-check in case the client wrapper is bypassed
-    if (inputSource >= AUDIO_SOURCE_CNT && inputSource != AUDIO_SOURCE_HOTWORD) {
-        return 0;
-    }
-
-    if ((inputSource == AUDIO_SOURCE_HOTWORD) && !captureHotwordAllowed()) {
-        return 0;
-    }
-
-    Mutex::Autolock _l(mLock);
-    // the audio_in_acoustics_t parameter is ignored by get_input()
-    audio_io_handle_t input = mpAudioPolicy->get_input(mpAudioPolicy, inputSource, samplingRate,
-                                                   format, channelMask, (audio_in_acoustics_t) 0);
-
-    if (input == 0) {
-        return input;
-    }
-    // create audio pre processors according to input source
-    audio_source_t aliasSource = (inputSource == AUDIO_SOURCE_HOTWORD) ?
-                                    AUDIO_SOURCE_VOICE_RECOGNITION : inputSource;
-
-    ssize_t index = mInputSources.indexOfKey(aliasSource);
-    if (index < 0) {
-        return input;
-    }
-    ssize_t idx = mInputs.indexOfKey(input);
-    InputDesc *inputDesc;
-    if (idx < 0) {
-        inputDesc = new InputDesc(audioSession);
-        mInputs.add(input, inputDesc);
-    } else {
-        inputDesc = mInputs.valueAt(idx);
-    }
-
-    Vector <EffectDesc *> effects = mInputSources.valueAt(index)->mEffects;
-    for (size_t i = 0; i < effects.size(); i++) {
-        EffectDesc *effect = effects[i];
-        sp<AudioEffect> fx = new AudioEffect(NULL, &effect->mUuid, -1, 0, 0, audioSession, input);
-        status_t status = fx->initCheck();
-        if (status != NO_ERROR && status != ALREADY_EXISTS) {
-            ALOGW("Failed to create Fx %s on input %d", effect->mName, input);
-            // fx goes out of scope and strong ref on AudioEffect is released
-            continue;
-        }
-        for (size_t j = 0; j < effect->mParams.size(); j++) {
-            fx->setParameter(effect->mParams[j]);
-        }
-        inputDesc->mEffects.add(fx);
-    }
-    setPreProcessorEnabled(inputDesc, true);
-    return input;
-}
-
-status_t AudioPolicyService::startInput(audio_io_handle_t input)
-{
-    if (mpAudioPolicy == NULL) {
-        return NO_INIT;
-    }
-    Mutex::Autolock _l(mLock);
-
-    return mpAudioPolicy->start_input(mpAudioPolicy, input);
-}
-
-status_t AudioPolicyService::stopInput(audio_io_handle_t input)
-{
-    if (mpAudioPolicy == NULL) {
-        return NO_INIT;
-    }
-    Mutex::Autolock _l(mLock);
-
-    return mpAudioPolicy->stop_input(mpAudioPolicy, input);
-}
-
-void AudioPolicyService::releaseInput(audio_io_handle_t input)
-{
-    if (mpAudioPolicy == NULL) {
-        return;
-    }
-    Mutex::Autolock _l(mLock);
-    mpAudioPolicy->release_input(mpAudioPolicy, input);
-
-    ssize_t index = mInputs.indexOfKey(input);
-    if (index < 0) {
-        return;
-    }
-    InputDesc *inputDesc = mInputs.valueAt(index);
-    setPreProcessorEnabled(inputDesc, false);
-    delete inputDesc;
-    mInputs.removeItemsAt(index);
-}
-
-status_t AudioPolicyService::initStreamVolume(audio_stream_type_t stream,
-                                            int indexMin,
-                                            int indexMax)
-{
-    if (mpAudioPolicy == NULL) {
-        return NO_INIT;
-    }
-    if (!settingsAllowed()) {
-        return PERMISSION_DENIED;
-    }
-    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
-        return BAD_VALUE;
-    }
-    Mutex::Autolock _l(mLock);
-    mpAudioPolicy->init_stream_volume(mpAudioPolicy, stream, indexMin, indexMax);
-    return NO_ERROR;
-}
-
-status_t AudioPolicyService::setStreamVolumeIndex(audio_stream_type_t stream,
-                                                  int index,
-                                                  audio_devices_t device)
-{
-    if (mpAudioPolicy == NULL) {
-        return NO_INIT;
-    }
-    if (!settingsAllowed()) {
-        return PERMISSION_DENIED;
-    }
-    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
-        return BAD_VALUE;
-    }
-    Mutex::Autolock _l(mLock);
-    if (mpAudioPolicy->set_stream_volume_index_for_device) {
-        return mpAudioPolicy->set_stream_volume_index_for_device(mpAudioPolicy,
-                                                                stream,
-                                                                index,
-                                                                device);
-    } else {
