diff --git a/camera/Android.bp b/camera/Android.bp
index 9e1efae..b6241f4 100644
--- a/camera/Android.bp
+++ b/camera/Android.bp
@@ -54,7 +54,13 @@
 cc_aconfig_library {
     name: "camera_platform_flags_c_lib",
     aconfig_declarations: "camera_platform_flags",
+}
+
+cc_aconfig_library {
+    name: "camera_platform_flags_c_lib_for_test",
+    aconfig_declarations: "camera_platform_flags",
     host_supported: true,
+    mode: "test",
 }
 
 java_aconfig_library {
diff --git a/drm/mediadrm/plugins/clearkey/aidl/Android.bp b/drm/mediadrm/plugins/clearkey/aidl/Android.bp
index 8e8e57d..cce6338 100644
--- a/drm/mediadrm/plugins/clearkey/aidl/Android.bp
+++ b/drm/mediadrm/plugins/clearkey/aidl/Android.bp
@@ -85,7 +85,7 @@
     ],
     srcs: ["Service.cpp"],
     init_rc: ["android.hardware.drm-service.clearkey.rc"],
-    vintf_fragments: ["android.hardware.drm-service.clearkey.xml"],
+    vintf_fragment_modules: ["android.hardware.drm-service.clearkey.xml_vintf"],
 }
 
 cc_binary {
@@ -97,7 +97,13 @@
     overrides: ["android.hardware.drm-service.clearkey"],
     srcs: ["ServiceLazy.cpp"],
     init_rc: ["android.hardware.drm-service-lazy.clearkey.rc"],
-    vintf_fragments: ["android.hardware.drm-service.clearkey.xml"],
+    vintf_fragment_modules: ["android.hardware.drm-service.clearkey.xml_vintf"],
+}
+
+vintf_fragment {
+    name: "android.hardware.drm-service.clearkey.xml_vintf",
+    src: "android.hardware.drm-service.clearkey.xml",
+    vendor: true,
 }
 
 cc_binary {
diff --git a/media/audio/aconfig/Android.bp b/media/audio/aconfig/Android.bp
index de8aca7..a5aeff2 100644
--- a/media/audio/aconfig/Android.bp
+++ b/media/audio/aconfig/Android.bp
@@ -50,6 +50,23 @@
 }
 
 cc_aconfig_library {
+    name: "com.android.media.audioserver-aconfig-cc-ro",
+    aconfig_declarations: "com.android.media.audioserver-aconfig",
+    defaults: ["audio-aconfig-cc-defaults"],
+    double_loadable: true,
+    host_supported: true,
+    product_available: true,
+    vendor_available: true,
+    apex_available: [
+        "//apex_available:platform",
+        "com.android.media",
+        "com.android.media.swcodec",
+    ],
+    min_sdk_version: "29",
+    mode: "force-read-only",
+}
+
+cc_aconfig_library {
     name: "com.android.media.audio-aconfig-cc",
     aconfig_declarations: "com.android.media.audio-aconfig",
     defaults: ["audio-aconfig-cc-defaults"],
diff --git a/media/codec2/hal/aidl/ComponentStore.cpp b/media/codec2/hal/aidl/ComponentStore.cpp
index 356bf72..de9332b 100644
--- a/media/codec2/hal/aidl/ComponentStore.cpp
+++ b/media/codec2/hal/aidl/ComponentStore.cpp
@@ -153,6 +153,13 @@
         mParamReflectors.push_back(paramReflector);
     }
 #endif
+    // MultiAccessUnit reflector helper is allocated once per store.
+    // All components in this store can reuse this reflector helper.
+    if (MultiAccessUnitHelper::isEnabledOnPlatform()) {
+        std::shared_ptr<C2ReflectorHelper> helper = std::make_shared<C2ReflectorHelper>();
+        mParamReflectors.push_back(helper);
+        mMultiAccessUnitReflector = helper;
+    }
 
     // Retrieve supported parameters from store
     using namespace std::placeholders;
@@ -240,11 +247,9 @@
                 // param reflectors. Currently filters work on video domain only,
                 // and the MultiAccessUnitHelper is only enabled on audio domain;
                 // thus we pass the component's param reflector, which is mParamReflectors[0].
-                std::shared_ptr<C2ReflectorHelper> multiAccessReflector(new C2ReflectorHelper());
                 multiAccessUnitIntf = std::make_shared<MultiAccessUnitInterface>(
                         c2interface,
-                        multiAccessReflector);
-                mParamReflectors.push_back(multiAccessReflector);
+                        mMultiAccessUnitReflector);
             }
         }
     }
diff --git a/media/codec2/hal/aidl/include/codec2/aidl/ComponentStore.h b/media/codec2/hal/aidl/include/codec2/aidl/ComponentStore.h
index b2158a6..de0f566 100644
--- a/media/codec2/hal/aidl/include/codec2/aidl/ComponentStore.h
+++ b/media/codec2/hal/aidl/include/codec2/aidl/ComponentStore.h
@@ -120,6 +120,9 @@
     std::shared_ptr<C2ComponentStore> mStore;
     std::vector<std::shared_ptr<C2ParamReflector>> mParamReflectors;
 
+    // Reflector helper for MultiAccessUnitHelper
+    std::shared_ptr<C2ReflectorHelper> mMultiAccessUnitReflector;
+
     std::map<C2Param::CoreIndex, std::shared_ptr<C2StructDescriptor>> mStructDescriptors;
     std::set<C2Param::CoreIndex> mUnsupportedStructDescriptors;
     std::set<C2String> mLoadedInterfaces;
diff --git a/media/codec2/hal/hidl/1.0/utils/ComponentStore.cpp b/media/codec2/hal/hidl/1.0/utils/ComponentStore.cpp
index f09e232..108ba06 100644
--- a/media/codec2/hal/hidl/1.0/utils/ComponentStore.cpp
+++ b/media/codec2/hal/hidl/1.0/utils/ComponentStore.cpp
@@ -149,6 +149,14 @@
     }
 #endif
 
+    // MultiAccessUnit reflector helper is allocated once per store.
+    // All components in this store can reuse this reflector helper.
+    if (MultiAccessUnitHelper::isEnabledOnPlatform()) {
+        std::shared_ptr<C2ReflectorHelper> helper = std::make_shared<C2ReflectorHelper>();
+        mParamReflectors.push_back(helper);
+        mMultiAccessUnitReflector = helper;
+    }
+
     // Retrieve supported parameters from store
     using namespace std::placeholders;
     mInit = mConfigurable->init(mParameterCache);
@@ -231,12 +239,9 @@
                 }
             }
             if (!isComponentSupportsLargeAudioFrame) {
-                std::shared_ptr<C2ReflectorHelper> multiAccessReflector(new C2ReflectorHelper());
                 multiAccessUnitIntf = std::make_shared<MultiAccessUnitInterface>(
                         c2interface,
-                        multiAccessReflector);
-                mParamReflectors.push_back(multiAccessReflector);
-
+                        mMultiAccessUnitReflector);
             }
         }
     }
diff --git a/media/codec2/hal/hidl/1.0/utils/include/codec2/hidl/1.0/ComponentStore.h b/media/codec2/hal/hidl/1.0/utils/include/codec2/hidl/1.0/ComponentStore.h
index 44b8ec1..847c90c 100644
--- a/media/codec2/hal/hidl/1.0/utils/include/codec2/hidl/1.0/ComponentStore.h
+++ b/media/codec2/hal/hidl/1.0/utils/include/codec2/hidl/1.0/ComponentStore.h
@@ -124,6 +124,9 @@
     std::shared_ptr<C2ComponentStore> mStore;
     std::vector<std::shared_ptr<C2ParamReflector>> mParamReflectors;
 
+    // Reflector helper for MultiAccessUnitHelper
+    std::shared_ptr<C2ReflectorHelper> mMultiAccessUnitReflector;
+
     std::map<C2Param::CoreIndex, std::shared_ptr<C2StructDescriptor>> mStructDescriptors;
     std::set<C2Param::CoreIndex> mUnsupportedStructDescriptors;
     std::set<C2String> mLoadedInterfaces;
diff --git a/media/codec2/hal/hidl/1.1/utils/ComponentStore.cpp b/media/codec2/hal/hidl/1.1/utils/ComponentStore.cpp
index 009a326..84f5d26 100644
--- a/media/codec2/hal/hidl/1.1/utils/ComponentStore.cpp
+++ b/media/codec2/hal/hidl/1.1/utils/ComponentStore.cpp
@@ -149,6 +149,14 @@
     }
 #endif
 
+    // MultiAccessUnit reflector helper is allocated once per store.
+    // All components in this store can reuse this reflector helper.
+    if (MultiAccessUnitHelper::isEnabledOnPlatform()) {
+        std::shared_ptr<C2ReflectorHelper> helper = std::make_shared<C2ReflectorHelper>();
+        mParamReflectors.push_back(helper);
+        mMultiAccessUnitReflector = helper;
+    }
+
     // Retrieve supported parameters from store
     using namespace std::placeholders;
     mInit = mConfigurable->init(mParameterCache);
@@ -230,13 +238,10 @@
                     break;
                 }
             }
-
             if (!isComponentSupportsLargeAudioFrame) {
-                std::shared_ptr<C2ReflectorHelper> multiAccessReflector(new C2ReflectorHelper());
                 multiAccessUnitIntf = std::make_shared<MultiAccessUnitInterface>(
                         c2interface,
-                        multiAccessReflector);
-                mParamReflectors.push_back(multiAccessReflector);
+                        mMultiAccessUnitReflector);
             }
         }
     }
diff --git a/media/codec2/hal/hidl/1.1/utils/include/codec2/hidl/1.1/ComponentStore.h b/media/codec2/hal/hidl/1.1/utils/include/codec2/hidl/1.1/ComponentStore.h
index 52d2945..9028149 100644
--- a/media/codec2/hal/hidl/1.1/utils/include/codec2/hidl/1.1/ComponentStore.h
+++ b/media/codec2/hal/hidl/1.1/utils/include/codec2/hidl/1.1/ComponentStore.h
@@ -132,6 +132,9 @@
     std::shared_ptr<C2ComponentStore> mStore;
     std::vector<std::shared_ptr<C2ParamReflector>> mParamReflectors;
 
+    // Reflector helper for MultiAccessUnitHelper
+    std::shared_ptr<C2ReflectorHelper> mMultiAccessUnitReflector;
+
     std::map<C2Param::CoreIndex, std::shared_ptr<C2StructDescriptor>> mStructDescriptors;
     std::set<C2Param::CoreIndex> mUnsupportedStructDescriptors;
     std::set<C2String> mLoadedInterfaces;
diff --git a/media/codec2/hal/hidl/1.2/utils/ComponentStore.cpp b/media/codec2/hal/hidl/1.2/utils/ComponentStore.cpp
index 89f71a9..5585be8 100644
--- a/media/codec2/hal/hidl/1.2/utils/ComponentStore.cpp
+++ b/media/codec2/hal/hidl/1.2/utils/ComponentStore.cpp
@@ -149,6 +149,14 @@
     }
 #endif
 
+    // MultiAccessUnit reflector helper is allocated once per store.
+    // All components in this store can reuse this reflector helper.
+    if (MultiAccessUnitHelper::isEnabledOnPlatform()) {
+        std::shared_ptr<C2ReflectorHelper> helper = std::make_shared<C2ReflectorHelper>();
+        mParamReflectors.push_back(helper);
+        mMultiAccessUnitReflector = helper;
+    }
+
     // Retrieve supported parameters from store
     using namespace std::placeholders;
     mInit = mConfigurable->init(mParameterCache);
@@ -231,11 +239,9 @@
                 }
             }
             if (!isComponentSupportsLargeAudioFrame) {
-                std::shared_ptr<C2ReflectorHelper> multiAccessReflector(new C2ReflectorHelper());
                 multiAccessUnitIntf = std::make_shared<MultiAccessUnitInterface>(
                         c2interface,
-                        multiAccessReflector);
-                mParamReflectors.push_back(multiAccessReflector);
+                        mMultiAccessUnitReflector);
             }
         }
     }
diff --git a/media/codec2/hal/hidl/1.2/utils/include/codec2/hidl/1.2/ComponentStore.h b/media/codec2/hal/hidl/1.2/utils/include/codec2/hidl/1.2/ComponentStore.h
index 1b209e2..4fd260b 100644
--- a/media/codec2/hal/hidl/1.2/utils/include/codec2/hidl/1.2/ComponentStore.h
+++ b/media/codec2/hal/hidl/1.2/utils/include/codec2/hidl/1.2/ComponentStore.h
@@ -139,6 +139,9 @@
     std::shared_ptr<C2ComponentStore> mStore;
     std::vector<std::shared_ptr<C2ParamReflector>> mParamReflectors;
 
+    // Reflector helper for MultiAccessUnitHelper
+    std::shared_ptr<C2ReflectorHelper> mMultiAccessUnitReflector;
+
     std::map<C2Param::CoreIndex, std::shared_ptr<C2StructDescriptor>> mStructDescriptors;
     std::set<C2Param::CoreIndex> mUnsupportedStructDescriptors;
     std::set<C2String> mLoadedInterfaces;
diff --git a/media/codec2/hal/plugin/samples/SampleFilterPlugin.cpp b/media/codec2/hal/plugin/samples/SampleFilterPlugin.cpp
index 47412b7..34872f0 100644
--- a/media/codec2/hal/plugin/samples/SampleFilterPlugin.cpp
+++ b/media/codec2/hal/plugin/samples/SampleFilterPlugin.cpp
@@ -856,21 +856,31 @@
     C2String getName() const override { return "android.sample.filter-plugin-store"; }
     c2_status_t createComponent(
             C2String name, std::shared_ptr<C2Component>* const component) override {
-        if (mFactories.count(name) == 0) {
+        auto it = std::find_if(
+                mFactories.begin(), mFactories.end(),
+                [&name](const std::unique_ptr<ComponentFactory> &factory) {
+                    return name == factory->getTraits()->name;
+                });
+        if (it == mFactories.end()) {
             return C2_BAD_VALUE;
         }
-        return mFactories.at(name)->createComponent(++mNodeId, component);
+        return (*it)->createComponent(++mNodeId, component);
     }
     c2_status_t createInterface(
             C2String name, std::shared_ptr<C2ComponentInterface>* const interface) override {
-        if (mFactories.count(name) == 0) {
+        auto it = std::find_if(
+                mFactories.begin(), mFactories.end(),
+                [&name](const std::unique_ptr<ComponentFactory> &factory) {
+                    return name == factory->getTraits()->name;
+                });
+        if (it == mFactories.end()) {
             return C2_BAD_VALUE;
         }
-        return mFactories.at(name)->createInterface(++mNodeId, interface);
+        return (*it)->createInterface(++mNodeId, interface);
     }
     std::vector<std::shared_ptr<const C2Component::Traits>> listComponents() override {
         std::vector<std::shared_ptr<const C2Component::Traits>> ret;
-        for (const auto &[name, factory] : mFactories) {
+        for (const auto &factory : mFactories) {
             ret.push_back(factory->getTraits());
         }
         return ret;
@@ -951,20 +961,18 @@
 
     template <class T>
     static void AddFactory(
-            std::map<C2String, std::unique_ptr<ComponentFactory>> *factories,
+            std::vector<std::unique_ptr<ComponentFactory>> *factories,
             const std::shared_ptr<C2ReflectorHelper> &reflector) {
         std::shared_ptr<C2ComponentInterface> intf{new typename T::Interface(0, reflector)};
         std::shared_ptr<C2Component::Traits> traits(new (std::nothrow) C2Component::Traits);
         CHECK(C2InterfaceUtils::FillTraitsFromInterface(traits.get(), intf))
                 << "Failed to fill traits from interface";
-        factories->emplace(
-                traits->name,
-                new ComponentFactoryImpl<T>(traits, reflector));
+        factories->emplace_back(new ComponentFactoryImpl<T>(traits, reflector));
     }
 
-    static std::map<C2String, std::unique_ptr<ComponentFactory>> CreateFactories(
+    static std::vector<std::unique_ptr<ComponentFactory>> CreateFactories(
             const std::shared_ptr<C2ReflectorHelper> &reflector) {
-        std::map<C2String, std::unique_ptr<ComponentFactory>> factories;
+        std::vector<std::unique_ptr<ComponentFactory>> factories;
         AddFactory<SampleToneMappingFilter>(&factories, reflector);
         return factories;
     }
@@ -977,7 +985,7 @@
         }
     } mIntf;
 