-        return mpAudioPolicy->set_stream_volume_index(mpAudioPolicy, stream, index);
-    }
-}
-
-status_t AudioPolicyService::getStreamVolumeIndex(audio_stream_type_t stream,
-                                                  int *index,
-                                                  audio_devices_t device)
-{
-    if (mpAudioPolicy == NULL) {
-        return NO_INIT;
-    }
-    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
-        return BAD_VALUE;
-    }
-    Mutex::Autolock _l(mLock);
-    if (mpAudioPolicy->get_stream_volume_index_for_device) {
-        return mpAudioPolicy->get_stream_volume_index_for_device(mpAudioPolicy,
-                                                                stream,
-                                                                index,
-                                                                device);
-    } else {
-        return mpAudioPolicy->get_stream_volume_index(mpAudioPolicy, stream, index);
-    }
-}
-
-uint32_t AudioPolicyService::getStrategyForStream(audio_stream_type_t stream)
-{
-    if (mpAudioPolicy == NULL) {
-        return 0;
-    }
-    return mpAudioPolicy->get_strategy_for_stream(mpAudioPolicy, stream);
-}
-
-//audio policy: use audio_device_t appropriately
-
-audio_devices_t AudioPolicyService::getDevicesForStream(audio_stream_type_t stream)
-{
-    if (mpAudioPolicy == NULL) {
-        return (audio_devices_t)0;
-    }
-    return mpAudioPolicy->get_devices_for_stream(mpAudioPolicy, stream);
-}
-
-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 0;
-    }
-    Mutex::Autolock _l(mLock);
-    return mpAudioPolicy->get_output_for_effect(mpAudioPolicy, desc);
-}
-
-status_t AudioPolicyService::registerEffect(const effect_descriptor_t *desc,
-                                audio_io_handle_t io,
-                                uint32_t strategy,
-                                int session,
-                                int id)
-{
-    if (mpAudioPolicy == NULL) {
-        return NO_INIT;
-    }
-    return mpAudioPolicy->register_effect(mpAudioPolicy, desc, io, strategy, session, id);
-}
-
-status_t AudioPolicyService::unregisterEffect(int id)
-{
-    if (mpAudioPolicy == NULL) {
-        return NO_INIT;
-    }
-    return mpAudioPolicy->unregister_effect(mpAudioPolicy, id);
-}
-
-status_t AudioPolicyService::setEffectEnabled(int id, bool enabled)
-{
-    if (mpAudioPolicy == NULL) {
-        return NO_INIT;
-    }
-    return mpAudioPolicy->set_effect_enabled(mpAudioPolicy, id, enabled);
-}
-
-bool AudioPolicyService::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
-{
-    if (mpAudioPolicy == NULL) {
-        return 0;
-    }
-    Mutex::Autolock _l(mLock);
-    return mpAudioPolicy->is_stream_active(mpAudioPolicy, stream, inPastMs);
-}
-
-bool AudioPolicyService::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const
-{
-    if (mpAudioPolicy == NULL) {
-        return 0;
-    }
-    Mutex::Autolock _l(mLock);
-    return mpAudioPolicy->is_stream_active_remotely(mpAudioPolicy, stream, inPastMs);
-}
-
-bool AudioPolicyService::isSourceActive(audio_source_t source) const
-{
-    if (mpAudioPolicy == NULL) {
-        return false;
-    }
-    if (mpAudioPolicy->is_source_active == 0) {
-        return false;
-    }
-    Mutex::Autolock _l(mLock);
-    return mpAudioPolicy->is_source_active(mpAudioPolicy, source);
-}
-
-status_t AudioPolicyService::queryDefaultPreProcessing(int audioSession,
-                                                       effect_descriptor_t *descriptors,
-                                                       uint32_t *count)
-{
-
-    if (mpAudioPolicy == NULL) {
-        *count = 0;
-        return NO_INIT;
-    }
-    Mutex::Autolock _l(mLock);
-    status_t status = NO_ERROR;
-
-    size_t index;
-    for (index = 0; index < mInputs.size(); index++) {
-        if (mInputs.valueAt(index)->mSessionId == audioSession) {
-            break;
-        }
-    }
-    if (index == mInputs.size()) {
-        *count = 0;
-        return BAD_VALUE;
-    }
-    Vector< sp<AudioEffect> > effects = mInputs.valueAt(index)->mEffects;
-
-    for (size_t i = 0; i < effects.size(); i++) {
-        effect_descriptor_t desc = effects[i]->descriptor();
-        if (i < *count) {
-            descriptors[i] = desc;
-        }
-    }
-    if (effects.size() > *count) {
-        status = NO_MEMORY;
-    }
-    *count = effects.size();
-    return status;
-}
 
 void AudioPolicyService::binderDied(const wp<IBinder>& who) {
     ALOGW("binderDied() %p, calling pid %d", who.unsafe_get(),
@@ -1144,21 +705,6 @@
     return (int)mAudioCommandThread->voiceVolumeCommand(volume, delayMs);
 }
 
-bool AudioPolicyService::isOffloadSupported(const audio_offload_info_t& info)
-{
-    if (mpAudioPolicy == NULL) {
-        ALOGV("mpAudioPolicy == NULL");
-        return false;
-    }
-
-    if (mpAudioPolicy->is_offload_supported == NULL) {
-        ALOGV("HAL does not implement is_offload_supported");
-        return false;
-    }
-
-    return mpAudioPolicy->is_offload_supported(mpAudioPolicy, &info);
-}
-
 // ----------------------------------------------------------------------------
 // Audio pre-processing configuration
 // ----------------------------------------------------------------------------
@@ -1457,42 +1003,18 @@
     return NO_ERROR;
 }
 
-/* implementation of the interface to the policy manager */
 extern "C" {
-
-
-static audio_module_handle_t aps_load_hw_module(void *service __unused,
-                                             const char *name)
-{
-    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
-    if (af == 0) {
-        ALOGW("%s: could not get AudioFlinger", __func__);
-        return 0;
-    }
-
-    return af->loadHwModule(name);
-}
-
-// deprecated: replaced by aps_open_output_on_module()
-static audio_io_handle_t aps_open_output(void *service __unused,
+audio_module_handle_t aps_load_hw_module(void *service __unused,
+                                             const char *name);
+audio_io_handle_t aps_open_output(void *service __unused,
                                          audio_devices_t *pDevices,
                                          uint32_t *pSamplingRate,
                                          audio_format_t *pFormat,
                                          audio_channel_mask_t *pChannelMask,
                                          uint32_t *pLatencyMs,
-                                         audio_output_flags_t flags)
-{
-    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
-    if (af == 0) {
-        ALOGW("%s: could not get AudioFlinger", __func__);
-        return 0;
-    }
+                                         audio_output_flags_t flags);
 