-    const std::map<C2String, std::unique_ptr<ComponentFactory>> mFactories;
+    const std::vector<std::unique_ptr<ComponentFactory>> mFactories;
 
     std::atomic_int32_t mNodeId{0};
 };
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.cpp b/media/codec2/sfplugin/CCodecBufferChannel.cpp
index d829523..3ef2f84 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -228,17 +228,23 @@
 status_t CCodecBufferChannel::setInputSurface(
         const std::shared_ptr<InputSurfaceWrapper> &surface) {
     ALOGV("[%s] setInputSurface", mName);
-    Mutexed<std::shared_ptr<InputSurfaceWrapper>>::Locked inputSurface(mInputSurface);
-    *inputSurface = surface;
-    return (*inputSurface)->connect(mComponent);
+    if (!surface) {
+        ALOGE("[%s] setInputSurface: surface must not be null", mName);
+        return BAD_VALUE;
+    }
+    Mutexed<InputSurface>::Locked inputSurface(mInputSurface);
+    inputSurface->numProcessingBuffersBalance = 0;
+    inputSurface->surface = surface;
+    mHasInputSurface = true;
+    return inputSurface->surface->connect(mComponent);
 }
 
 status_t CCodecBufferChannel::signalEndOfInputStream() {
-    Mutexed<std::shared_ptr<InputSurfaceWrapper>>::Locked inputSurface(mInputSurface);
-    if ((*inputSurface) == nullptr) {
+    Mutexed<InputSurface>::Locked inputSurface(mInputSurface);
+    if (inputSurface->surface == nullptr) {
         return INVALID_OPERATION;
     }
-    return (*inputSurface)->signalEndOfInputStream();
+    return inputSurface->surface->signalEndOfInputStream();
 }
 
 status_t CCodecBufferChannel::queueInputBufferInternal(
@@ -1063,19 +1069,36 @@
     if (mInputMetEos) {
         return;
     }
-    {
+    int64_t numOutputSlots = 0;
+    bool outputFull = [this, &numOutputSlots]() {
         Mutexed<Output>::Locked output(mOutput);
-        if (!output->buffers ||
-                output->buffers->hasPending() ||
+        if (!output->buffers) {
+            ALOGV("[%s] feedInputBufferIfAvailableInternal: "
+                  "return because output buffers are null", mName);
+            return true;
+        }
+        numOutputSlots = int64_t(output->numSlots);
+        if (output->buffers->hasPending() ||
                 (!output->bounded && output->buffers->numActiveSlots() >= output->numSlots)) {
-            return;
+            ALOGV("[%s] feedInputBufferIfAvailableInternal: "
+                  "return because there are no room for more output buffers", mName);
+            return true;
+        }
+        return false;
+    }();
+    if (android::media::codec::provider_->input_surface_throttle()) {
+        Mutexed<InputSurface>::Locked inputSurface(mInputSurface);
+        if (inputSurface->surface) {
+            if (inputSurface->numProcessingBuffersBalance <= numOutputSlots) {
+                ++inputSurface->numProcessingBuffersBalance;
+                ALOGV("[%s] feedInputBufferIfAvailableInternal: numProcessingBuffersBalance = %lld",
+                      mName, static_cast<long long>(inputSurface->numProcessingBuffersBalance));
+                inputSurface->surface->onInputBufferEmptied();
+            }
         }
     }
-    if (android::media::codec::provider_->input_surface_throttle()) {
-        Mutexed<std::shared_ptr<InputSurfaceWrapper>>::Locked inputSurface(mInputSurface);
-        if ((*inputSurface) != nullptr) {
-            (*inputSurface)->onInputBufferEmptied();
-        }
+    if (outputFull) {
+        return;
     }
     size_t numActiveSlots = 0;
     while (!mPipelineWatcher.lock()->pipelineFull()) {
@@ -1704,7 +1727,7 @@
                 && (hasCryptoOrDescrambler() || conforming)) {
             input->buffers.reset(new SlotInputBuffers(mName));
         } else if (graphic) {
-            if (mInputSurface.lock()->get()) {
+            if (mHasInputSurface) {
                 input->buffers.reset(new DummyInputBuffers(mName));
             } else if (mMetaMode == MODE_ANW) {
                 input->buffers.reset(new GraphicMetadataInputBuffers(mName));
@@ -1987,7 +2010,7 @@
 
 status_t CCodecBufferChannel::prepareInitialInputBuffers(
         std::map<size_t, sp<MediaCodecBuffer>> *clientInputBuffers, bool retry) {
-    if (mInputSurface.lock()->get()) {
+    if (mHasInputSurface) {
         return OK;
     }
 
@@ -2113,9 +2136,13 @@
 
 void CCodecBufferChannel::reset() {
     stop();
-    mInputSurface.lock()->reset();
     mPipelineWatcher.lock()->flush();
     {
+        mHasInputSurface = false;
+        Mutexed<InputSurface>::Locked inputSurface(mInputSurface);
+        inputSurface->surface.reset();
+    }
+    {
         Mutexed<Input>::Locked input(mInput);
         input->buffers.reset(new DummyInputBuffers(""));
         input->extraBuffers.flush();
@@ -2208,9 +2235,6 @@
 
 void CCodecBufferChannel::onInputBufferDone(
         uint64_t frameIndex, size_t arrayIndex) {
-    if (mInputSurface.lock()->get()) {
-        return;
-    }
     std::shared_ptr<C2Buffer> buffer =
             mPipelineWatcher.lock()->onInputBufferReleased(frameIndex, arrayIndex);
     bool newInputSlotAvailable = false;
@@ -2265,8 +2289,7 @@
         notifyClient = false;
     }
 
-    bool hasInputSurface = (mInputSurface.lock()->get() != nullptr);
-    if (!hasInputSurface && (work->worklets.size() != 1u
+    if (!mHasInputSurface && (work->worklets.size() != 1u
             || !work->worklets.front()
             || !(work->worklets.front()->output.flags &
                  C2FrameData::FLAG_INCOMPLETE))) {
@@ -2475,7 +2498,7 @@
     c2_cntr64_t timestamp =
         worklet->output.ordinal.timestamp + work->input.ordinal.customOrdinal
                 - work->input.ordinal.timestamp;
-    if (hasInputSurface) {
+    if (mHasInputSurface) {
         // When using input surface we need to restore the original input timestamp.
         timestamp = work->input.ordinal.customOrdinal;
     }
@@ -2605,8 +2628,6 @@
         switch (action) {
         case OutputBuffers::SKIP:
             return;
-        case OutputBuffers::DISCARD:
-            break;
         case OutputBuffers::NOTIFY_CLIENT:
         {
             // TRICKY: we want popped buffers reported in order, so sending
@@ -2625,8 +2646,8 @@
                                 bufferMetadata->m.values[nMeta];
                         flag = convertFlags(bufferMetadataStruct.flags, false);
                         accessUnitInfos.emplace_back(flag,
-                                static_cast<size_t>(bufferMetadataStruct.size),
-                                static_cast<size_t>(bufferMetadataStruct.timestamp));
+                                bufferMetadataStruct.size,
+                                bufferMetadataStruct.timestamp);
                     }
                     sp<WrapperObject<std::vector<AccessUnitInfo>>> obj{
                         new WrapperObject<std::vector<AccessUnitInfo>>{accessUnitInfos}};
@@ -2634,6 +2655,15 @@
                 }
             }
             mCallback->onOutputBufferAvailable(index, outBuffer);
+            [[fallthrough]];
+        }
+        case OutputBuffers::DISCARD: {
+            if (mHasInputSurface && android::media::codec::provider_->input_surface_throttle()) {
+                Mutexed<InputSurface>::Locked inputSurface(mInputSurface);
+                --inputSurface->numProcessingBuffersBalance;
+                ALOGV("[%s] onWorkDone: numProcessingBuffersBalance = %lld",
+                        mName, static_cast<long long>(inputSurface->numProcessingBuffersBalance));
+            }
             break;
         }
         case OutputBuffers::REALLOCATE:
@@ -2802,7 +2832,7 @@
 }
 
 void CCodecBufferChannel::setInfoBuffer(const std::shared_ptr<C2InfoBuffer> &buffer) {
-    if (mInputSurface.lock()->get() == nullptr) {
+    if (!mHasInputSurface) {
         mInfoBuffers.push_back(buffer);
     } else {
         std::list<std::unique_ptr<C2Work>> items;
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.h b/media/codec2/sfplugin/CCodecBufferChannel.h
index f5e268a..4d296fd 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.h
+++ b/media/codec2/sfplugin/CCodecBufferChannel.h
@@ -391,7 +391,21 @@
     };
     Mutexed<BlockPools> mBlockPools;
 
-    Mutexed<std::shared_ptr<InputSurfaceWrapper>> mInputSurface;
+    std::atomic_bool mHasInputSurface;
+    struct InputSurface {
+        std::shared_ptr<InputSurfaceWrapper> surface;
+        // This variable tracks the number of buffers processing
+        // in the input surface and codec by counting the # of buffers to
+        // be filled in and queued from the input surface and the # of
+        // buffers generated from the codec.
+        //
+        // Note that this variable can go below 0, because it does not take
+        // account the number of buffers initially in the buffer queue at
+        // start. This is okay, as we only track how many more we allow
+        // from the initial state.
+        int64_t numProcessingBuffersBalance;
+    };
+    Mutexed<InputSurface> mInputSurface;
 
     MetaMode mMetaMode;
 
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index b1517bb..61204ae 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -158,6 +158,7 @@
         "framework-permission-aidl-cpp",
         "libbinder",
         "libmediametrics",
+        "libmediautils",
         "spatializer-aidl-cpp",
     ],
 
diff --git a/media/libaudioclient/TrackPlayerBase.cpp b/media/libaudioclient/TrackPlayerBase.cpp
index 4fc1c44..bc38251 100644
--- a/media/libaudioclient/TrackPlayerBase.cpp
+++ b/media/libaudioclient/TrackPlayerBase.cpp
@@ -38,12 +38,12 @@
                            player_type_t playerType, audio_usage_t usage,
                            audio_session_t sessionId) {
     PlayerBase::init(playerType, usage, sessionId);
-    mAudioTrack = pat;
-    if (mAudioTrack != 0) {
+    mAudioTrack.store(pat);
+    if (pat != 0) {
         mCallbackHandle = callback;
         mSelfAudioDeviceCallback = new SelfAudioDeviceCallback(*this);
-        mAudioTrack->addAudioDeviceCallback(mSelfAudioDeviceCallback);
-        mAudioTrack->setPlayerIId(mPIId); // set in PlayerBase::init().
+        pat->addAudioDeviceCallback(mSelfAudioDeviceCallback);
+        pat->setPlayerIId(mPIId);  // set in PlayerBase::init().
     }
 }
 
@@ -65,12 +65,15 @@
 }
 
 void TrackPlayerBase::doDestroy() {
-    if (mAudioTrack != 0) {
-        mAudioTrack->stop();
-        mAudioTrack->removeAudioDeviceCallback(mSelfAudioDeviceCallback);
+    sp<AudioTrack> audioTrack = getAudioTrack();
+
+    // Note that there may still be another reference in post-unlock phase of SetPlayState
+    clearAudioTrack();
+
+    if (audioTrack != 0) {
+        audioTrack->stop();
+        audioTrack->removeAudioDeviceCallback(mSelfAudioDeviceCallback);
         mSelfAudioDeviceCallback.clear();
-        // Note that there may still be another reference in post-unlock phase of SetPlayState
-        mAudioTrack.clear();
     }
 }
 
@@ -87,16 +90,16 @@
 // Implementation of IPlayer
 status_t TrackPlayerBase::playerStart() {
     status_t status = NO_INIT;
-    if (mAudioTrack != 0) {
-        status = mAudioTrack->start();
+    if (sp<AudioTrack> audioTrack = getAudioTrack(); audioTrack != 0) {
+        status = audioTrack->start();
     }
     return status;
 }
 
 status_t TrackPlayerBase::playerPause() {
     status_t status = NO_INIT;
-    if (mAudioTrack != 0) {
-        mAudioTrack->pause();
+    if (sp<AudioTrack> audioTrack = getAudioTrack(); audioTrack != 0) {
+        audioTrack->pause();
         status = NO_ERROR;
     }
     return status;
@@ -105,8 +108,8 @@
 
 status_t TrackPlayerBase::playerStop() {
     status_t status = NO_INIT;
-    if (mAudioTrack != 0) {
-        mAudioTrack->stop();
+    if (sp<AudioTrack> audioTrack = getAudioTrack(); audioTrack != 0) {
+        audioTrack->stop();
         status = NO_ERROR;
     }
     return status;
@@ -118,10 +121,10 @@
 
 status_t TrackPlayerBase::doSetVolume() {
     status_t status = NO_INIT;
-    if (mAudioTrack != 0) {
+    if (sp<AudioTrack> audioTrack = getAudioTrack(); audioTrack != 0) {
         float tl = mPlayerVolumeL * mPanMultiplierL * mVolumeMultiplierL;
         float tr = mPlayerVolumeR * mPanMultiplierR * mVolumeMultiplierR;
-        mAudioTrack->setVolume(tl, tr);
+        audioTrack->setVolume(tl, tr);
         status = NO_ERROR;
     }
     return status;
@@ -140,10 +143,9 @@
     if (s != OK) {
         return binderStatusFromStatusT(s);
     }
-
-    if (mAudioTrack != 0) {
+    if (sp<AudioTrack> audioTrack = getAudioTrack(); audioTrack != 0) {
         ALOGD("TrackPlayerBase::applyVolumeShaper() from IPlayer");
-        VolumeShaper::Status status = mAudioTrack->applyVolumeShaper(spConfiguration, spOperation);
+        VolumeShaper::Status status = audioTrack->applyVolumeShaper(spConfiguration, spOperation);
         if (status < 0) { // a non-negative value is the volume shaper id.
             ALOGE("TrackPlayerBase::applyVolumeShaper() failed with status %d", status);
         }
diff --git a/media/libaudioclient/include/media/TrackPlayerBase.h b/media/libaudioclient/include/media/TrackPlayerBase.h
index fe88116..8df9ff8 100644
--- a/media/libaudioclient/include/media/TrackPlayerBase.h
+++ b/media/libaudioclient/include/media/TrackPlayerBase.h
@@ -19,6 +19,7 @@
 
 #include <media/AudioTrack.h>
 #include <media/PlayerBase.h>
+#include <mediautils/Synchronization.h>
 
 namespace android {
 
@@ -37,10 +38,11 @@
             const media::VolumeShaperConfiguration& configuration,
             const media::VolumeShaperOperation& operation);
 
-    //FIXME move to protected field, so far made public to minimize changes to AudioTrack logic
-    sp<AudioTrack> mAudioTrack;
+    sp<AudioTrack> getAudioTrack() { return mAudioTrack.load(); }
 
-            void setPlayerVolume(float vl, float vr);
+    void clearAudioTrack() { mAudioTrack.store(nullptr); }
+
+    void setPlayerVolume(float vl, float vr);
 
 protected:
 
@@ -68,6 +70,7 @@
     float mPlayerVolumeL, mPlayerVolumeR;
     sp<AudioTrack::IAudioTrackCallback> mCallbackHandle;
     sp<SelfAudioDeviceCallback> mSelfAudioDeviceCallback;
+    mediautils::atomic_sp<AudioTrack> mAudioTrack;
 };
 
 } // namespace android
diff --git a/media/libaudioclient/tests/trackplayerbase_tests.cpp b/media/libaudioclient/tests/trackplayerbase_tests.cpp
index 7317bf0..a4dba9b 100644
--- a/media/libaudioclient/tests/trackplayerbase_tests.cpp
+++ b/media/libaudioclient/tests/trackplayerbase_tests.cpp
@@ -54,7 +54,7 @@
         mPlayer = new TrackPlayer();
         mPlayer->init(track.get(), mPlayer, PLAYER_TYPE_AAUDIO, AUDIO_USAGE_MEDIA,
                       AUDIO_SESSION_NONE);
-        sp<AudioTrack> playerTrack = mPlayer->mAudioTrack;
+        sp<AudioTrack> playerTrack = mPlayer->getAudioTrack();
         ASSERT_EQ(playerTrack->initCheck(), NO_ERROR);
 
         mBufferSize = mFrameCount * playerTrack->frameSize();
@@ -74,7 +74,7 @@
 
     void playBuffer() {
         bool blocking = true;
-        ssize_t nbytes = mPlayer->mAudioTrack->write(mBuffer.data(), mBufferSize, blocking);
+        ssize_t nbytes = mPlayer->getAudioTrack()->write(mBuffer.data(), mBufferSize, blocking);
         EXPECT_EQ(nbytes, mBufferSize) << "Did not write all data in blocking mode";
     }
 
diff --git a/media/libaudiohal/impl/DeviceHalAidl.cpp b/media/libaudiohal/impl/DeviceHalAidl.cpp
index 3cc923d..629cd7c 100644
--- a/media/libaudiohal/impl/DeviceHalAidl.cpp
+++ b/media/libaudiohal/impl/DeviceHalAidl.cpp
@@ -22,6 +22,7 @@
 #include <aidl/android/hardware/audio/core/BnStreamCallback.h>
 #include <aidl/android/hardware/audio/core/BnStreamOutEventCallback.h>
 #include <aidl/android/hardware/audio/core/StreamDescriptor.h>
+#include <android/binder_ibinder_platform.h>
 #include <error/expected_utils.h>
 #include <media/AidlConversionCppNdk.h>
 #include <media/AidlConversionNdk.h>
@@ -29,6 +30,8 @@
 #include <media/AidlConversionUtil.h>
 #include <mediautils/TimeCheck.h>
 #include <system/audio.h>
+#include <system/thread_defs.h>
+
 #include <Utils.h>
 #include <utils/Log.h>
 
@@ -504,8 +507,15 @@
     std::shared_ptr<OutputStreamCallbackAidl> streamCb;
     if (isOffload) {
         streamCb = ndk::SharedRefBase::make<OutputStreamCallbackAidl>(this);
+        ndk::SpAIBinder binder = streamCb->asBinder();
+        AIBinder_setMinSchedulerPolicy(binder.get(), SCHED_NORMAL, ANDROID_PRIORITY_AUDIO);
+        AIBinder_setInheritRt(binder.get(), true);
     }
     auto eventCb = ndk::SharedRefBase::make<OutputStreamEventCallbackAidl>(this);
+    ndk::SpAIBinder binder = eventCb->asBinder();
+    AIBinder_setMinSchedulerPolicy(binder.get(), SCHED_NORMAL, ANDROID_PRIORITY_AUDIO);
+    AIBinder_setInheritRt(binder.get(), true);
+
     if (isOffload || isHwAvSync) {
         args.offloadInfo = aidlConfig.offloadInfo;
     }
diff --git a/media/libaudiohal/impl/EffectConversionHelperAidl.cpp b/media/libaudiohal/impl/EffectConversionHelperAidl.cpp
index a13903b..f719d97 100644
--- a/media/libaudiohal/impl/EffectConversionHelperAidl.cpp
+++ b/media/libaudiohal/impl/EffectConversionHelperAidl.cpp
@@ -532,5 +532,13 @@
                            AudioChannelLayout::LAYOUT_HAPTIC_AB /* mask */);
 }
 