-    return af->openOutput((audio_module_handle_t)0, pDevices, pSamplingRate, pFormat, pChannelMask,
-                          pLatencyMs, flags);
-}
-
-static audio_io_handle_t aps_open_output_on_module(void *service __unused,
+audio_io_handle_t aps_open_output_on_module(void *service __unused,
                                                    audio_module_handle_t module,
                                                    audio_devices_t *pDevices,
                                                    uint32_t *pSamplingRate,
@@ -1500,174 +1022,42 @@
                                                    audio_channel_mask_t *pChannelMask,
                                                    uint32_t *pLatencyMs,
                                                    audio_output_flags_t flags,
-                                                   const audio_offload_info_t *offloadInfo)
-{
-    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
-    if (af == 0) {
-        ALOGW("%s: could not get AudioFlinger", __func__);
-        return 0;
-    }
-    return af->openOutput(module, pDevices, pSamplingRate, pFormat, pChannelMask,
-                          pLatencyMs, flags, offloadInfo);
-}
-
-static audio_io_handle_t aps_open_dup_output(void *service __unused,
+                                                   const audio_offload_info_t *offloadInfo);
+audio_io_handle_t aps_open_dup_output(void *service __unused,
                                                  audio_io_handle_t output1,
-                                                 audio_io_handle_t output2)
-{
-    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
-    if (af == 0) {
-        ALOGW("%s: could not get AudioFlinger", __func__);
-        return 0;
-    }
-    return af->openDuplicateOutput(output1, output2);
-}
-
-static int aps_close_output(void *service __unused, audio_io_handle_t output)
-{
-    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
-    if (af == 0) {
-        return PERMISSION_DENIED;
-    }
-
-    return af->closeOutput(output);
-}
-
-static int aps_suspend_output(void *service __unused, audio_io_handle_t output)
-{
-    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
-    if (af == 0) {
-        ALOGW("%s: could not get AudioFlinger", __func__);
-        return PERMISSION_DENIED;
-    }
-
-    return af->suspendOutput(output);
-}
-
-static int aps_restore_output(void *service __unused, audio_io_handle_t output)
-{
-    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
-    if (af == 0) {
-        ALOGW("%s: could not get AudioFlinger", __func__);
-        return PERMISSION_DENIED;
-    }
-
-    return af->restoreOutput(output);
-}
-
-// deprecated: replaced by aps_open_input_on_module(), and acoustics parameter is ignored
-static audio_io_handle_t aps_open_input(void *service __unused,
+                                                 audio_io_handle_t output2);
+int aps_close_output(void *service __unused, audio_io_handle_t output);
+int aps_suspend_output(void *service __unused, audio_io_handle_t output);
+int aps_restore_output(void *service __unused, audio_io_handle_t output);
+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 __unused)
-{
-    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
-    if (af == 0) {
-        ALOGW("%s: could not get AudioFlinger", __func__);
-        return 0;
-    }
-
-    return af->openInput((audio_module_handle_t)0, pDevices, pSamplingRate, pFormat, pChannelMask);
-}
-
-static audio_io_handle_t aps_open_input_on_module(void *service __unused,
+                                        audio_in_acoustics_t acoustics __unused);
+audio_io_handle_t aps_open_input_on_module(void *service __unused,
                                                   audio_module_handle_t module,
                                                   audio_devices_t *pDevices,
                                                   uint32_t *pSamplingRate,
                                                   audio_format_t *pFormat,
-                                                  audio_channel_mask_t *pChannelMask)
-{
-    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
-    if (af == 0) {
-        ALOGW("%s: could not get AudioFlinger", __func__);
-        return 0;
-    }
-
-    return af->openInput(module, pDevices, pSamplingRate, pFormat, pChannelMask);
-}
-
-static int aps_close_input(void *service __unused, audio_io_handle_t input)
-{
-    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
-    if (af == 0) {
-        return PERMISSION_DENIED;
-    }
-
-    return af->closeInput(input);
-}
-
-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->invalidateStream(stream);
-}
-
-static int aps_move_effects(void *service __unused, int session,
+                                                  audio_channel_mask_t *pChannelMask);
+int aps_close_input(void *service __unused, audio_io_handle_t input);
+int aps_invalidate_stream(void *service __unused, audio_stream_type_t stream);
+int aps_move_effects(void *service __unused, int session,
                                 audio_io_handle_t src_output,
-                                audio_io_handle_t dst_output)
-{
-    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
-    if (af == 0) {
-        return PERMISSION_DENIED;
-    }
-
-    return af->moveEffects(session, src_output, dst_output);
-}
-
-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));
-    return strdup(result.string());
-}
-
-static void aps_set_parameters(void *service, audio_io_handle_t io_handle,
-                                   const char *kv_pairs, int delay_ms)
-{
-    AudioPolicyService *audioPolicyService = (AudioPolicyService *)service;
-
-    audioPolicyService->setParameters(io_handle, kv_pairs, delay_ms);
-}
-
-static int aps_set_stream_volume(void *service, audio_stream_type_t stream,
+                                audio_io_handle_t dst_output);
+char * aps_get_parameters(void *service __unused, audio_io_handle_t io_handle,
+                                     const char *keys);
+void aps_set_parameters(void *service, audio_io_handle_t io_handle,
+                                   const char *kv_pairs, int delay_ms);
+int aps_set_stream_volume(void *service, audio_stream_type_t stream,
                                      float volume, audio_io_handle_t output,
-                                     int delay_ms)
-{
-    AudioPolicyService *audioPolicyService = (AudioPolicyService *)service;
-
-    return audioPolicyService->setStreamVolume(stream, volume, output,
-                                               delay_ms);
-}
-
-static int aps_start_tone(void *service, audio_policy_tone_t tone,
-                              audio_stream_type_t stream)
-{
-    AudioPolicyService *audioPolicyService = (AudioPolicyService *)service;
-
-    return audioPolicyService->startTone(tone, stream);
-}
-
-static int aps_stop_tone(void *service)
-{
-    AudioPolicyService *audioPolicyService = (AudioPolicyService *)service;
-
-    return audioPolicyService->stopTone();
-}
-
-static int aps_set_voice_volume(void *service, float volume, int delay_ms)
-{
-    AudioPolicyService *audioPolicyService = (AudioPolicyService *)service;
-
-    return audioPolicyService->setVoiceVolume(volume, delay_ms);
-}
-
-}; // extern "C"
+                                     int delay_ms);
+int aps_start_tone(void *service, audio_policy_tone_t tone,
+                              audio_stream_type_t stream);
+int aps_stop_tone(void *service);
+int aps_set_voice_volume(void *service, float volume, int delay_ms);
+};
 