+size_t EffectConversionHelperAidl::getInputChannelCount() const {
+    return getChannelCount(mCommon.input.base.channelMask);
+}
+
+size_t EffectConversionHelperAidl::getOutputChannelCount() const {
+    return getChannelCount(mCommon.output.base.channelMask);
+}
+
 }  // namespace effect
 }  // namespace android
diff --git a/media/libaudiohal/impl/EffectConversionHelperAidl.h b/media/libaudiohal/impl/EffectConversionHelperAidl.h
index 50b47a9..e9e9fc2 100644
--- a/media/libaudiohal/impl/EffectConversionHelperAidl.h
+++ b/media/libaudiohal/impl/EffectConversionHelperAidl.h
@@ -51,6 +51,8 @@
 
     size_t getAudioChannelCount() const;
     size_t getHapticChannelCount() const;
+    size_t getInputChannelCount() const;
+    size_t getOutputChannelCount() const;
 
     uint8_t mOutputAccessMode = EFFECT_BUFFER_ACCESS_WRITE;
 
diff --git a/media/libaudiohal/impl/EffectHalAidl.cpp b/media/libaudiohal/impl/EffectHalAidl.cpp
index ea4dbf6..9fdde49 100644
--- a/media/libaudiohal/impl/EffectHalAidl.cpp
+++ b/media/libaudiohal/impl/EffectHalAidl.cpp
@@ -75,7 +75,14 @@
       mEffect(effect),
       mSessionId(sessionId),
       mIoId(ioId),
-      mIsProxyEffect(isProxyEffect) {
+      mIsProxyEffect(isProxyEffect),
+      mHalVersion([factory]() {
+          int version = 0;
+          // use factory HAL version because effect can be an EffectProxy instance
+          return factory->getInterfaceVersion(&version).isOk() ? version : 0;
+      }()),
+      mEventFlagDataMqNotEmpty(mHalVersion >= kReopenSupportedVersion ? kEventFlagDataMqNotEmpty
+                                                                      : kEventFlagNotEmpty) {
     assert(mFactory != nullptr);
     assert(mEffect != nullptr);
     createAidlConversion(effect, sessionId, ioId, desc);
@@ -159,6 +166,7 @@
         mConversion = std::make_unique<android::effect::AidlConversionVendorExtension>(
                 effect, sessionId, ioId, desc, mIsProxyEffect);
     }
+    mEffectName = mConversion->getDescriptor().common.name;
     return OK;
 }
 
@@ -174,100 +182,153 @@
 
 // write to input FMQ here, wait for statusMQ STATUS_OK, and read from output FMQ
 status_t EffectHalAidl::process() {
-    const std::string effectName = mConversion->getDescriptor().common.name;
     State state = State::INIT;
     if (mConversion->isBypassing() || !mEffect->getState(&state).isOk() ||
         state != State::PROCESSING) {
-        ALOGI("%s skipping %s process because it's %s", __func__, effectName.c_str(),
+        ALOGI("%s skipping process because it's %s", mEffectName.c_str(),
               mConversion->isBypassing()
                       ? "bypassing"
                       : aidl::android::hardware::audio::effect::toString(state).c_str());
         return -ENODATA;
     }
 
-    // check if the DataMq needs any update, timeout at 1ns to avoid being blocked
-    auto efGroup = mConversion->getEventFlagGroup();
+    const std::shared_ptr<android::hardware::EventFlag> efGroup = mConversion->getEventFlagGroup();
     if (!efGroup) {
-        ALOGE("%s invalid efGroup", __func__);
+        ALOGE("%s invalid efGroup", mEffectName.c_str());
         return INVALID_OPERATION;
     }
 
-    // use IFactory HAL version because IEffect can be an EffectProxy instance
-    static const int halVersion = [&]() {
-        int version = 0;
-        return mFactory->getInterfaceVersion(&version).isOk() ? version : 0;
-    }();
+    // reopen if halVersion >= kReopenSupportedVersion and receive kEventFlagDataMqUpdate
+    RETURN_STATUS_IF_ERROR(maybeReopen(efGroup));
+    const size_t samplesWritten = writeToHalInputFmqAndSignal(efGroup);
+    if (0 == samplesWritten) {
+        return INVALID_OPERATION;
+    }
 
-    if (uint32_t efState = 0; halVersion >= kReopenSupportedVersion &&
-                              ::android::OK == efGroup->wait(kEventFlagDataMqUpdate, &efState,
+    RETURN_STATUS_IF_ERROR(waitHalStatusFmq(samplesWritten));
+    RETURN_STATUS_IF_ERROR(readFromHalOutputFmq(samplesWritten));
+    return OK;
+}
+
+status_t EffectHalAidl::maybeReopen(
+        const std::shared_ptr<android::hardware::EventFlag>& efGroup) const {
+    if (mHalVersion < kReopenSupportedVersion) {
+        return OK;
+    }
+
+    // check if the DataMq needs any update, timeout at 1ns to avoid being blocked
+    if (uint32_t efState = 0; ::android::OK == efGroup->wait(kEventFlagDataMqUpdate, &efState,
                                                              1 /* ns */, true /* retry */) &&
                               efState & kEventFlagDataMqUpdate) {
-        ALOGD("%s %s V%d receive dataMQUpdate eventFlag from HAL", __func__, effectName.c_str(),
-              halVersion);
-
-        mConversion->reopen();
+        ALOGD("%s V%d receive dataMQUpdate eventFlag from HAL", mEffectName.c_str(), mHalVersion);
+        return mConversion->reopen();
     }
-    auto statusQ = mConversion->getStatusMQ();
-    auto inputQ = mConversion->getInputMQ();
-    auto outputQ = mConversion->getOutputMQ();
-    if (!statusQ || !statusQ->isValid() || !inputQ || !inputQ->isValid() || !outputQ ||
-        !outputQ->isValid()) {
-        ALOGE("%s invalid FMQ [Status %d I %d O %d]", __func__, statusQ ? statusQ->isValid() : 0,
-              inputQ ? inputQ->isValid() : 0, outputQ ? outputQ->isValid() : 0);
-        return INVALID_OPERATION;
+    return OK;
+}
+
+size_t EffectHalAidl::writeToHalInputFmqAndSignal(
+        const std::shared_ptr<android::hardware::EventFlag>& efGroup) const {
+    const auto inputQ = mConversion->getInputMQ();
+    if (!inputQ || !inputQ->isValid()) {
+        ALOGE("%s invalid input FMQ", mEffectName.c_str());
+        return 0;
     }
 
-    size_t available = inputQ->availableToWrite();
-    const size_t floatsToWrite = std::min(available, mInBuffer->getSize() / sizeof(float));
-    if (floatsToWrite == 0) {
-        ALOGE("%s not able to write, floats in buffer %zu, space in FMQ %zu", __func__,
-              mInBuffer->getSize() / sizeof(float), available);
-        return INVALID_OPERATION;
-    }
-    if (!mInBuffer->audioBuffer() ||
-        !inputQ->write((float*)mInBuffer->audioBuffer()->f32, floatsToWrite)) {
-        ALOGE("%s failed to write %zu floats from audiobuffer %p to inputQ [avail %zu]", __func__,
-              floatsToWrite, mInBuffer->audioBuffer(), inputQ->availableToWrite());
-        return INVALID_OPERATION;
+    const size_t fmqSpaceSamples = inputQ->availableToWrite();
+    const size_t samplesInBuffer =
+            mInBuffer->audioBuffer()->frameCount * mConversion->getInputChannelCount();
+    const size_t samplesToWrite = std::min(fmqSpaceSamples, samplesInBuffer);
+    if (samplesToWrite == 0) {
+        ALOGE("%s not able to write, samplesInBuffer %zu, fmqSpaceSamples %zu", mEffectName.c_str(),
+              samplesInBuffer, fmqSpaceSamples);
+        return 0;
     }
 
-    // for V2 audio effect HAL, expect different EventFlag to avoid bit conflict with FMQ_NOT_EMPTY
-    efGroup->wake(halVersion >= kReopenSupportedVersion ? kEventFlagDataMqNotEmpty
-                                                        : kEventFlagNotEmpty);
+    const float* const inputRawBuffer = static_cast<const float*>(mInBuffer->audioBuffer()->f32);
+    if (!inputQ->write(inputRawBuffer, samplesToWrite)) {
+        ALOGE("%s failed to write %zu samples to inputQ [avail %zu]", mEffectName.c_str(),
+              samplesToWrite, inputQ->availableToWrite());
+        return 0;
+    }
+
+    efGroup->wake(mEventFlagDataMqNotEmpty);
+    return samplesToWrite;
+}
+
+void EffectHalAidl::writeHapticGeneratorData(size_t totalSamples, float* const outputRawBuffer,
+                                             float* const fmqOutputBuffer) const {
+    const auto audioChNum = mConversion->getAudioChannelCount();
+    const auto audioSamples =
+            totalSamples * audioChNum / (audioChNum + mConversion->getHapticChannelCount());
+
+    static constexpr float kHalFloatSampleLimit = 2.0f;
+    // for HapticGenerator, the input data buffer will be updated
+    float* const inputRawBuffer = static_cast<float*>(mInBuffer->audioBuffer()->f32);
+    // accumulate or copy input to output, haptic samples remains all zero
+    if (mConversion->mOutputAccessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
+        accumulate_float(outputRawBuffer, inputRawBuffer, audioSamples);
+    } else {
+        memcpy_to_float_from_float_with_clamping(outputRawBuffer, inputRawBuffer, audioSamples,
+                                                 kHalFloatSampleLimit);
+    }
+    // append the haptic sample at the end of input audio samples
+    memcpy_to_float_from_float_with_clamping(inputRawBuffer + audioSamples,
+                                             fmqOutputBuffer + audioSamples,
+                                             totalSamples - audioSamples, kHalFloatSampleLimit);
+}
+
+status_t EffectHalAidl::waitHalStatusFmq(size_t samplesWritten) const {
+    const auto statusQ = mConversion->getStatusMQ();
+    if (const bool statusValid = statusQ && statusQ->isValid(); !statusValid) {
+        ALOGE("%s statusFMQ %s", mEffectName.c_str(), statusValid ? "valid" : "invalid");
+        return INVALID_OPERATION;
+    }
 
     IEffect::Status retStatus{};
     if (!statusQ->readBlocking(&retStatus, 1)) {
-        ALOGE("%s %s V%d read status from status FMQ failed", __func__, effectName.c_str(),
-              halVersion);
+        ALOGE("%s V%d read status from status FMQ failed", mEffectName.c_str(), mHalVersion);
         return INVALID_OPERATION;
     }
-    if (retStatus.status != OK || (size_t)retStatus.fmqConsumed != floatsToWrite ||
+    if (retStatus.status != OK || (size_t)retStatus.fmqConsumed != samplesWritten ||
         retStatus.fmqProduced == 0) {
-        ALOGE("%s read status failed: %s, consumed %d (of %zu) produced %d", __func__,
-              retStatus.toString().c_str(), retStatus.fmqConsumed, floatsToWrite,
-              retStatus.fmqProduced);
+        ALOGE("%s read status failed: %s, FMQ consumed %d (of %zu) produced %d",
+              mEffectName.c_str(), retStatus.toString().c_str(), retStatus.fmqConsumed,
+              samplesWritten, retStatus.fmqProduced);
         return INVALID_OPERATION;
     }
 
-    available = outputQ->availableToRead();
-    const size_t floatsToRead = std::min(available, mOutBuffer->getSize() / sizeof(float));
-    if (floatsToRead == 0) {
-        ALOGE("%s not able to read, buffer space %zu, floats in FMQ %zu", __func__,
-              mOutBuffer->getSize() / sizeof(float), available);
+    return OK;
+}
+
+status_t EffectHalAidl::readFromHalOutputFmq(size_t samplesWritten) const {
+    const auto outputQ = mConversion->getOutputMQ();
+    if (const bool outputValid = outputQ && outputQ->isValid(); !outputValid) {
+        ALOGE("%s outputFMQ %s", mEffectName.c_str(), outputValid ? "valid" : "invalid");
         return INVALID_OPERATION;
     }
 
-    float *outputRawBuffer = mOutBuffer->audioBuffer()->f32;
+    const size_t fmqProducedSamples = outputQ->availableToRead();
+    const size_t bufferSpaceSamples =
+            mOutBuffer->audioBuffer()->frameCount * mConversion->getOutputChannelCount();
+    const size_t samplesToRead = std::min(fmqProducedSamples, bufferSpaceSamples);
+    if (samplesToRead == 0) {
+        ALOGE("%s unable to read, bufferSpace %zu, fmqProduced %zu samplesWritten %zu",
+              mEffectName.c_str(), bufferSpaceSamples, fmqProducedSamples, samplesWritten);
+        return INVALID_OPERATION;
+    }
+
+    float* const outputRawBuffer = static_cast<float*>(mOutBuffer->audioBuffer()->f32);
+    float* fmqOutputBuffer = outputRawBuffer;
     std::vector<float> tempBuffer;
     // keep original data in the output buffer for accumulate mode or HapticGenerator effect
     if (mConversion->mOutputAccessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE || mIsHapticGenerator) {
-        tempBuffer.resize(floatsToRead);
-        outputRawBuffer = tempBuffer.data();
+        tempBuffer.resize(samplesToRead, 0);
+        fmqOutputBuffer = tempBuffer.data();
     }
     // always read floating point data for AIDL
-    if (!outputQ->read(outputRawBuffer, floatsToRead)) {
-        ALOGE("%s failed to read %zu from outputQ to audioBuffer %p", __func__, floatsToRead,
-              mOutBuffer->audioBuffer());
+    if (!outputQ->read(fmqOutputBuffer, samplesToRead)) {
+        ALOGE("%s failed to read %zu from outputQ to audioBuffer %p", mEffectName.c_str(),
+              samplesToRead, fmqOutputBuffer);
         return INVALID_OPERATION;
     }
 
@@ -276,26 +337,10 @@
     // offset as input buffer, here we skip the audio samples in output FMQ and append haptic
     // samples to the end of input buffer
     if (mIsHapticGenerator) {
-        static constexpr float kHalFloatSampleLimit = 2.0f;
-        assert(floatsToRead == floatsToWrite);
-        const auto audioChNum = mConversion->getAudioChannelCount();
-        const auto audioSamples =
-                floatsToWrite * audioChNum / (audioChNum + mConversion->getHapticChannelCount());
-        // accumulate or copy input to output, haptic samples remains all zero
-        if (mConversion->mOutputAccessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
-            accumulate_float(mOutBuffer->audioBuffer()->f32, mInBuffer->audioBuffer()->f32,
-                             audioSamples);
-        } else {
-            memcpy_to_float_from_float_with_clamping(mOutBuffer->audioBuffer()->f32,
-                                                     mInBuffer->audioBuffer()->f32, audioSamples,
-                                                     kHalFloatSampleLimit);
-        }
-        // append the haptic sample at the end of input audio samples
-        memcpy_to_float_from_float_with_clamping(mInBuffer->audioBuffer()->f32 + audioSamples,
-                                                 outputRawBuffer + audioSamples,
-                                                 floatsToRead - audioSamples, kHalFloatSampleLimit);
+        assert(samplesRead == samplesWritten);
+        writeHapticGeneratorData(samplesToRead, outputRawBuffer, fmqOutputBuffer);
     } else if (mConversion->mOutputAccessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
-        accumulate_float(mOutBuffer->audioBuffer()->f32, outputRawBuffer, floatsToRead);
+        accumulate_float(outputRawBuffer, fmqOutputBuffer, samplesToRead);
     }
 
     return OK;
diff --git a/media/libaudiohal/impl/EffectHalAidl.h b/media/libaudiohal/impl/EffectHalAidl.h
index 4f7de7c..a775337 100644
--- a/media/libaudiohal/impl/EffectHalAidl.h
+++ b/media/libaudiohal/impl/EffectHalAidl.h
@@ -73,7 +73,12 @@
     const int32_t mSessionId;
     const int32_t mIoId;
     const bool mIsProxyEffect;
+    const int32_t mHalVersion;
+    // Audio effect HAL v2+ changes flag to kEventFlagDataMqNotEmpty to avoid conflict from using
+    // kEventFlagNotEmpty
+    const uint32_t mEventFlagDataMqNotEmpty;
     bool mIsHapticGenerator = false;
+    std::string mEffectName;
 
     std::unique_ptr<EffectConversionHelperAidl> mConversion;
 
@@ -93,6 +98,14 @@
     bool setEffectReverse(bool reverse);
     bool needUpdateReturnParam(uint32_t cmdCode);
 
+    status_t maybeReopen(const std::shared_ptr<android::hardware::EventFlag>& efGroup) const;
+    void writeHapticGeneratorData(size_t totalSamples, float* const outputRawBuffer,
+                                  float* const fmqOutputBuffer) const;
+    size_t writeToHalInputFmqAndSignal(
+            const std::shared_ptr<android::hardware::EventFlag>& efGroup) const;
+    status_t waitHalStatusFmq(size_t samplesWritten) const;
+    status_t readFromHalOutputFmq(size_t samplesWritten) const;
+
     // The destructor automatically releases the effect.
     virtual ~EffectHalAidl();
 };
diff --git a/media/libaudiohal/impl/StreamHalAidl.cpp b/media/libaudiohal/impl/StreamHalAidl.cpp
index 63be1bc..d65701a 100644
--- a/media/libaudiohal/impl/StreamHalAidl.cpp
+++ b/media/libaudiohal/impl/StreamHalAidl.cpp
@@ -59,6 +59,16 @@
 template<HalCommand::Tag cmd, typename T> HalCommand makeHalCommand(T data) {
     return HalCommand::make<cmd>(data);
 }
+
+template <typename MQTypeError>
+auto fmqErrorHandler(const char* mqName) {
+    return [m = std::string(mqName)](MQTypeError fmqError, std::string&& errorMessage) {
+        mediautils::TimeCheck::signalAudioHals();
+        LOG_ALWAYS_FATAL_IF(fmqError != MQTypeError::NONE, "%s: %s",
+                m.c_str(), errorMessage.c_str());
+    };
+}
+
 }  // namespace
 
 // static
@@ -103,6 +113,17 @@
             StreamHalAidl::getAudioProperties(&config) == NO_ERROR) {
         mStreamPowerLog.init(config.sample_rate, config.channel_mask, config.format);
     }
+
+    if (mStream != nullptr) {
+        mContext.getCommandMQ()->setErrorHandler(
+                fmqErrorHandler<StreamContextAidl::CommandMQ::Error>("CommandMQ"));
+        mContext.getReplyMQ()->setErrorHandler(
+                fmqErrorHandler<StreamContextAidl::ReplyMQ::Error>("ReplyMQ"));
+        if (mContext.getDataMQ() != nullptr) {
+            mContext.getDataMQ()->setErrorHandler(
+                    fmqErrorHandler<StreamContextAidl::DataMQ::Error>("DataMQ"));
+        }
+    }
 }
 