 namespace {
     struct audio_policy_service_ops aps_ops = {
diff --git a/services/audioflinger/AudioPolicyService.h b/services/audiopolicy/AudioPolicyService.h
similarity index 100%
rename from services/audioflinger/AudioPolicyService.h
rename to services/audiopolicy/AudioPolicyService.h
diff --git a/services/camera/libcameraservice/Android.mk b/services/camera/libcameraservice/Android.mk
index 51ba698..4e2272d 100644
--- a/services/camera/libcameraservice/Android.mk
+++ b/services/camera/libcameraservice/Android.mk
@@ -1,3 +1,17 @@
+# Copyright 2010 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.
+
 LOCAL_PATH:= $(call my-dir)
 
 #
@@ -53,11 +67,13 @@
 
 LOCAL_C_INCLUDES += \
     system/media/camera/include \
+    system/media/private/camera/include \
     external/jpeg
 
 
 LOCAL_CFLAGS += -Wall -Wextra
 
 LOCAL_MODULE:= libcameraservice
+LOCAL_32_BIT_ONLY := true
 
 include $(BUILD_SHARED_LIBRARY)
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index 79fbf76..b83c315 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -1,24 +1,24 @@
 /*
-**
-** 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.
-*/
+ * 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.
+ */
 
 #define LOG_TAG "CameraService"
 //#define LOG_NDEBUG 0
 
 #include <stdio.h>
+#include <string.h>
 #include <sys/types.h>
 #include <pthread.h>
 
@@ -37,6 +37,8 @@
 #include <utils/Errors.h>
 #include <utils/Log.h>
 #include <utils/String16.h>
+#include <utils/Trace.h>
+#include <system/camera_vendor_tags.h>
 
 #include "CameraService.h"
 #include "api1/CameraClient.h"
@@ -131,6 +133,12 @@
             mModule->set_callbacks(this);
         }
 
+        VendorTagDescriptor::clearGlobalVendorTagDescriptor();
+
+        if (mModule->common.module_api_version >= CAMERA_MODULE_API_VERSION_2_2) {
+            setUpVendorTags();
+        }
+
         CameraDeviceFactory::registerService(this);
     }
 }
@@ -142,6 +150,7 @@
         }
     }
 
+    VendorTagDescriptor::clearGlobalVendorTagDescriptor();
     gCameraService = NULL;
 }
 