 StreamHalAidl::~StreamHalAidl() {
@@ -388,11 +409,8 @@
             return INVALID_OPERATION;
         }
     }
-    StreamContextAidl::DataMQ::Error fmqError = StreamContextAidl::DataMQ::Error::NONE;
-    std::string fmqErrorMsg;
     if (!mIsInput) {
-        bytes = std::min(bytes,
-                mContext.getDataMQ()->availableToWrite(&fmqError, &fmqErrorMsg));
+        bytes = std::min(bytes, mContext.getDataMQ()->availableToWrite());
     }
     StreamDescriptor::Command burst =
             StreamDescriptor::Command::make<StreamDescriptor::Command::Tag::burst>(bytes);
@@ -409,14 +427,12 @@
         LOG_ALWAYS_FATAL_IF(*transferred > bytes,
                 "%s: HAL module read %zu bytes, which exceeds requested count %zu",
                 __func__, *transferred, bytes);
-        if (auto toRead = mContext.getDataMQ()->availableToRead(&fmqError, &fmqErrorMsg);
+        if (auto toRead = mContext.getDataMQ()->availableToRead();
                 toRead != 0 && !mContext.getDataMQ()->read(static_cast<int8_t*>(buffer), toRead)) {
             AUGMENT_LOG(E, "failed to read %zu bytes to data MQ", toRead);
             return NOT_ENOUGH_DATA;
         }
     }
-    LOG_ALWAYS_FATAL_IF(fmqError != StreamContextAidl::DataMQ::Error::NONE,
-            "%s", fmqErrorMsg.c_str());
     mStreamPowerLog.log(buffer, *transferred);
     return OK;
 }
diff --git a/media/libaudioprocessing/AudioMixerOps.h b/media/libaudioprocessing/AudioMixerOps.h
index ab6a8b6..8f60d29 100644
--- a/media/libaudioprocessing/AudioMixerOps.h
+++ b/media/libaudioprocessing/AudioMixerOps.h
@@ -347,6 +347,7 @@
         [6] = AUDIO_CHANNEL_OUT_5POINT1,
         [7] = AUDIO_CHANNEL_OUT_6POINT1,
         [8] = AUDIO_CHANNEL_OUT_7POINT1,
+        [10] = AUDIO_CHANNEL_OUT_5POINT1POINT4,
         [12] = AUDIO_CHANNEL_OUT_7POINT1POINT4,
         [14] = AUDIO_CHANNEL_OUT_9POINT1POINT4,
         [16] = AUDIO_CHANNEL_OUT_9POINT1POINT6,
diff --git a/media/libaudioprocessing/tests/mixerops_tests.cpp b/media/libaudioprocessing/tests/mixerops_tests.cpp
index 2500ba9..235129f 100644
--- a/media/libaudioprocessing/tests/mixerops_tests.cpp
+++ b/media/libaudioprocessing/tests/mixerops_tests.cpp
@@ -154,6 +154,9 @@
 TEST(mixerops, stereovolume_8) {
     MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 8>::testStereoVolume();
 }
+TEST(mixerops, stereovolume_10) {
+    MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 10>::testStereoVolume();
+}
 TEST(mixerops, stereovolume_12) {
     if constexpr (FCC_LIMIT >= 12) { // NOTE: FCC_LIMIT is an enum, so can't #if
         MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 12>::testStereoVolume();
diff --git a/media/libmediaplayerservice/Android.bp b/media/libmediaplayerservice/Android.bp
index 718f782..a10c509 100644
--- a/media/libmediaplayerservice/Android.bp
+++ b/media/libmediaplayerservice/Android.bp
@@ -45,6 +45,7 @@
         "android.hardware.media.omx@1.0",
         "av-types-aidl-cpp",
         "framework-permission-aidl-cpp",
+        "libaconfig_storage_read_api_cc",
         "libaudioclient_aidl_conversion",
         "libbase",
         "libbinder_ndk",
@@ -76,6 +77,7 @@
         "libstagefright_httplive",
         "libutils",
         "packagemanager_aidl-cpp",
+        "server_configurable_flags",
     ],
 
     header_libs: [
@@ -86,6 +88,7 @@
     ],
 
     static_libs: [
+        "com.android.media.flags.editing-aconfig-cc",
         "libplayerservice_datasource",
         "libstagefright_nuplayer",
         "libstagefright_rtsp",
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index dce6ba8..086baa3 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -68,6 +68,7 @@
 #include <system/audio.h>
 
 #include <media/stagefright/rtsp/ARTPWriter.h>
+#include <com_android_media_editing_flags.h>
 
 namespace android {
 
@@ -2121,7 +2122,8 @@
         uint32_t bLayers = std::min(2u, tsLayers - 1); // use up-to 2 B-layers
         // TODO(b/341121900): Remove this once B frames are handled correctly in screen recorder
         // use case in case of mic only
-        if (mAudioSource == AUDIO_SOURCE_MIC && mVideoSource == VIDEO_SOURCE_SURFACE) {
+        if (!com::android::media::editing::flags::stagefrightrecorder_enable_b_frames()
+                && mAudioSource == AUDIO_SOURCE_MIC && mVideoSource == VIDEO_SOURCE_SURFACE) {
             bLayers = 0;
         }
         uint32_t pLayers = tsLayers - bLayers;
diff --git a/media/libstagefright/Android.bp b/media/libstagefright/Android.bp
index ac178aa..d084f10 100644
--- a/media/libstagefright/Android.bp
+++ b/media/libstagefright/Android.bp
@@ -321,6 +321,7 @@
 
     static_libs: [
         "android.media.codec-aconfig-cc",
+        "com.android.media.flags.editing-aconfig-cc",
         "libstagefright_esds",
         "libstagefright_color_conversion",
         "libyuv",
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index 76b6aa6..3aa0107 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -53,6 +53,8 @@
 #include <media/esds/ESDS.h>
 #include "include/HevcUtils.h"
 
+#include <com_android_media_editing_flags.h>
+
 #ifndef __predict_false
 #define __predict_false(exp) __builtin_expect((exp) != 0, 0)
 #endif
@@ -4944,6 +4946,8 @@
             // Track with start offset.
             ALOGV("Tracks starting > 0");
             int32_t editDurationTicks = 0;
+            int32_t trackStartOffsetBFramesUs = getMinCttsOffsetTimeUs() - kMaxCttsOffsetTimeUs;
+            ALOGV("trackStartOffsetBFramesUs:%" PRId32, trackStartOffsetBFramesUs);
             if (mMinCttsOffsetTicks == mMaxCttsOffsetTicks) {
                 // Video with no B frame or non-video track.
                 editDurationTicks =
@@ -4952,8 +4956,6 @@
                 ALOGV("editDuration:%" PRId64 "us", (trackStartOffsetUs + movieStartOffsetBFramesUs));
             } else {
                 // Track with B frame.
-                int32_t trackStartOffsetBFramesUs = getMinCttsOffsetTimeUs() - kMaxCttsOffsetTimeUs;
-                ALOGV("trackStartOffsetBFramesUs:%" PRId32, trackStartOffsetBFramesUs);
                 editDurationTicks =
                         ((trackStartOffsetUs + movieStartOffsetBFramesUs +
                           trackStartOffsetBFramesUs) * mvhdTimeScale + 5E5) / 1E6;
@@ -4967,7 +4969,15 @@
             } else if (editDurationTicks < 0) {
                 // Only video tracks with B Frames would hit this case.
                 ALOGV("Edit list entry to negate start offset by B frames in other tracks");
-                addOneElstTableEntry(tkhdDurationTicks, std::abs(editDurationTicks), 1, 0);
+                if (com::android::media::editing::flags::
+                        stagefrightrecorder_enable_b_frames()) {
+                    int32_t mediaTimeTicks =
+                            ((trackStartOffsetUs + movieStartOffsetBFramesUs +
+                              trackStartOffsetBFramesUs) * mTimeScale - 5E5) / 1E6;
+                    addOneElstTableEntry(tkhdDurationTicks, std::abs(mediaTimeTicks), 1, 0);
+                } else {
+                    addOneElstTableEntry(tkhdDurationTicks, std::abs(editDurationTicks), 1, 0);
+                }
             } else {
                 ALOGV("No edit list entry needed for this track");
             }
diff --git a/media/libstagefright/writer_fuzzers/Android.bp b/media/libstagefright/writer_fuzzers/Android.bp
index 58aa7cd..840c6b3c 100644
--- a/media/libstagefright/writer_fuzzers/Android.bp
+++ b/media/libstagefright/writer_fuzzers/Android.bp
@@ -24,6 +24,7 @@
     // to get the below license kinds:
     //   SPDX-license-identifier-Apache-2.0
     default_applicable_licenses: ["frameworks_av_media_libstagefright_license"],
+    default_team: "trendy_team_android_media_solutions_editing",
 }
 
 cc_defaults {
@@ -35,14 +36,17 @@
         "include",
     ],
     static_libs: [
+        "com.android.media.flags.editing-aconfig-cc",
         "liblog",
-        "libstagefright_foundation",
         "libstagefright",
+        "libstagefright_foundation",
     ],
     shared_libs: [
+        "libaconfig_storage_read_api_cc",
         "libbinder",
         "libcutils",
         "libutils",
+        "server_configurable_flags",
     ],
 }
 
@@ -96,9 +100,9 @@
 }
 
 cc_fuzz {
-    name : "mpeg4_writer_fuzzer",
-    defaults : ["writer-fuzzer-defaults"],
-    srcs : [
+    name: "mpeg4_writer_fuzzer",
+    defaults: ["writer-fuzzer-defaults"],
+    srcs: [
         "mpeg4_writer_fuzzer.cpp",
     ],
     static_libs: [
@@ -107,9 +111,9 @@
 }
 
 cc_fuzz {
-    name : "ogg_writer_fuzzer",
-    defaults : ["writer-fuzzer-defaults"],
-    srcs : [
+    name: "ogg_writer_fuzzer",
+    defaults: ["writer-fuzzer-defaults"],
+    srcs: [
         "ogg_writer_fuzzer.cpp",
     ],
     static_libs: [
@@ -118,9 +122,9 @@
 }
 
 cc_fuzz {
-    name : "webm_writer_fuzzer",
-    defaults : ["writer-fuzzer-defaults"],
-    srcs : [
+    name: "webm_writer_fuzzer",
+    defaults: ["writer-fuzzer-defaults"],
+    srcs: [
         "webm_writer_fuzzer.cpp",
     ],
     static_libs: [
diff --git a/media/libstagefright/xmlparser/MediaCodecsXmlParser.cpp b/media/libstagefright/xmlparser/MediaCodecsXmlParser.cpp
index 8c1ef3b..bd11326 100644
--- a/media/libstagefright/xmlparser/MediaCodecsXmlParser.cpp
+++ b/media/libstagefright/xmlparser/MediaCodecsXmlParser.cpp
@@ -1069,7 +1069,7 @@
         codec.rank = rank;
     }
 
-    codec.variantSet = variants;
+    codec.variantSet.insert(variants.begin(), variants.end());
 
     // we allow sets of domains...
     for (const std::string &domain : domains) {
diff --git a/media/mediaserver/Android.bp b/media/mediaserver/Android.bp
index 6ea40e3..d916fd1 100644
--- a/media/mediaserver/Android.bp
+++ b/media/mediaserver/Android.bp
@@ -54,6 +54,8 @@
 mediaserver_cc_binary {
     name: "mediaserver",
 
+    defaults: ["libcodec2_hal_selection"],
+
     srcs: ["main_mediaserver.cpp"],
 
     shared_libs: [
@@ -61,6 +63,7 @@
         "libicu",
         "libfmq",
         "libbinder",
+        "libbinder_ndk",
         "libhidlbase",
         "liblog",
         "libmediaplayerservice",
diff --git a/media/mediaserver/main_mediaserver.cpp b/media/mediaserver/main_mediaserver.cpp
index 026847a..8a62f30 100644
--- a/media/mediaserver/main_mediaserver.cpp
+++ b/media/mediaserver/main_mediaserver.cpp
@@ -17,11 +17,12 @@
 
 #define LOG_TAG "mediaserver"
 //#define LOG_NDEBUG 0
-
+#include <android/binder_process.h>
 #include <binder/IPCThreadState.h>
 #include <binder/ProcessState.h>
 #include <binder/IServiceManager.h>
 #include <hidl/HidlTransportSupport.h>
+#include <codec2/common/HalSelection.h>
 #include <utils/Log.h>
 #include "RegisterExtensions.h"
 
@@ -30,6 +31,14 @@
 
 using namespace android;
 
+namespace {
+    constexpr int kCodecThreadPoolCount = 16;
+
+    // This is the default thread count for binder thread pool
+    // if the thread count is not configured.
+    constexpr int kDefaultBinderThreadPoolCount = 15;
+}; // anonymous
+
 int main(int argc __unused, char **argv __unused)
 {
     signal(SIGPIPE, SIG_IGN);
@@ -40,8 +49,14 @@
     MediaPlayerService::instantiate();
     ResourceManagerService::instantiate();
     registerExtensions();
-    ::android::hardware::configureRpcThreadpool(16, false);
+
+    bool aidl = ::android::IsCodec2AidlHalSelected();
+    if (!aidl) {
+        ::android::hardware::configureRpcThreadpool(kCodecThreadPoolCount, false);
+    } else {
+        ABinderProcess_setThreadPoolMaxThreadCount(
+                kCodecThreadPoolCount + kDefaultBinderThreadPoolCount);
+    }
     ProcessState::self()->startThreadPool();
     IPCThreadState::self()->joinThreadPool();
-    ::android::hardware::joinRpcThreadpool();
 }
diff --git a/media/module/codecs/amrnb/dec/test/AmrnbDecoderTest.cpp b/media/module/codecs/amrnb/dec/test/AmrnbDecoderTest.cpp
index af62074..984baf8 100644
--- a/media/module/codecs/amrnb/dec/test/AmrnbDecoderTest.cpp
+++ b/media/module/codecs/amrnb/dec/test/AmrnbDecoderTest.cpp
@@ -22,6 +22,7 @@
 
 #include <audio_utils/sndfile.h>
 #include <stdio.h>
+#include <fstream>
 
 #include "gsmamr_dec.h"
 
@@ -40,7 +41,7 @@
 
 static AmrnbDecTestEnvironment *gEnv = nullptr;
 
-class AmrnbDecoderTest : public ::testing::TestWithParam<string> {
+class AmrnbDecoderTest : public ::testing::TestWithParam<std::tuple<string, string>> {
   public:
     AmrnbDecoderTest() : mFpInput(nullptr) {}
 
@@ -54,6 +55,7 @@
     FILE *mFpInput;
     SNDFILE *openOutputFile(SF_INFO *sfInfo);
     int32_t DecodeFrames(void *amrHandle, SNDFILE *outFileHandle, int32_t frameCount = INT32_MAX);
+    bool compareBinaryFiles(const std::string& refFilePath, const std::string& outFilePath);
 };
 
 SNDFILE *AmrnbDecoderTest::openOutputFile(SF_INFO *sfInfo) {
@@ -97,6 +99,42 @@
     return 0;
 }
 
+bool AmrnbDecoderTest::compareBinaryFiles(const std::string &refFilePath,
+                                          const std::string &outFilePath) {
+    std::ifstream refFile(refFilePath, std::ios::binary | std::ios::ate);
+    std::ifstream outFile(outFilePath, std::ios::binary | std::ios::ate);
+    assert(refFile.is_open() && "Error opening reference file " + refFilePath);
+    assert(outFile.is_open() && "Error opening output file " + outFilePath);
+
+    std::streamsize refFileSize = refFile.tellg();
+    std::streamsize outFileSize = outFile.tellg();
+    if (refFileSize != outFileSize) {
+        ALOGE("Error, File size mismatch: Reference file size = %td bytes,"
+              " but output file size = %td bytes.", refFileSize, outFileSize);
+        return false;
+    }
+
+    refFile.seekg(0, std::ios::beg);
+    outFile.seekg(0, std::ios::beg);
+    constexpr std::streamsize kBufferSize = 16 * 1024;
+    char refBuffer[kBufferSize];
+    char outBuffer[kBufferSize];
+
+    while (refFile && outFile) {
+        refFile.read(refBuffer, kBufferSize);
+        outFile.read(outBuffer, kBufferSize);
+
+        std::streamsize refBytesRead = refFile.gcount();
+        std::streamsize outBytesRead = outFile.gcount();
+
+        if (refBytesRead != outBytesRead || memcmp(refBuffer, outBuffer, refBytesRead) != 0) {
+            ALOGE("Error, File content mismatch.");
+            return false;
+        }
+    }
+    return true;
+}
+
 TEST_F(AmrnbDecoderTest, CreateAmrnbDecoderTest) {
     void *amrHandle;
     int32_t status = GSMInitDecode(&amrHandle, (Word8 *)"AMRNBDecoder");
@@ -106,7 +144,7 @@
 }
 
 TEST_P(AmrnbDecoderTest, DecodeTest) {
-    string inputFile = gEnv->getRes() + GetParam();
+    string inputFile = gEnv->getRes() + std::get<0>(GetParam());
     mFpInput = fopen(inputFile.c_str(), "rb");
     ASSERT_NE(mFpInput, nullptr) << "Error opening input file " << inputFile;
 
@@ -126,10 +164,15 @@
     sf_close(outFileHandle);
     GSMDecodeFrameExit(&amrHandle);
     ASSERT_EQ(amrHandle, nullptr) << "Error deleting AMR-NB decoder";
+
+    string refFilePath = gEnv->getRes() + std::get<1>(GetParam());
+    ASSERT_TRUE(compareBinaryFiles(refFilePath, OUTPUT_FILE))
+       << "Error, Binary file comparison failed: Output file " << OUTPUT_FILE
+       << " does not match the reference file " << refFilePath << ".";
 }
 
 TEST_P(AmrnbDecoderTest, ResetDecodeTest) {
-    string inputFile = gEnv->getRes() + GetParam();
+    string inputFile = gEnv->getRes() + std::get<0>(GetParam());
     mFpInput = fopen(inputFile.c_str(), "rb");
     ASSERT_NE(mFpInput, nullptr) << "Error opening input file " << inputFile;
 
@@ -159,8 +202,24 @@
 }
 
 INSTANTIATE_TEST_SUITE_P(AmrnbDecoderTestAll, AmrnbDecoderTest,
-                         ::testing::Values(("bbb_8000hz_1ch_8kbps_amrnb_30sec.amrnb"),
-                                           ("sine_amrnb_1ch_12kbps_8000hz.amrnb")));
+                         ::testing::Values(std::make_tuple(
+                                                   "bbb_8000hz_1ch_8kbps_amrnb_30sec.amrnb",
+                                                   "bbb_8000hz_1ch_8kbps_amrnb_30sec_ref.pcm"),
+                                           std::make_tuple(
+                                                   "sine_amrnb_1ch_12kbps_8000hz.amrnb",
+                                                   "sine_amrnb_1ch_12kbps_8000hz_ref.pcm"),
+                                           std::make_tuple(
+                                                   "trim_8000hz_1ch_12kpbs_amrnb_200ms.amrnb",
+                                                   "trim_8000hz_1ch_12kpbs_amrnb_200ms_ref.pcm"),
+                                           std::make_tuple(
+                                                   "bbb_8kHz_1ch_4.75kbps_amrnb_3sec.amrnb",
+                                                   "bbb_8kHz_1ch_4.75kbps_amrnb_3sec_ref.pcm"),
+                                           std::make_tuple(
+                                                   "bbb_8kHz_1ch_10kbps_amrnb_1sec.amrnb",
+                                                   "bbb_8kHz_1ch_10kbps_amrnb_1sec_ref.pcm"),
+                                           std::make_tuple(
+                                                   "bbb_8kHz_1ch_12.2kbps_amrnb_3sec.amrnb",
+                                                   "bbb_8kHz_1ch_12.2kbps_amrnb_3sec_ref.pcm")));
 
 int main(int argc, char **argv) {
     gEnv = new AmrnbDecTestEnvironment();
diff --git a/media/module/codecs/amrnb/dec/test/AndroidTest.xml b/media/module/codecs/amrnb/dec/test/AndroidTest.xml
index 539fa5c..7b2ba15 100644
--- a/media/module/codecs/amrnb/dec/test/AndroidTest.xml
+++ b/media/module/codecs/amrnb/dec/test/AndroidTest.xml
@@ -23,17 +23,17 @@
     <target_preparer class="com.android.compatibility.common.tradefed.targetprep.DynamicConfigPusher">
         <option name="target" value="host" />
         <option name="config-filename" value="AmrnbDecoderTest" />
-        <option name="version" value="1.0"/>
+        <option name="version" value="2.0"/>
     </target_preparer>
     <target_preparer class="com.android.compatibility.common.tradefed.targetprep.MediaPreparer">
         <option name="push-all" value="true" />
-        <option name="media-folder-name" value="AmrnbDecoderTest-1.0" />
+        <option name="media-folder-name" value="AmrnbDecoderTest-2.0" />
         <option name="dynamic-config-module" value="AmrnbDecoderTest" />
     </target_preparer>
 
     <test class="com.android.tradefed.testtype.GTest" >
         <option name="native-test-device-path" value="/data/local/tmp" />
         <option name="module-name" value="AmrnbDecoderTest" />
-        <option name="native-test-flag" value="-P /sdcard/test/AmrnbDecoderTest-1.0/" />
+        <option name="native-test-flag" value="-P /sdcard/test/AmrnbDecoderTest-2.0/" />
     </test>
 </configuration>
diff --git a/media/module/codecs/amrnb/dec/test/DynamicConfig.xml b/media/module/codecs/amrnb/dec/test/DynamicConfig.xml
index 701a752..02b869a 100644
--- a/media/module/codecs/amrnb/dec/test/DynamicConfig.xml
+++ b/media/module/codecs/amrnb/dec/test/DynamicConfig.xml
@@ -15,6 +15,6 @@
 
 <dynamicConfig>
     <entry key="media_files_url">
-            <value>https://dl.google.com/android-unittest/media/frameworks/av/media/module/codecs/amrnb/dec/test/AmrnbDecoderTest-1.0.zip</value>
+            <value>https://dl.google.com/android-unittest/media/frameworks/av/media/module/codecs/amrnb/dec/test/AmrnbDecoderTest-2.0.zip</value>
     </entry>
 </dynamicConfig>
diff --git a/media/module/codecs/amrnb/dec/test/README.md b/media/module/codecs/amrnb/dec/test/README.md
index 41fb80a..ea54975 100644
--- a/media/module/codecs/amrnb/dec/test/README.md
+++ b/media/module/codecs/amrnb/dec/test/README.md
@@ -22,15 +22,15 @@
 adb push ${OUT}/data/nativetest/AmrnbDecoderTest/AmrnbDecoderTest /data/local/tmp/
 ```
 
-The resource file for the tests is taken from [here](https://dl.google.com/android-unittest/media/frameworks/av/media/module/codecs/amrnb/dec/test/AmrnbDecoderTest-1.0.zip). Download, unzip and push these files into device for testing.
+The resource file for the tests is taken from [here](https://dl.google.com/android-unittest/media/frameworks/av/media/module/codecs/amrnb/dec/test/AmrnbDecoderTest-2.0.zip). Download, unzip and push these files into device for testing.
 
 ```
-adb push AmrnbDecoderTest-1.0 /data/local/tmp/
+adb push AmrnbDecoderTest-2.0 /data/local/tmp/
 ```
 
 usage: AmrnbDecoderTest -P \<path_to_folder\>
 ```
-adb shell /data/local/tmp/AmrnbDecoderTest -P /data/local/tmp/AmrnbDecoderTest-1.0/
+adb shell /data/local/tmp/AmrnbDecoderTest -P /data/local/tmp/AmrnbDecoderTest-2.0/
 ```
 Alternatively, the test can also be run using atest command.
 
diff --git a/media/module/codecs/amrnb/enc/test/AmrnbEncoderTest.cpp b/media/module/codecs/amrnb/enc/test/AmrnbEncoderTest.cpp
index fb72998..e3bd0e0 100644
--- a/media/module/codecs/amrnb/enc/test/AmrnbEncoderTest.cpp
+++ b/media/module/codecs/amrnb/enc/test/AmrnbEncoderTest.cpp
@@ -21,6 +21,7 @@
 
 #include <audio_utils/sndfile.h>
 #include <stdio.h>
+#include <fstream>
 
 #include "gsmamr_enc.h"
 
@@ -39,7 +40,7 @@
 
 static AmrnbEncTestEnvironment *gEnv = nullptr;
 
-class AmrnbEncoderTest : public ::testing::TestWithParam<pair<string, int32_t>> {
+class AmrnbEncoderTest : public ::testing::TestWithParam<tuple<string, int32_t, string>> {
   public:
     AmrnbEncoderTest() : mAmrEncHandle(nullptr) {}
 
@@ -53,6 +54,7 @@
     AmrNbEncState *mAmrEncHandle;
     int32_t EncodeFrames(int32_t mode, FILE *fpInput, FILE *mFpOutput,
                          int32_t frameCount = INT32_MAX);
+    bool compareBinaryFiles(const string& refFilePath, const string& outFilePath);
 };
 
 int32_t AmrnbEncoderTest::EncodeFrames(int32_t mode, FILE *fpInput, FILE *mFpOutput,
@@ -87,6 +89,42 @@
     return 0;
 }
 
+bool AmrnbEncoderTest::compareBinaryFiles(const std::string &refFilePath,
+                                          const std::string &outFilePath) {
+    std::ifstream refFile(refFilePath, std::ios::binary | std::ios::ate);
+    std::ifstream outFile(outFilePath, std::ios::binary | std::ios::ate);
+    assert(refFile.is_open() && "Error opening reference file " + refFilePath);
+    assert(outFile.is_open() && "Error opening output file " + outFilePath);
+
+    std::streamsize refFileSize = refFile.tellg();
+    std::streamsize outFileSize = outFile.tellg();
+    if (refFileSize != outFileSize) {
+        ALOGE("Error, File size mismatch: Reference file size = %td bytes,"
+              " but output file size = %td bytes.", refFileSize, outFileSize);
+        return false;
+    }
+
+    refFile.seekg(0, std::ios::beg);
+    outFile.seekg(0, std::ios::beg);
+    constexpr std::streamsize kBufferSize = 16 * 1024;
+    char refBuffer[kBufferSize];
+    char outBuffer[kBufferSize];
+
+    while (refFile && outFile) {
+        refFile.read(refBuffer, kBufferSize);
+        outFile.read(outBuffer, kBufferSize);
+
+        std::streamsize refBytesRead = refFile.gcount();
+        std::streamsize outBytesRead = outFile.gcount();
+
+        if (refBytesRead != outBytesRead || memcmp(refBuffer, outBuffer, refBytesRead) != 0) {
+            ALOGE("Error, File content mismatch.");
+            return false;
+        }
+    }
+    return true;
+}
+
 TEST_F(AmrnbEncoderTest, CreateAmrnbEncoderTest) {
     mAmrEncHandle = (AmrNbEncState *)malloc(sizeof(AmrNbEncState));
     ASSERT_NE(mAmrEncHandle, nullptr) << "Error in allocating memory to Codec handle";
@@ -111,7 +149,7 @@
     int32_t status = AMREncodeInit(&mAmrEncHandle->encCtx, &mAmrEncHandle->pidSyncCtx, 0);
     ASSERT_EQ(status, 0) << "Error creating AMR-NB encoder";
 
-    string inputFile = gEnv->getRes() + GetParam().first;
+    string inputFile = gEnv->getRes() + std::get<0>(GetParam());
     FILE *fpInput = fopen(inputFile.c_str(), "rb");
     ASSERT_NE(fpInput, nullptr) << "Error opening input file " << inputFile;
 
@@ -121,7 +159,7 @@
     // Write file header.
     fwrite("#!AMR\n", 1, 6, fpOutput);
 
-    int32_t mode = GetParam().second;
+    int32_t mode = std::get<1>(GetParam());
     int32_t encodeErr = EncodeFrames(mode, fpInput, fpOutput);
     ASSERT_EQ(encodeErr, 0) << "EncodeFrames returned error for Codec mode: " << mode;
 
@@ -134,6 +172,11 @@
     free(mAmrEncHandle);
     mAmrEncHandle = nullptr;
     ALOGV("Successfully deleted encoder");
+
+    string refFilePath = gEnv->getRes() + std::get<2>(GetParam());
+    ASSERT_TRUE(compareBinaryFiles(refFilePath, OUTPUT_FILE))
+       << "Error, Binary file comparison failed: Output file " << OUTPUT_FILE
+       << " does not match the reference file " << refFilePath << ".";
 }
 
 TEST_P(AmrnbEncoderTest, ResetEncoderTest) {
@@ -142,7 +185,7 @@
     int32_t status = AMREncodeInit(&mAmrEncHandle->encCtx, &mAmrEncHandle->pidSyncCtx, 0);
     ASSERT_EQ(status, 0) << "Error creating AMR-NB encoder";
 
-    string inputFile = gEnv->getRes() + GetParam().first;
+    string inputFile = gEnv->getRes() + std::get<0>(GetParam());
     FILE *fpInput = fopen(inputFile.c_str(), "rb");
     ASSERT_NE(fpInput, nullptr) << "Error opening input file " << inputFile;
 
@@ -152,7 +195,7 @@
     // Write file header.
     fwrite("#!AMR\n", 1, 6, fpOutput);
 
-    int32_t mode = GetParam().second;
+    int32_t mode = std::get<1>(GetParam());
     // Encode kNumFrameReset first
     int32_t encodeErr = EncodeFrames(mode, fpInput, fpOutput, kNumFrameReset);
     ASSERT_EQ(encodeErr, 0) << "EncodeFrames returned error for Codec mode: " << mode;
@@ -177,22 +220,23 @@
 
 // TODO: Add more test vectors
 INSTANTIATE_TEST_SUITE_P(AmrnbEncoderTestAll, AmrnbEncoderTest,
-                         ::testing::Values(make_pair("bbb_raw_1ch_8khz_s16le.raw", MR475),
-                                           make_pair("bbb_raw_1ch_8khz_s16le.raw", MR515),
-                                           make_pair("bbb_raw_1ch_8khz_s16le.raw", MR59),
-                                           make_pair("bbb_raw_1ch_8khz_s16le.raw", MR67),
-                                           make_pair("bbb_raw_1ch_8khz_s16le.raw", MR74),
-                                           make_pair("bbb_raw_1ch_8khz_s16le.raw", MR795),
-                                           make_pair("bbb_raw_1ch_8khz_s16le.raw", MR102),
-                                           make_pair("bbb_raw_1ch_8khz_s16le.raw", MR122),
-                                           make_pair("sinesweepraw.raw", MR475),
-                                           make_pair("sinesweepraw.raw", MR515),
-                                           make_pair("sinesweepraw.raw", MR59),
-                                           make_pair("sinesweepraw.raw", MR67),
-                                           make_pair("sinesweepraw.raw", MR74),
-                                           make_pair("sinesweepraw.raw", MR795),
-                                           make_pair("sinesweepraw.raw", MR102),
-                                           make_pair("sinesweepraw.raw", MR122)));
+    ::testing::Values(
+        make_tuple("bbb_raw_1ch_8khz_s16le.raw", MR475, "bbb_raw_1ch_8khz_s16le_MR475_ref.amrnb"),
+        make_tuple("bbb_raw_1ch_8khz_s16le.raw", MR515, "bbb_raw_1ch_8khz_s16le_MR515_ref.amrnb"),
+        make_tuple("bbb_raw_1ch_8khz_s16le.raw", MR59, "bbb_raw_1ch_8khz_s16le_MR59_ref.amrnb"),
+        make_tuple("bbb_raw_1ch_8khz_s16le.raw", MR67, "bbb_raw_1ch_8khz_s16le_MR67_ref.amrnb"),
+        make_tuple("bbb_raw_1ch_8khz_s16le.raw", MR74, "bbb_raw_1ch_8khz_s16le_MR74_ref.amrnb"),
+        make_tuple("bbb_raw_1ch_8khz_s16le.raw", MR795, "bbb_raw_1ch_8khz_s16le_MR795_ref.amrnb"),
+        make_tuple("bbb_raw_1ch_8khz_s16le.raw", MR102, "bbb_raw_1ch_8khz_s16le_MR102_ref.amrnb"),
+        make_tuple("bbb_raw_1ch_8khz_s16le.raw", MR122, "bbb_raw_1ch_8khz_s16le_MR122_ref.amrnb"),
+        make_tuple("sinesweepraw.raw", MR475, "sinesweepraw_MR475_ref.amrnb"),
+        make_tuple("sinesweepraw.raw", MR515, "sinesweepraw_MR515_ref.amrnb"),
+        make_tuple("sinesweepraw.raw", MR59, "sinesweepraw_MR59_ref.amrnb"),
+        make_tuple("sinesweepraw.raw", MR67, "sinesweepraw_MR67_ref.amrnb"),
+        make_tuple("sinesweepraw.raw", MR74, "sinesweepraw_MR74_ref.amrnb"),
+        make_tuple("sinesweepraw.raw", MR795, "sinesweepraw_MR795_ref.amrnb"),
+        make_tuple("sinesweepraw.raw", MR102, "sinesweepraw_MR102_ref.amrnb"),
+        make_tuple("sinesweepraw.raw", MR122, "sinesweepraw_MR122_ref.amrnb")));
 
 int main(int argc, char **argv) {
     gEnv = new AmrnbEncTestEnvironment();
diff --git a/media/module/codecs/amrnb/enc/test/AndroidTest.xml b/media/module/codecs/amrnb/enc/test/AndroidTest.xml
index 1509728..a325ee8 100644
--- a/media/module/codecs/amrnb/enc/test/AndroidTest.xml
+++ b/media/module/codecs/amrnb/enc/test/AndroidTest.xml
@@ -23,17 +23,17 @@
     <target_preparer class="com.android.compatibility.common.tradefed.targetprep.DynamicConfigPusher">
         <option name="target" value="host" />
         <option name="config-filename" value="AmrnbEncoderTest" />
-        <option name="version" value="1.0"/>
+        <option name="version" value="2.0"/>
     </target_preparer>
     <target_preparer class="com.android.compatibility.common.tradefed.targetprep.MediaPreparer">
         <option name="push-all" value="true" />
-        <option name="media-folder-name" value="AmrnbEncoderTest-1.0" />
+        <option name="media-folder-name" value="AmrnbEncoderTest-2.0" />
         <option name="dynamic-config-module" value="AmrnbEncoderTest" />
     </target_preparer>
 
     <test class="com.android.tradefed.testtype.GTest" >
         <option name="native-test-device-path" value="/data/local/tmp" />
         <option name="module-name" value="AmrnbEncoderTest" />
-        <option name="native-test-flag" value="-P /sdcard/test/AmrnbEncoderTest-1.0/" />
+        <option name="native-test-flag" value="-P /sdcard/test/AmrnbEncoderTest-2.0/" />
     </test>
 </configuration>
diff --git a/media/module/codecs/amrnb/enc/test/DynamicConfig.xml b/media/module/codecs/amrnb/enc/test/DynamicConfig.xml
index 713667a..fdc0daa 100644
--- a/media/module/codecs/amrnb/enc/test/DynamicConfig.xml
+++ b/media/module/codecs/amrnb/enc/test/DynamicConfig.xml
@@ -15,6 +15,6 @@
 
 <dynamicConfig>
     <entry key="media_files_url">
-            <value>https://dl.google.com/android-unittest/media/frameworks/av/media/module/codecs/amrnb/enc/test/AmrnbEncoderTest-1.0.zip</value>
+            <value>https://dl.google.com/android-unittest/media/frameworks/av/media/module/codecs/amrnb/enc/test/AmrnbEncoderTest-2.0.zip</value>
     </entry>
 </dynamicConfig>
diff --git a/media/module/codecs/amrnb/enc/test/README.md b/media/module/codecs/amrnb/enc/test/README.md
index f896bd1..c7b9964 100644
--- a/media/module/codecs/amrnb/enc/test/README.md
+++ b/media/module/codecs/amrnb/enc/test/README.md
@@ -22,15 +22,15 @@
 adb push ${OUT}/data/nativetest/AmrnbEncoderTest/AmrnbEncoderTest /data/local/tmp/
 ```
 