@@ -270,6 +279,22 @@
     return ret;
 }
 
+status_t CameraService::getCameraVendorTagDescriptor(/*out*/sp<VendorTagDescriptor>& desc) {
+    if (!mModule) {
+        ALOGE("%s: camera hardware module doesn't exist", __FUNCTION__);
+        return -ENODEV;
+    }
+
+    if (mModule->common.module_api_version < CAMERA_MODULE_API_VERSION_2_2) {
+        // TODO: Remove this check once HAL1 shim is in place.
+        ALOGW("%s: Only HAL module version V2.2 or higher supports vendor tags", __FUNCTION__);
+        return -EOPNOTSUPP;
+    }
+
+    desc = VendorTagDescriptor::getGlobalVendorTagDescriptor();
+    return OK;
+}
+
 int CameraService::getDeviceVersion(int cameraId, int* facing) {
     struct camera_info info;
     if (mModule->get_camera_info(cameraId, &info) != OK) {
@@ -307,6 +332,44 @@
     return false;
 }
 
+bool CameraService::setUpVendorTags() {
+    vendor_tag_ops_t vOps = vendor_tag_ops_t();
+
+    // Check if vendor operations have been implemented
+    if (mModule->get_vendor_tag_ops == NULL) {
+        ALOGI("%s: No vendor tags defined for this device.", __FUNCTION__);
+        return false;
+    }
+
+    ATRACE_BEGIN("camera3->get_metadata_vendor_tag_ops");
+    mModule->get_vendor_tag_ops(&vOps);
+    ATRACE_END();
+
+    // Ensure all vendor operations are present
+    if (vOps.get_tag_count == NULL || vOps.get_all_tags == NULL ||
+            vOps.get_section_name == NULL || vOps.get_tag_name == NULL ||
+            vOps.get_tag_type == NULL) {
+        ALOGE("%s: Vendor tag operations not fully defined. Ignoring definitions."
+               , __FUNCTION__);
+        return false;
+    }
+
+    // Read all vendor tag definitions into a descriptor
+    sp<VendorTagDescriptor> desc;
+    status_t res;
+    if ((res = VendorTagDescriptor::createDescriptorFromOps(&vOps, /*out*/desc))
+            != OK) {
+        ALOGE("%s: Could not generate descriptor from vendor tag operations,"
+              "received error %s (%d). Camera clients will not be able to use"
+              "vendor tags", __FUNCTION__, strerror(res), res);
+        return false;
+    }
+
+    // Set the global descriptor to use with camera metadata
+    VendorTagDescriptor::setAsGlobalVendorTagDescriptor(desc);
+    return true;
+}
+
 status_t CameraService::validateConnect(int cameraId,
                                     /*inout*/
                                     int& clientUid) const {
@@ -656,6 +719,11 @@
                                 const sp<ICameraServiceListener>& listener) {
     ALOGV("%s: Add listener %p", __FUNCTION__, listener.get());
 
+    if (listener == 0) {
+        ALOGE("%s: Listener must not be null", __FUNCTION__);
+        return BAD_VALUE;
+    }
+
     Mutex::Autolock lock(mServiceLock);
 
     Vector<sp<ICameraServiceListener> >::iterator it, end;
@@ -684,6 +752,11 @@
                                 const sp<ICameraServiceListener>& listener) {
     ALOGV("%s: Remove listener %p", __FUNCTION__, listener.get());
 
+    if (listener == 0) {
+        ALOGE("%s: Listener must not be null", __FUNCTION__);
+        return BAD_VALUE;
+    }
+
     Mutex::Autolock lock(mServiceLock);
 
     Vector<sp<ICameraServiceListener> >::iterator it;
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index ad6a582..8853e48 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -1,19 +1,18 @@
 /*
-**
-** 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.
-*/
+ * 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 ANDROID_SERVERS_CAMERA_CAMERASERVICE_H
 #define ANDROID_SERVERS_CAMERA_CAMERASERVICE_H
@@ -31,6 +30,7 @@
 #include <camera/IProCameraCallbacks.h>
 #include <camera/camera2/ICameraDeviceUser.h>
 #include <camera/camera2/ICameraDeviceCallbacks.h>
+#include <camera/VendorTagDescriptor.h>
 