-The resource file for the tests is taken from [here](https://dl.google.com/android-unittest/media/frameworks/av/media/module/codecs/amrnb/enc/test/AmrnbEncoderTest-1.0.zip). Download, unzip and push these files into device for testing.
+The resource file for the tests is taken from [here](https://dl.google.com/android-unittest/media/frameworks/av/media/module/codecs/amrnb/enc/test/AmrnbEncoderTest-2.0.zip). Download, unzip and push these files into device for testing.
 
 ```
-adb push AmrnbEncoderTest-1.0 /data/local/tmp/
+adb push AmrnbEncoderTest-2.0 /data/local/tmp/
 ```
 
 usage: AmrnbEncoderTest -P \<path_to_folder\>
 ```
-adb shell /data/local/tmp/AmrnbEncoderTest -P /data/local/tmp/AmrnbEncoderTest-1.0/
+adb shell /data/local/tmp/AmrnbEncoderTest -P /data/local/tmp/AmrnbEncoderTest-2.0/
 ```
 Alternatively, the test can also be run using atest command.
 
diff --git a/media/module/codecs/amrwb/dec/test/AmrwbDecoderTest.cpp b/media/module/codecs/amrwb/dec/test/AmrwbDecoderTest.cpp
index 2cc88ce..c0e032f 100644
--- a/media/module/codecs/amrwb/dec/test/AmrwbDecoderTest.cpp
+++ b/media/module/codecs/amrwb/dec/test/AmrwbDecoderTest.cpp
@@ -23,6 +23,7 @@
 #include <audio_utils/sndfile.h>
 #include <memory>
 #include <stdio.h>
+#include <fstream>
 
 #include "pvamrwbdecoder.h"
 #include "pvamrwbdecoder_api.h"
@@ -44,7 +45,7 @@
 
 static AmrwbDecTestEnvironment *gEnv = nullptr;
 
-class AmrwbDecoderTest : public ::testing::TestWithParam<string> {
+class AmrwbDecoderTest : public ::testing::TestWithParam<std::tuple<string, string>> {
   public:
     AmrwbDecoderTest() : mFpInput(nullptr) {}
 
@@ -59,6 +60,7 @@
     int32_t DecodeFrames(int16_t *decoderCookie, void *decoderBuf, SNDFILE *outFileHandle,
                          int32_t frameCount = INT32_MAX);
     SNDFILE *openOutputFile(SF_INFO *sfInfo);
+    bool compareBinaryFiles(const std::string& refFilePath, const std::string& outFilePath);
 };
 
 SNDFILE *AmrwbDecoderTest::openOutputFile(SF_INFO *sfInfo) {
@@ -120,6 +122,42 @@
     return 0;
 }
 
+bool AmrwbDecoderTest::compareBinaryFiles(const std::string &refFilePath,
+                                          const std::string &outFilePath) {
+    std::ifstream refFile(refFilePath, std::ios::binary | std::ios::ate);
+    std::ifstream outFile(outFilePath, std::ios::binary | std::ios::ate);
+    assert(refFile.is_open() && "Error opening reference file " + refFilePath);
+    assert(outFile.is_open() && "Error opening output file " + outFilePath);
+
+    std::streamsize refFileSize = refFile.tellg();
+    std::streamsize outFileSize = outFile.tellg();
+    if (refFileSize != outFileSize) {
+        ALOGE("Error, File size mismatch: Reference file size = %td bytes,"
+               "but output file size = %td bytes", refFileSize, outFileSize);
+        return false;
+    }
+
+    refFile.seekg(0, std::ios::beg);
+    outFile.seekg(0, std::ios::beg);
+    constexpr std::streamsize kBufferSize = 16 * 1024;
+    char refBuffer[kBufferSize];
+    char outBuffer[kBufferSize];
+
+    while (refFile && outFile) {
+        refFile.read(refBuffer, kBufferSize);
+        outFile.read(outBuffer, kBufferSize);
+
+        std::streamsize refBytesRead = refFile.gcount();
+        std::streamsize outBytesRead = outFile.gcount();
+
+        if (refBytesRead != outBytesRead || memcmp(refBuffer, outBuffer, refBytesRead) != 0) {
+            ALOGE("Error, File content mismatch.");
+            return false;
+        }
+    }
+    return true;
+}
+
 TEST_F(AmrwbDecoderTest, MultiCreateAmrwbDecoderTest) {
     uint32_t memRequirements = pvDecoder_AmrWbMemRequirements();
     std::unique_ptr<char[]> decoderBuf(new char[memRequirements]);
@@ -147,7 +185,7 @@
     pvDecoder_AmrWb_Init(&amrHandle, decoderBuf.get(), &decoderCookie);
     ASSERT_NE(amrHandle, nullptr) << "Failed to initialize decoder";
 
-    string inputFile = gEnv->getRes() + GetParam();
+    string inputFile = gEnv->getRes() + std::get<0>(GetParam());
     mFpInput = fopen(inputFile.c_str(), "rb");
     ASSERT_NE(mFpInput, nullptr) << "Error opening input file " << inputFile;
 
@@ -160,6 +198,10 @@
     ASSERT_EQ(decoderErr, 0) << "DecodeFrames returned error";
 
     sf_close(outFileHandle);
+    string refFilePath = gEnv->getRes() + std::get<1>(GetParam());
+    ASSERT_TRUE(compareBinaryFiles(refFilePath, OUTPUT_FILE))
+    << "Error, Binary file comparison failed: Output file "
+    << OUTPUT_FILE << " does not match the reference file " << refFilePath << ".";
 }
 
 TEST_P(AmrwbDecoderTest, ResetDecoderTest) {
@@ -173,7 +215,7 @@
     pvDecoder_AmrWb_Init(&amrHandle, decoderBuf.get(), &decoderCookie);
     ASSERT_NE(amrHandle, nullptr) << "Failed to initialize decoder";
 
-    string inputFile = gEnv->getRes() + GetParam();
+    string inputFile = gEnv->getRes() + std::get<0>(GetParam());
     mFpInput = fopen(inputFile.c_str(), "rb");
     ASSERT_NE(mFpInput, nullptr) << "Error opening input file " << inputFile;
 
@@ -198,8 +240,21 @@
 }
 
 INSTANTIATE_TEST_SUITE_P(AmrwbDecoderTestAll, AmrwbDecoderTest,
-                         ::testing::Values(("bbb_amrwb_1ch_14kbps_16000hz.amrwb"),
-                                           ("bbb_16000hz_1ch_9kbps_amrwb_30sec.amrwb")));
+                         ::testing::Values(std::make_tuple(
+                                                "bbb_amrwb_1ch_14kbps_16000hz.amrwb",
+                                                "bbb_amrwb_1ch_14kbps_16000hz_ref.pcm"),
+                                           std::make_tuple(
+                                                "bbb_16000hz_1ch_9kbps_amrwb_30sec.amrwb",
+                                                "bbb_16000hz_1ch_9kbps_amrwb_30sec_ref.pcm"),
+                                           std::make_tuple(
+                                                "bbb_16kHz_1ch_16bps_1sec.amrwb",
+                                                "bbb_16kHz_1ch_16bps_1sec_ref.pcm"),
+                                           std::make_tuple(
+                                                "bbb_16kHz_1ch_6.6bps_3sec.amrwb",
+                                                "bbb_16kHz_1ch_6.6bps_3sec_ref.pcm"),
+                                           std::make_tuple(
+                                                "bbb_16kHz_1ch_23.85bps_3sec.amrwb",
+                                                "bbb_16kHz_1ch_23.85bps_3sec_ref.pcm")));
 
 int main(int argc, char **argv) {
     gEnv = new AmrwbDecTestEnvironment();
diff --git a/media/module/codecs/amrwb/dec/test/AndroidTest.xml b/media/module/codecs/amrwb/dec/test/AndroidTest.xml
index 392df03..dbd1407 100644
--- a/media/module/codecs/amrwb/dec/test/AndroidTest.xml
+++ b/media/module/codecs/amrwb/dec/test/AndroidTest.xml
@@ -23,17 +23,17 @@
     <target_preparer class="com.android.compatibility.common.tradefed.targetprep.DynamicConfigPusher">
         <option name="target" value="host" />
         <option name="config-filename" value="AmrwbDecoderTest" />
-        <option name="version" value="1.0"/>
+        <option name="version" value="2.0"/>
     </target_preparer>
     <target_preparer class="com.android.compatibility.common.tradefed.targetprep.MediaPreparer">
         <option name="push-all" value="true" />
-        <option name="media-folder-name" value="AmrwbDecoderTest-1.0" />
+        <option name="media-folder-name" value="AmrwbDecoderTest-2.0" />
         <option name="dynamic-config-module" value="AmrwbDecoderTest" />
     </target_preparer>
 
     <test class="com.android.tradefed.testtype.GTest" >
         <option name="native-test-device-path" value="/data/local/tmp" />
         <option name="module-name" value="AmrwbDecoderTest" />
-        <option name="native-test-flag" value="-P /sdcard/test/AmrwbDecoderTest-1.0/" />
+        <option name="native-test-flag" value="-P /sdcard/test/AmrwbDecoderTest-2.0/" />
     </test>
 </configuration>
diff --git a/media/module/codecs/amrwb/dec/test/DynamicConfig.xml b/media/module/codecs/amrwb/dec/test/DynamicConfig.xml
index 506cc3d..52453ee 100644
--- a/media/module/codecs/amrwb/dec/test/DynamicConfig.xml
+++ b/media/module/codecs/amrwb/dec/test/DynamicConfig.xml
@@ -15,6 +15,6 @@
 
 <dynamicConfig>
     <entry key="media_files_url">
-            <value>https://dl.google.com/android-unittest/media/frameworks/av/media/module/codecs/amrwb/dec/test/AmrwbDecoderTest-1.0.zip</value>
+            <value>https://dl.google.com/android-unittest/media/frameworks/av/media/module/codecs/amrwb/dec/test/AmrwbDecoderTest-2.0.zip</value>
     </entry>
 </dynamicConfig>
diff --git a/media/module/codecs/amrwb/dec/test/README.md b/media/module/codecs/amrwb/dec/test/README.md
index 8e77456..ed76051 100644
--- a/media/module/codecs/amrwb/dec/test/README.md
+++ b/media/module/codecs/amrwb/dec/test/README.md
@@ -22,15 +22,15 @@
 adb push ${OUT}/data/nativetest/AmrwbDecoderTest/AmrwbDecoderTest /data/local/tmp/
 ```
 
-The resource file for the tests is taken from [here](https://dl.google.com/android-unittest/media/frameworks/av/media/module/codecs/amrwb/dec/test/AmrwbDecoderTest-1.0.zip). Download, unzip and push these files into device for testing.
+The resource file for the tests is taken from [here](https://dl.google.com/android-unittest/media/frameworks/av/media/module/codecs/amrwb/dec/test/AmrwbDecoderTest-2.0.zip). Download, unzip and push these files into device for testing.
 
 ```
-adb push AmrwbDecoderTest-1.0 /data/local/tmp/
+adb push AmrwbDecoderTest-2.0 /data/local/tmp/
 ```
 
 usage: AmrwbDecoderTest -P \<path_to_folder\>
 ```
-adb shell /data/local/tmp/AmrwbDecoderTest -P /data/local/tmp/AmrwbDecoderTest-1.0/
+adb shell /data/local/tmp/AmrwbDecoderTest -P /data/local/tmp/AmrwbDecoderTest-2.0/
 ```
 Alternatively, the test can also be run using atest command.
 
diff --git a/media/module/codecs/amrwb/enc/test/AmrwbEncoderTest.cpp b/media/module/codecs/amrwb/enc/test/AmrwbEncoderTest.cpp
index 1a6ee27..dc9c1b1 100644
--- a/media/module/codecs/amrwb/enc/test/AmrwbEncoderTest.cpp
+++ b/media/module/codecs/amrwb/enc/test/AmrwbEncoderTest.cpp
@@ -20,6 +20,7 @@
 #include <utils/Log.h>
 
 #include <stdio.h>
+#include <fstream>
 
 #include "cmnMemory.h"
 #include "voAMRWB.h"
@@ -34,13 +35,15 @@
 
 static AmrwbEncTestEnvironment *gEnv = nullptr;
 
-class AmrwbEncoderTest : public ::testing::TestWithParam<tuple<string, int32_t, VOAMRWBFRAMETYPE>> {
+class AmrwbEncoderTest : public ::testing::TestWithParam<tuple<string, int32_t,
+                                                               VOAMRWBFRAMETYPE, string>> {
   public:
     AmrwbEncoderTest() : mEncoderHandle(nullptr) {
-        tuple<string, int32_t, VOAMRWBFRAMETYPE> params = GetParam();
+        tuple<string, int32_t, VOAMRWBFRAMETYPE, string> params = GetParam();
         mInputFile = gEnv->getRes() + get<0>(params);
         mMode = get<1>(params);
         mFrameType = get<2>(params);
+        refFilePath = gEnv->getRes() + get<3>(params);
         mMemOperator.Alloc = cmnMemAlloc;
         mMemOperator.Copy = cmnMemCopy;
         mMemOperator.Free = cmnMemFree;
@@ -66,8 +69,47 @@
     VO_CODEC_INIT_USERDATA mUserData;
     VO_HANDLE mEncoderHandle;
     int32_t mMode;
+    string refFilePath;
+
+    bool compareBinaryFiles(const string& refFilePath, const string& outFilePath);
 };
 
+bool AmrwbEncoderTest::compareBinaryFiles(const std::string &refFilePath,
+                                          const std::string &outFilePath) {
+    std::ifstream refFile(refFilePath, std::ios::binary | std::ios::ate);
+    std::ifstream outFile(outFilePath, std::ios::binary | std::ios::ate);
+    assert(refFile.is_open() && "Error opening reference file " + refFilePath);
+    assert(outFile.is_open() && "Error opening output file " + outFilePath);
+
+    std::streamsize refFileSize = refFile.tellg();
+    std::streamsize outFileSize = outFile.tellg();
+    if (refFileSize != outFileSize) {
+        ALOGE("Error, File size mismatch: Reference file size = %td bytes,"
+               "but output file size = %td bytes", refFileSize, outFileSize);
+        return false;
+    }
+
+    refFile.seekg(0, std::ios::beg);
+    outFile.seekg(0, std::ios::beg);
+    constexpr std::streamsize kBufferSize = 16 * 1024;
+    char refBuffer[kBufferSize];
+    char outBuffer[kBufferSize];
+
+    while (refFile && outFile) {
+        refFile.read(refBuffer, kBufferSize);
+        outFile.read(outBuffer, kBufferSize);
+
+        std::streamsize refBytesRead = refFile.gcount();
+        std::streamsize outBytesRead = outFile.gcount();
+
+        if (refBytesRead != outBytesRead || memcmp(refBuffer, outBuffer, refBytesRead) != 0) {
+            ALOGE("Error, File content mismatch.");
+            return false;
+        }
+    }
+    return true;
+}
+
 TEST_P(AmrwbEncoderTest, CreateAmrwbEncoderTest) {
     int32_t status = voGetAMRWBEncAPI(&mApiHandle);
     ASSERT_EQ(status, VO_ERR_NONE) << "Failed to get api handle";
@@ -152,38 +194,69 @@
     if (fpOutput) {
         fclose(fpOutput);
     }
+
+    ASSERT_TRUE(compareBinaryFiles(refFilePath, OUTPUT_FILE))
+    << "Error, Binary file comparison failed: Output file "
+    << OUTPUT_FILE << " does not match the reference file " << refFilePath << ".";
 }
 
 INSTANTIATE_TEST_SUITE_P(
-        AmrwbEncoderTestAll, AmrwbEncoderTest,
-        ::testing::Values(
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD66, VOAMRWB_DEFAULT),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD885, VOAMRWB_DEFAULT),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1265, VOAMRWB_DEFAULT),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1425, VOAMRWB_DEFAULT),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1585, VOAMRWB_DEFAULT),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1825, VOAMRWB_DEFAULT),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1985, VOAMRWB_DEFAULT),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD2305, VOAMRWB_DEFAULT),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD2385, VOAMRWB_DEFAULT),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD66, VOAMRWB_ITU),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD885, VOAMRWB_ITU),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1265, VOAMRWB_ITU),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1425, VOAMRWB_ITU),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1585, VOAMRWB_ITU),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1825, VOAMRWB_ITU),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1985, VOAMRWB_ITU),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD2305, VOAMRWB_ITU),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD2385, VOAMRWB_ITU),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD66, VOAMRWB_RFC3267),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD885, VOAMRWB_RFC3267),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1265, VOAMRWB_RFC3267),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1425, VOAMRWB_RFC3267),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1585, VOAMRWB_RFC3267),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1825, VOAMRWB_RFC3267),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1985, VOAMRWB_RFC3267),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD2305, VOAMRWB_RFC3267),
-                make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD2385, VOAMRWB_RFC3267)));
+    AmrwbEncoderTestAll, AmrwbEncoderTest,
+    ::testing::Values(
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD66, VOAMRWB_DEFAULT,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD66_VOAMRWB_DEFAULT_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD885, VOAMRWB_DEFAULT,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD885_VOAMRWB_DEFAULT_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1265, VOAMRWB_DEFAULT,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD1265_VOAMRWB_DEFAULT_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1425, VOAMRWB_DEFAULT,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD1425_VOAMRWB_DEFAULT_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1585, VOAMRWB_DEFAULT,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD1585_VOAMRWB_DEFAULT_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1825, VOAMRWB_DEFAULT,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD1825_VOAMRWB_DEFAULT_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1985, VOAMRWB_DEFAULT,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD1985_VOAMRWB_DEFAULT_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD2305, VOAMRWB_DEFAULT,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD2305_VOAMRWB_DEFAULT_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD2385, VOAMRWB_DEFAULT,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD2385_VOAMRWB_DEFAULT_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD66, VOAMRWB_ITU,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD66_VOAMRWB_ITU_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD885, VOAMRWB_ITU,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD885_VOAMRWB_ITU_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1265, VOAMRWB_ITU,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD1265_VOAMRWB_ITU_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1425, VOAMRWB_ITU,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD1425_VOAMRWB_ITU_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1585, VOAMRWB_ITU,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD1585_VOAMRWB_ITU_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1825, VOAMRWB_ITU,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD1825_VOAMRWB_ITU_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1985, VOAMRWB_ITU,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD1985_VOAMRWB_ITU_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD2305, VOAMRWB_ITU,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD2305_VOAMRWB_ITU_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD2385, VOAMRWB_ITU,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD2385_VOAMRWB_ITU_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD66, VOAMRWB_RFC3267,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD66_VOAMRWB_RFC3267_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD885, VOAMRWB_RFC3267,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD885_VOAMRWB_RFC3267_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1265, VOAMRWB_RFC3267,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD1265_VOAMRWB_RFC3267_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1425, VOAMRWB_RFC3267,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD1425_VOAMRWB_RFC3267_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1585, VOAMRWB_RFC3267,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD1585_VOAMRWB_RFC3267_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1825, VOAMRWB_RFC3267,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD1825_VOAMRWB_RFC3267_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1985, VOAMRWB_RFC3267,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD1985_VOAMRWB_RFC3267_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD2305, VOAMRWB_RFC3267,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD2305_VOAMRWB_RFC3267_ref.amrwb"),
+        make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD2385, VOAMRWB_RFC3267,
+                    "bbb_raw_1ch_16khz_s16le_VOAMRWB_MD2385_VOAMRWB_RFC3267_ref.amrwb")));
 
 int main(int argc, char **argv) {
     gEnv = new AmrwbEncTestEnvironment();
diff --git a/media/module/codecs/amrwb/enc/test/AndroidTest.xml b/media/module/codecs/amrwb/enc/test/AndroidTest.xml
index 8822cb2..1f4121f 100644
--- a/media/module/codecs/amrwb/enc/test/AndroidTest.xml
+++ b/media/module/codecs/amrwb/enc/test/AndroidTest.xml
@@ -23,17 +23,17 @@
     <target_preparer class="com.android.compatibility.common.tradefed.targetprep.DynamicConfigPusher">
         <option name="target" value="host" />
         <option name="config-filename" value="AmrwbEncoderTest" />
-        <option name="version" value="1.0"/>
+        <option name="version" value="2.0"/>
     </target_preparer>
     <target_preparer class="com.android.compatibility.common.tradefed.targetprep.MediaPreparer">
         <option name="push-all" value="true" />
-        <option name="media-folder-name" value="AmrwbEncoderTest-1.0" />
+        <option name="media-folder-name" value="AmrwbEncoderTest-2.0" />
         <option name="dynamic-config-module" value="AmrwbEncoderTest" />
     </target_preparer>
 
     <test class="com.android.tradefed.testtype.GTest" >
         <option name="native-test-device-path" value="/data/local/tmp" />
         <option name="module-name" value="AmrwbEncoderTest" />
-        <option name="native-test-flag" value="-P /sdcard/test/AmrwbEncoderTest-1.0/" />
+        <option name="native-test-flag" value="-P /sdcard/test/AmrwbEncoderTest-2.0/" />
     </test>
 </configuration>
diff --git a/media/module/codecs/amrwb/enc/test/DynamicConfig.xml b/media/module/codecs/amrwb/enc/test/DynamicConfig.xml
index a0b6218..59701ea 100644
--- a/media/module/codecs/amrwb/enc/test/DynamicConfig.xml
+++ b/media/module/codecs/amrwb/enc/test/DynamicConfig.xml
@@ -15,6 +15,6 @@
 
 <dynamicConfig>
     <entry key="media_files_url">
-            <value>https://dl.google.com/android-unittest/media/frameworks/av/media/module/codecs/amrwb/enc/test/AmrwbEncoderTest-1.0.zip</value>
+            <value>https://dl.google.com/android-unittest/media/frameworks/av/media/module/codecs/amrwb/enc/test/AmrwbEncoderTest-2.0.zip</value>
     </entry>
 </dynamicConfig>
diff --git a/media/module/codecs/amrwb/enc/test/README.md b/media/module/codecs/amrwb/enc/test/README.md
index 3b9cc39..ea2c31e 100644
--- a/media/module/codecs/amrwb/enc/test/README.md
+++ b/media/module/codecs/amrwb/enc/test/README.md
@@ -22,7 +22,7 @@
 adb push ${OUT}/data/nativetest/AmrwbEncoderTest/AmrwbEncoderTest /data/local/tmp/
 ```
 
-The resource file for the tests is taken from [here](https://dl.google.com/android-unittest/media/frameworks/av/media/module/codecs/amrwb/enc/test/AmrwbEncoderTest-1.0.zip). Download, unzip and push these files into device for testing.
+The resource file for the tests is taken from [here](https://dl.google.com/android-unittest/media/frameworks/av/media/module/codecs/amrwb/enc/test/AmrwbEncoderTest-2.0.zip). Download, unzip and push these files into device for testing.
 
 ```
 adb push AmrwbEncoderTest-1.0 /data/local/tmp/
@@ -30,7 +30,7 @@
 
 usage: AmrwbEncoderTest -P \<path_to_folder\>
 ```
-adb shell /data/local/tmp/AmrwbEncoderTest -P /data/local/tmp/AmrwbEncoderTest-1.0/
+adb shell /data/local/tmp/AmrwbEncoderTest -P /data/local/tmp/AmrwbEncoderTest-2.0/
 ```
 Alternatively, the test can also be run using atest command.
 
diff --git a/media/module/codecserviceregistrant/CodecServiceRegistrant.cpp b/media/module/codecserviceregistrant/CodecServiceRegistrant.cpp
index 42fd94e..433b5e9 100644
--- a/media/module/codecserviceregistrant/CodecServiceRegistrant.cpp
+++ b/media/module/codecserviceregistrant/CodecServiceRegistrant.cpp
@@ -790,47 +790,6 @@
     }
 
     using namespace ::android::hardware::media::c2;
-
-    if (!ionPropertiesDefined()) {
-        using IComponentStore =
-            ::android::hardware::media::c2::V1_0::IComponentStore;
-        std::string const preferredStoreName = "default";
-        if (aidlSelected) {
-            std::shared_ptr<c2_aidl::IComponentStore> preferredStore;
-            if (__builtin_available(android __ANDROID_API_S__, *)) {
-                std::string instanceName = ::android::base::StringPrintf(
-                        "%s/%s", c2_aidl::IComponentStore::descriptor, preferredStoreName.c_str());
-                if (AServiceManager_isDeclared(instanceName.c_str())) {
-                    preferredStore = c2_aidl::IComponentStore::fromBinder(::ndk::SpAIBinder(
-                            AServiceManager_waitForService(instanceName.c_str())));
-                }
-            }
-            if (preferredStore) {
-                ::android::SetPreferredCodec2ComponentStore(
-                        std::make_shared<H2C2ComponentStore>(preferredStore));
-                LOG(INFO) <<
-                        "Preferred Codec2 AIDL store is set to \"" <<
-                        preferredStoreName << "\".";
-            } else {
-                LOG(INFO) <<
-                        "Preferred Codec2 AIDL store is defaulted to \"software\".";
-            }
-        } else {
-            sp<IComponentStore> preferredStore =
-                IComponentStore::getService(preferredStoreName.c_str());
-            if (preferredStore) {
-                ::android::SetPreferredCodec2ComponentStore(
-                        std::make_shared<H2C2ComponentStore>(preferredStore));
-                LOG(INFO) <<
-                        "Preferred Codec2 HIDL store is set to \"" <<
-                        preferredStoreName << "\".";
-            } else {
-                LOG(INFO) <<
-                        "Preferred Codec2 HIDL store is defaulted to \"software\".";
-            }
-        }
-    }
-
     bool registered = false;
     const std::string aidlServiceName =
         std::string(c2_aidl::IComponentStore::descriptor) + "/software";
@@ -876,6 +835,48 @@
                      " so it is not being registered with hwservicemanager.";
     }
 
+    // Preferred store must be set after the store is registered to ensure that
+    // the correct preferred store is set.
+    if (!ionPropertiesDefined()) {
+        using IComponentStore =
+            ::android::hardware::media::c2::V1_0::IComponentStore;
+        std::string const preferredStoreName = "default";
+        if (aidlSelected) {
+            std::shared_ptr<c2_aidl::IComponentStore> preferredStore;
+            if (__builtin_available(android __ANDROID_API_S__, *)) {
+                std::string instanceName = ::android::base::StringPrintf(
+                        "%s/%s", c2_aidl::IComponentStore::descriptor, preferredStoreName.c_str());
+                if (AServiceManager_isDeclared(instanceName.c_str())) {
+                    preferredStore = c2_aidl::IComponentStore::fromBinder(::ndk::SpAIBinder(
+                            AServiceManager_waitForService(instanceName.c_str())));
+                }
+            }
+            if (preferredStore) {
+                ::android::SetPreferredCodec2ComponentStore(
+                        std::make_shared<H2C2ComponentStore>(preferredStore));
+                LOG(INFO) <<
+                        "Preferred Codec2 AIDL store is set to \"" <<
+                        preferredStoreName << "\".";
+            } else {
+                LOG(INFO) <<
+                        "Preferred Codec2 AIDL store is defaulted to \"software\".";
+            }
+        } else {
+            sp<IComponentStore> preferredStore =
+                IComponentStore::getService(preferredStoreName.c_str());
+            if (preferredStore) {
+                ::android::SetPreferredCodec2ComponentStore(
+                        std::make_shared<H2C2ComponentStore>(preferredStore));
+                LOG(INFO) <<
+                        "Preferred Codec2 HIDL store is set to \"" <<
+                        preferredStoreName << "\".";
+            } else {
+                LOG(INFO) <<
+                        "Preferred Codec2 HIDL store is defaulted to \"software\".";
+            }
+        }
+    }
+
     if (registered) {
         LOG(INFO) << "Software Codec2 service created and registered.";
     }
diff --git a/media/utils/TimeCheck.cpp b/media/utils/TimeCheck.cpp
index 658191e..6a5bbbe 100644
--- a/media/utils/TimeCheck.cpp
+++ b/media/utils/TimeCheck.cpp
@@ -184,6 +184,22 @@
 }
 
 /* static */
+std::string TimeCheck::signalAudioHals() {
+    std::vector<pid_t> pids = getAudioHalPids();
+    std::string halPids;
+    if (pids.size() != 0) {
+        for (const auto& pid : pids) {
+            ALOGI("requesting tombstone for pid: %d", pid);
+            halPids.append(std::to_string(pid)).append(" ");
+            signalAudioHAL(pid);
+        }
+        // Allow time to complete, usually the caller is forcing restart afterwards.
+        sleep(1);
+    }
+    return halPids;
+}
+
+/* static */
 TimerThread& TimeCheck::getTimeCheckThread() {
     static TimerThread sTimeCheckThread{};
     return sTimeCheckThread;
@@ -302,21 +318,14 @@
     // HAL processes which can affect thread behavior.
     const auto snapshotAnalysis = getTimeCheckThread().getSnapshotAnalysis(4 /* retiredCount */);
 
-    // Generate audio HAL processes tombstones and allow time to complete
-    // before forcing restart
-    std::vector<pid_t> pids = TimeCheck::getAudioHalPids();
-    std::string halPids = "HAL pids [ ";
-    if (pids.size() != 0) {
-        for (const auto& pid : pids) {
-            ALOGI("requesting tombstone for pid: %d", pid);
-            halPids.append(std::to_string(pid)).append(" ");
-            signalAudioHAL(pid);
-        }
-        sleep(1);
+    // Generate audio HAL processes tombstones.
+    std::string halPids = signalAudioHals();
+    if (!halPids.empty()) {
+        halPids = "HAL pids [ " + halPids + "]";
     } else {
-        ALOGI("No HAL process pid available, skipping tombstones");
+        halPids = "No HAL process pids available";
+        ALOGI("%s", (halPids + ", skipping tombstones").c_str());
     }
-    halPids.append("]");
 
     LOG_EVENT_STRING(LOGTAG_AUDIO_BINDER_TIMEOUT, tag.c_str());
 
diff --git a/media/utils/include/mediautils/TimeCheck.h b/media/utils/include/mediautils/TimeCheck.h
index 3e8d35d..c112863 100644
--- a/media/utils/include/mediautils/TimeCheck.h
+++ b/media/utils/include/mediautils/TimeCheck.h
@@ -107,6 +107,7 @@
     static std::string toString();
     static void setAudioHalPids(const std::vector<pid_t>& pids);
     static std::vector<pid_t> getAudioHalPids();
+    static std::string signalAudioHals();
 
   private:
     // Helper class for handling events.
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 2c1976b..537a097 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -3948,8 +3948,12 @@
         // TODO: We could check compatibility of the secondaryThread with the PatchTrack
         // for fast usage: thread has fast mixer, sample rate matches, etc.;
         // for now, we exclude fast tracks by removing the Fast flag.
+        constexpr audio_output_flags_t kIncompatiblePatchTrackFlags =
+                static_cast<audio_output_flags_t>(AUDIO_OUTPUT_FLAG_FAST
+                        | AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+
         const audio_output_flags_t outputFlags =
-                (audio_output_flags_t)(track->getOutputFlags() & ~AUDIO_OUTPUT_FLAG_FAST);
+                (audio_output_flags_t)(track->getOutputFlags() & ~kIncompatiblePatchTrackFlags);
         sp<IAfPatchTrack> patchTrack = IAfPatchTrack::create(secondaryThread,
                                                        track->streamType(),
                                                        track->sampleRate(),
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
index 794c7c0..b193cb8 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
@@ -16,10 +16,10 @@
 
 #pragma once
 
+#include <optional>
 #include <string>
 #include <unordered_map>
 #include <unordered_set>
-#include <vector>
 
 #include <DeviceDescriptor.h>
 #include <HwModule.h>
@@ -141,6 +141,12 @@
 
     void setDefault();
 
+    void setUseDeepBufferForMediaOverrideForTests(bool useDeepBufferForMedia)
+    {
+        mUseDeepBufferForMediaOverride = useDeepBufferForMedia;
+    }
+    bool useDeepBufferForMedia() const;
+
 private:
     friend class sp<AudioPolicyConfig>;
 
@@ -158,6 +164,7 @@
     sp<DeviceDescriptor> mDefaultOutputDevice;
     bool mIsCallScreenModeSupported = false;
     SurroundFormats mSurroundFormats;
+    std::optional<bool> mUseDeepBufferForMediaOverride;
 };
 
 } // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyConfig.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyConfig.cpp
index abeaaf8..f5e135e 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyConfig.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyConfig.cpp
@@ -16,6 +16,7 @@
 
 #define LOG_TAG "APM_Config"
 
+#include <android-base/properties.h>
 #include <AudioPolicyConfig.h>
 #include <IOProfile.h>
 #include <Serializer.h>
@@ -344,4 +345,9 @@
         {AUDIO_FORMAT_AC4, {}}};
 }
 
+bool AudioPolicyConfig::useDeepBufferForMedia() const {
+    if (mUseDeepBufferForMediaOverride.has_value()) return *mUseDeepBufferForMediaOverride;
+    return property_get_bool("audio.deep_buffer.media", false /* default_value */);
+}
+
 } // namespace android
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index f36d8d5..2ff2907 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -1153,8 +1153,7 @@
 
     SortedVector<audio_io_handle_t> outputs = getOutputsForDevices(devices, mOutputs);
     audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
-    if (stream == AUDIO_STREAM_MUSIC &&
-        property_get_bool("audio.deep_buffer.media", false /* default_value */)) {
+    if (stream == AUDIO_STREAM_MUSIC && mConfig->useDeepBufferForMedia()) {
         flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
     }
     const audio_io_handle_t output = selectOutput(outputs, flags);
@@ -1251,7 +1250,8 @@
 
     // FIXME: in case of RENDER policy, the output capabilities should be checked
     if ((secondaryMixes != nullptr && !secondaryMixes->empty())
-            && !audio_is_linear_pcm(config->format)) {
+            && (!audio_is_linear_pcm(config->format) ||
+                    *flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
         ALOGD("%s: rejecting request as secondary mixes only support pcm", __func__);
         return BAD_VALUE;
     }
@@ -1527,6 +1527,13 @@
         return NAME_NOT_FOUND;
     }
 
+    // Reject flag combinations that do not make sense. Note that the requested flags might not
+    // have the 'DIRECT' flag set, however once a direct-capable profile is found, it will
+    // combine the requested flags with its own flags, yielding an unsupported combination.
+    if ((flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) {
+        return NAME_NOT_FOUND;
+    }
+
     // Do not allow offloading if one non offloadable effect is enabled or MasterMono is enabled.
     // This prevents creating an offloaded track and tearing it down immediately after start
     // when audioflinger detects there is an active non offloadable effect.
@@ -1675,8 +1682,7 @@
     if (stream != AUDIO_STREAM_MUSIC) {
         *flags = (audio_output_flags_t)(*flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
     } else if (/* stream == AUDIO_STREAM_MUSIC && */
-            *flags == AUDIO_OUTPUT_FLAG_NONE &&
-            property_get_bool("audio.deep_buffer.media", false /* default_value */)) {
+            *flags == AUDIO_OUTPUT_FLAG_NONE && mConfig->useDeepBufferForMedia()) {
         // use DEEP_BUFFER as default output for music stream type
         *flags = (audio_output_flags_t)AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
     }
@@ -7415,7 +7421,8 @@
                     client->getSecondaryOutputs().begin(),
                     client->getSecondaryOutputs().end(),
                     secondaryDescs.begin(), secondaryDescs.end())) {
-                if (!audio_is_linear_pcm(client->config().format)) {
+                if (client->flags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD
+                        || !audio_is_linear_pcm(client->config().format)) {
                     // If the format is not PCM, the tracks should be invalidated to get correct
                     // behavior when the secondary output is changed.
                     clientsToInvalidate.push_back(client->portId());
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index 8d53bd5..d529130 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -1801,6 +1801,7 @@
                   ++numTimesBecameEmpty;
                 }
                 mLastCommand = command;
+                status_t createAudioPatchStatus;
 
                 switch (command->mCommand) {
                 case SET_VOLUME: {
@@ -1858,10 +1859,11 @@
                     ALOGV("AudioCommandThread() processing create audio patch");
                     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
                     if (af == 0) {
-                        command->mStatus = PERMISSION_DENIED;
+                        createAudioPatchStatus = PERMISSION_DENIED;
                     } else {
                         ul.unlock();
-                        command->mStatus = af->createAudioPatch(&data->mPatch, &data->mHandle);
+                        createAudioPatchStatus = af->createAudioPatch(&data->mPatch,
+                                                                      &data->mHandle);
                         ul.lock();
                     }
                     } break;
@@ -2030,8 +2032,28 @@
                 {
                     audio_utils::lock_guard _l(command->mMutex);
                     if (command->mWaitStatus) {
+                        if (command->mCommand == CREATE_AUDIO_PATCH) {
+                            command->mStatus = createAudioPatchStatus;
+                        }
                         command->mWaitStatus = false;
                         command->mCond.notify_one();
+                    } else if (command->mCommand == CREATE_AUDIO_PATCH &&
+                               command->mStatus == TIMED_OUT &&
+                               createAudioPatchStatus == NO_ERROR) {
+                        // Because of special handling in insertCommand_l() the CREATE_AUDIO_PATCH
+                        // command wait status can be only false in case timeout (see TIMED_OUT)
+                        // happened.
+                        CreateAudioPatchData *createData =
+                                (CreateAudioPatchData *)command->mParam.get();
+                        ALOGW("AudioCommandThread() no caller awaiting for handle(%d) after \