 #include <camera/ICameraServiceListener.h>
 
@@ -73,6 +73,7 @@
                                       struct CameraInfo* cameraInfo);
     virtual status_t    getCameraCharacteristics(int cameraId,
                                                  CameraMetadata* cameraInfo);
+    virtual status_t    getCameraVendorTagDescriptor(/*out*/ sp<VendorTagDescriptor>& desc);
 
     virtual status_t connect(const sp<ICameraClient>& cameraClient, int cameraId,
             const String16& clientPackageName, int clientUid,
@@ -387,6 +388,8 @@
     // Helpers
 
     bool                isValidCameraId(int cameraId);
+
+    bool                setUpVendorTags();
 };
 
 } // namespace android
diff --git a/services/camera/libcameraservice/device2/Camera2Device.cpp b/services/camera/libcameraservice/device2/Camera2Device.cpp
index dc97c47..f60ca98 100644
--- a/services/camera/libcameraservice/device2/Camera2Device.cpp
+++ b/services/camera/libcameraservice/device2/Camera2Device.cpp
@@ -112,20 +112,6 @@
         return res;
     }
 
-    res = device->ops->get_metadata_vendor_tag_ops(device, &mVendorTagOps);
-    if (res != OK ) {
-        ALOGE("%s: Camera %d: Unable to retrieve tag ops from device: %s (%d)",
-                __FUNCTION__, mId, strerror(-res), res);
-        device->common.close(&device->common);
-        return res;
-    }
-    res = set_camera_metadata_vendor_tag_ops(mVendorTagOps);
-    if (res != OK) {
-        ALOGE("%s: Camera %d: Unable to set tag ops: %s (%d)",
-            __FUNCTION__, mId, strerror(-res), res);
-        device->common.close(&device->common);
-        return res;
-    }
     res = device->ops->set_notify_callback(device, notificationCallback,
             NULL);
     if (res != OK) {
diff --git a/services/camera/libcameraservice/device2/Camera2Device.h b/services/camera/libcameraservice/device2/Camera2Device.h
index 1f53c56..5b91f88 100644
--- a/services/camera/libcameraservice/device2/Camera2Device.h
+++ b/services/camera/libcameraservice/device2/Camera2Device.h
@@ -78,7 +78,6 @@
     camera2_device_t *mHal2Device;
 
     CameraMetadata mDeviceInfo;
-    vendor_tag_query_ops_t *mVendorTagOps;
 
     /**
      * Queue class for both sending requests to a camera2 device, and for
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index da3e121..08e03ce 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -146,24 +146,6 @@
         return BAD_VALUE;
     }
 
-    /** Get vendor metadata tags */
-
-    mVendorTagOps.get_camera_vendor_section_name = NULL;
-
-    ATRACE_BEGIN("camera3->get_metadata_vendor_tag_ops");
-    device->ops->get_metadata_vendor_tag_ops(device, &mVendorTagOps);
-    ATRACE_END();
-
-    if (mVendorTagOps.get_camera_vendor_section_name != NULL) {
-        res = set_camera_metadata_vendor_tag_ops(&mVendorTagOps);
-        if (res != OK) {
-            SET_ERR_L("Unable to set tag ops: %s (%d)",
-                    strerror(-res), res);
-            device->common.close(&device->common);
-            return res;
-        }
-    }
-
     /** Start up status tracker thread */
     mStatusTracker = new StatusTracker(this);
     res = mStatusTracker->run(String8::format("C3Dev-%d-Status", mId).string());
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index 468f641..9007a9b 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -157,7 +157,6 @@
     camera3_device_t          *mHal3Device;
 
     CameraMetadata             mDeviceInfo;
-    vendor_tag_query_ops_t     mVendorTagOps;
 
     enum Status {
         STATUS_ERROR,
diff --git a/services/medialog/Android.mk b/services/medialog/Android.mk
index 08006c8..95f2fef 100644
--- a/services/medialog/Android.mk
+++ b/services/medialog/Android.mk
@@ -8,4 +8,6 @@
 
 LOCAL_MODULE:= libmedialogservice
 
+LOCAL_32_BIT_ONLY := true
+
 include $(BUILD_SHARED_LIBRARY)