+                                processing create audio patch, going to release it",
+                                createData->mHandle);
+                        sp<AudioCommand> releaseCommand = new AudioCommand();
+                        releaseCommand->mCommand = RELEASE_AUDIO_PATCH;
+                        ReleaseAudioPatchData *releaseData = new ReleaseAudioPatchData();
+                        releaseData->mHandle = createData->mHandle;
+                        releaseCommand->mParam = releaseData;
+                        insertCommand_l(releaseCommand, 0);
                     }
                 }
                 waitTime = -1;
@@ -2549,7 +2571,8 @@
 
     // Disable wait for status if delay is not 0.
     // Except for create audio patch command because the returned patch handle
-    // is needed by audio policy manager
+    // is needed by audio policy manager. Audio patch created after timeout
+    // (see TIMED_OUT) will be released from threadLoop().
     if (delayMs != 0 && command->mCommand != CREATE_AUDIO_PATCH) {
         command->mWaitStatus = false;
     }
diff --git a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
index 5a25a77..ca7ad40 100644
--- a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
@@ -17,6 +17,7 @@
 #include <map>
 #include <set>
 
+#include <media/TypeConverter.h>
 #include <system/audio.h>
 #include <utils/Log.h>
 #include <utils/String8.h>
@@ -37,11 +38,11 @@
 
     status_t openOutput(audio_module_handle_t module,
                         audio_io_handle_t *output,
-                        audio_config_t * /*halConfig*/,
-                        audio_config_base_t * /*mixerConfig*/,
+                        audio_config_t *halConfig,
+                        audio_config_base_t *mixerConfig,
                         const sp<DeviceDescriptorBase>& /*device*/,
                         uint32_t * /*latencyMs*/,
-                        audio_output_flags_t /*flags*/,
+                        audio_output_flags_t flags,
                         audio_attributes_t /*attributes*/) override {
         if (module >= mNextModuleHandle) {
             ALOGE("%s: Module handle %d has not been allocated yet (next is %d)",
@@ -49,6 +50,13 @@
             return BAD_VALUE;
         }
         *output = mNextIoHandle++;
+        mOpenedOutputs[*output] = flags;
+        ALOGD("%s: opened output %d: HAL(%s %s %d) Mixer(%s %s %d) %s", __func__, *output,
+              audio_channel_out_mask_to_string(halConfig->channel_mask),
+              audio_format_to_string(halConfig->format), halConfig->sample_rate,
+              audio_channel_out_mask_to_string(mixerConfig->channel_mask),
+              audio_format_to_string(mixerConfig->format), mixerConfig->sample_rate,
+              android::toString(flags).c_str());
         return NO_ERROR;
     }
 
@@ -58,6 +66,16 @@
         return id;
     }
 
+    status_t closeOutput(audio_io_handle_t output) override {
+        if (auto iter = mOpenedOutputs.find(output); iter != mOpenedOutputs.end()) {
+            mOpenedOutputs.erase(iter);
+            return NO_ERROR;
+        } else {
+            ALOGE("%s: Unknown output %d", __func__, output);
+            return BAD_VALUE;
+        }
+    }
+
     status_t openInput(audio_module_handle_t module,
                        audio_io_handle_t *input,
                        audio_config_t * /*config*/,
@@ -262,6 +280,13 @@
         return it == mTracksInternalMute.end() ? false : it->second;
     }
 
+    std::optional<audio_output_flags_t> getOpenOutputFlags(audio_io_handle_t output) const {
+        if (auto iter = mOpenedOutputs.find(output); iter != mOpenedOutputs.end()) {
+            return iter->second;
+        }
+        return std::nullopt;
+    }
+
 private:
     audio_module_handle_t mNextModuleHandle = AUDIO_MODULE_HANDLE_NONE + 1;
     audio_io_handle_t mNextIoHandle = AUDIO_IO_HANDLE_NONE + 1;
@@ -276,6 +301,7 @@
     std::set<audio_channel_mask_t> mSupportedChannelMasks;
     std::map<audio_port_handle_t, bool> mTracksInternalMute;
     std::set<audio_io_handle_t> mOpenedInputs;
+    std::map<audio_io_handle_t, audio_output_flags_t> mOpenedOutputs;
 };
 
 } // namespace android
diff --git a/services/audiopolicy/tests/AudioPolicyTestManager.h b/services/audiopolicy/tests/AudioPolicyTestManager.h
index 34ceeab..bf45bb2 100644
--- a/services/audiopolicy/tests/AudioPolicyTestManager.h
+++ b/services/audiopolicy/tests/AudioPolicyTestManager.h
@@ -24,9 +24,11 @@
     explicit AudioPolicyTestManager(AudioPolicyClientInterface *clientInterface)
             : AudioPolicyTestManager(AudioPolicyConfig::createDefault(), clientInterface) {}
     AudioPolicyTestManager(const sp<const AudioPolicyConfig>& config,
-            AudioPolicyClientInterface *clientInterface)
+            AudioPolicyClientInterface *clientInterface,
+            std::string engineConfig = "")
             : AudioPolicyManager(config,
-                    loadApmEngineLibraryAndCreateEngine(config->getEngineLibraryNameSuffix()),
+                    loadApmEngineLibraryAndCreateEngine(config->getEngineLibraryNameSuffix(),
+                                                        engineConfig),
                     clientInterface) {}
     using AudioPolicyManager::getConfig;
     using AudioPolicyManager::initialize;
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index f958ede..08855c9 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -35,6 +35,7 @@
 #include <media/AudioPolicy.h>
 #include <media/PatchBuilder.h>
 #include <media/RecordingActivityTracker.h>
+#include <media/TypeConverter.h>
 #include <utils/Log.h>
 #include <utils/Vector.h>
 #include <cutils/multiuser.h>
@@ -175,6 +176,11 @@
 };
 
 class AudioPolicyManagerTest : public testing::Test {
+  public:
+    constexpr static uint32_t k384000SamplingRate = 384000;
+    constexpr static uint32_t k48000SamplingRate = 48000;
+    constexpr static uint32_t k96000SamplingRate = 96000;
+
   protected:
     void SetUp() override;
     void TearDown() override;
@@ -191,7 +197,7 @@
             audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
             audio_io_handle_t *output = nullptr,
             audio_port_handle_t *portId = nullptr,
-            audio_attributes_t attr = {},
+            audio_attributes_t attr = AUDIO_ATTRIBUTES_INITIALIZER,
             audio_session_t session = AUDIO_SESSION_NONE,
             int uid = 0,
             bool* isBitPerfect = nullptr);
@@ -222,13 +228,16 @@
     std::unique_ptr<AudioPolicyManagerTestClient> mClient;
     std::unique_ptr<AudioPolicyTestManager> mManager;
 
-    constexpr static const uint32_t k48000SamplingRate = 48000;
+    static const std::string sTestEngineConfig;
 };
 
+const std::string AudioPolicyManagerTest::sTestEngineConfig =
+        base::GetExecutableDirectory() + "/engine/test_audio_policy_engine_configuration.xml";
+
 void AudioPolicyManagerTest::SetUp() {
     mClient.reset(getClient());
     ASSERT_NO_FATAL_FAILURE(SetUpManagerConfig());  // Subclasses may want to customize the config.
-    mManager.reset(new AudioPolicyTestManager(mConfig, mClient.get()));
+    mManager.reset(new AudioPolicyTestManager(mConfig, mClient.get(), sTestEngineConfig));
     ASSERT_EQ(NO_ERROR, mManager->initialize());
     ASSERT_EQ(NO_ERROR, mManager->initCheck());
 }
@@ -782,27 +791,27 @@
 
     audio_config_base_t directConfig = AUDIO_CONFIG_BASE_INITIALIZER;
     directConfig.format = AUDIO_FORMAT_DTS;
-    directConfig.sample_rate = 48000;
+    directConfig.sample_rate = k48000SamplingRate;
     directConfig.channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
 
     audio_config_base_t nonDirectConfig = AUDIO_CONFIG_BASE_INITIALIZER;
     nonDirectConfig.format = AUDIO_FORMAT_PCM_16_BIT;
-    nonDirectConfig.sample_rate = 48000;
+    nonDirectConfig.sample_rate = k48000SamplingRate;
     nonDirectConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
 
     audio_config_base_t nonExistentConfig = AUDIO_CONFIG_BASE_INITIALIZER;
     nonExistentConfig.format = AUDIO_FORMAT_E_AC3;
-    nonExistentConfig.sample_rate = 48000;
+    nonExistentConfig.sample_rate = k48000SamplingRate;
     nonExistentConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
 
     audio_config_base_t msdDirectConfig1 = AUDIO_CONFIG_BASE_INITIALIZER;
     msdDirectConfig1.format = AUDIO_FORMAT_AC3;
-    msdDirectConfig1.sample_rate = 48000;
+    msdDirectConfig1.sample_rate = k48000SamplingRate;
     msdDirectConfig1.channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
 
     audio_config_base_t msdDirectConfig2 = AUDIO_CONFIG_BASE_INITIALIZER;
     msdDirectConfig2.format = AUDIO_FORMAT_IEC60958;
-    msdDirectConfig2.sample_rate = 48000;
+    msdDirectConfig2.sample_rate = k48000SamplingRate;
     msdDirectConfig2.channel_mask = AUDIO_CHANNEL_INDEX_MASK_24;
 
     audio_config_base_t msdNonDirectConfig = AUDIO_CONFIG_BASE_INITIALIZER;
@@ -849,27 +858,27 @@
 
     audio_config_t directConfig = AUDIO_CONFIG_INITIALIZER;
     directConfig.format = AUDIO_FORMAT_DTS;
-    directConfig.sample_rate = 48000;
+    directConfig.sample_rate = k48000SamplingRate;
     directConfig.channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
 
     audio_config_t nonDirectConfig = AUDIO_CONFIG_INITIALIZER;
     nonDirectConfig.format = AUDIO_FORMAT_PCM_16_BIT;
-    nonDirectConfig.sample_rate = 48000;
+    nonDirectConfig.sample_rate = k48000SamplingRate;
     nonDirectConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
 
     audio_config_t nonExistentConfig = AUDIO_CONFIG_INITIALIZER;
     nonExistentConfig.format = AUDIO_FORMAT_E_AC3;
-    nonExistentConfig.sample_rate = 48000;
+    nonExistentConfig.sample_rate = k48000SamplingRate;
     nonExistentConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
 
     audio_config_t msdDirectConfig1 = AUDIO_CONFIG_INITIALIZER;
     msdDirectConfig1.format = AUDIO_FORMAT_AC3;
-    msdDirectConfig1.sample_rate = 48000;
+    msdDirectConfig1.sample_rate = k48000SamplingRate;
     msdDirectConfig1.channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
 
     audio_config_t msdDirectConfig2 = AUDIO_CONFIG_INITIALIZER;
     msdDirectConfig2.format = AUDIO_FORMAT_IEC60958;
-    msdDirectConfig2.sample_rate = 48000;
+    msdDirectConfig2.sample_rate = k48000SamplingRate;
     msdDirectConfig2.channel_mask = AUDIO_CHANNEL_INDEX_MASK_24;
 
     audio_config_t msdNonDirectConfig = AUDIO_CONFIG_INITIALIZER;
@@ -1125,12 +1134,12 @@
     audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
     audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
     getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
-            48000, AUDIO_OUTPUT_FLAG_NONE, &output, &portId, mediaAttr,
+            k48000SamplingRate, AUDIO_OUTPUT_FLAG_NONE, &output, &portId, mediaAttr,
             AUDIO_SESSION_NONE, uid);
     status_t status = mManager->startOutput(portId);
     if (status == DEAD_OBJECT) {
         getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
-                48000, AUDIO_OUTPUT_FLAG_NONE, &output, &portId, mediaAttr,
+                k48000SamplingRate, AUDIO_OUTPUT_FLAG_NONE, &output, &portId, mediaAttr,
                 AUDIO_SESSION_NONE, uid);
         status = mManager->startOutput(portId);
     }
@@ -1171,9 +1180,9 @@
     audio_io_handle_t input = AUDIO_PORT_HANDLE_NONE;
     AttributionSourceState attributionSource = createAttributionSourceState(/*uid=*/ 0);
     audio_config_base_t requestedConfig = {
+            .sample_rate = k48000SamplingRate,
             .channel_mask = AUDIO_CHANNEL_IN_STEREO,
             .format = AUDIO_FORMAT_PCM_16_BIT,
-            .sample_rate = 48000
     };
     audio_config_base_t config = requestedConfig;
     audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
@@ -3179,6 +3188,259 @@
             "low latency");
 }
 
+class AudioPolicyManagerPhoneTest : public AudioPolicyManagerTestWithConfigurationFile {
+protected:
+    std::string getConfigFile() override { return sPhoneConfig; }
+    void testOutputMixPortSelectionForAttr(audio_output_flags_t flags, audio_format_t format,
+            int samplingRate, bool isMusic, const char* expectedMixPortName);
+    void testOutputMixPortSelectionForStream(
+            audio_stream_type_t stream, const char* expectedMixPortName);
+    void verifyMixPortNameAndFlags(audio_io_handle_t output, const char* expectedMixPortName);
+
+    static const std::string sPhoneConfig;
+    static const std::map<std::string, audio_output_flags_t> sMixPortFlags;
+};
+
+const std::string AudioPolicyManagerPhoneTest::sPhoneConfig =
+        AudioPolicyManagerPhoneTest::sExecutableDir + "test_phone_apm_configuration.xml";
+
+// Must be in sync with the contents of the sPhoneConfig file.
+const std::map<std::string, audio_output_flags_t> AudioPolicyManagerPhoneTest::sMixPortFlags = {
+        {"primary output",
+         (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_PRIMARY | AUDIO_OUTPUT_FLAG_FAST)},
+        {"direct", AUDIO_OUTPUT_FLAG_DIRECT},
+        {"deep buffer", AUDIO_OUTPUT_FLAG_DEEP_BUFFER},
+        {"compressed_offload",
+         (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD |
+                                AUDIO_OUTPUT_FLAG_NON_BLOCKING |
+                                AUDIO_OUTPUT_FLAG_GAPLESS_OFFLOAD)},
+        {"raw", (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_RAW | AUDIO_OUTPUT_FLAG_FAST)},
+        {"mmap_no_irq_out",
+         (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_MMAP_NOIRQ)},
+        {"voip_rx", AUDIO_OUTPUT_FLAG_VOIP_RX},
+};
+
+void AudioPolicyManagerPhoneTest::testOutputMixPortSelectionForAttr(
+        audio_output_flags_t flags, audio_format_t format, int samplingRate, bool isMusic,
+        const char* expectedMixPortName) {
+    audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+    audio_io_handle_t output;
+    audio_port_handle_t portId;
+    audio_attributes_t attr = AUDIO_ATTRIBUTES_INITIALIZER;
+    if (isMusic) {
+        attr.content_type = AUDIO_CONTENT_TYPE_MUSIC;
+        attr.usage = AUDIO_USAGE_MEDIA;
+    }
+    getOutputForAttr(&selectedDeviceId, format, AUDIO_CHANNEL_OUT_STEREO, samplingRate, flags,
+            &output, &portId, attr);
+    EXPECT_NO_FATAL_FAILURE(verifyMixPortNameAndFlags(output, expectedMixPortName));
+    mManager->releaseOutput(portId);
+}
+
+void AudioPolicyManagerPhoneTest::testOutputMixPortSelectionForStream(
+        audio_stream_type_t stream, const char* expectedMixPortName) {
+    audio_io_handle_t output = mManager->getOutput(stream);
+    EXPECT_NO_FATAL_FAILURE(verifyMixPortNameAndFlags(output, expectedMixPortName));
+}
+
+void AudioPolicyManagerPhoneTest::verifyMixPortNameAndFlags(audio_io_handle_t output,
+                                                            const char* expectedMixPortName) {
+    ALOGI("%s: checking output %d", __func__, output);
+    sp<SwAudioOutputDescriptor> outDesc = mManager->getOutputs().valueFor(output);
+    ASSERT_NE(nullptr, outDesc.get());
+    audio_port_v7 port = {};
+    outDesc->toAudioPort(&port);
+    EXPECT_EQ(AUDIO_PORT_TYPE_MIX, port.type);
+    EXPECT_EQ(AUDIO_PORT_ROLE_SOURCE, port.role);
+    ASSERT_STREQ(expectedMixPortName, port.name);
+
+    auto iter = sMixPortFlags.find(port.name);
+    ASSERT_NE(iter, sMixPortFlags.end()) << "\"" << port.name << "\" is not in sMixPortFlags";
+    auto actualFlags = mClient->getOpenOutputFlags(output);
+    ASSERT_TRUE(actualFlags.has_value()) << "\"" << port.name << "\" was not opened via client";
+    EXPECT_EQ(*actualFlags, iter->second);
+}
+
+TEST_F(AudioPolicyManagerPhoneTest, InitSuccess) {
+    // SetUp must finish with no assertions.
+}
+
+enum {
+    MIX_PORT_ATTR_EXPECTED_NAME_PARAMETER,
+    MIX_PORT_ATTR_EXPECTED_NAME_WITH_DBFM_PARAMETER,
+    MIX_PORT_ATTR_FLAGS_PARAMETER,
+    MIX_PORT_ATTR_FORMAT_PARAMETER,
+    MIX_PORT_ATTR_SAMPLING_RATE_PARAMETER,
+};
+using MixPortSelectionForAttr =
+        std::tuple<const char*, const char*, audio_output_flags_t, audio_format_t, int>;
+
+class AudioPolicyManagerOutputMixPortForAttrSelectionTest
+    : public AudioPolicyManagerPhoneTest,
+      public testing::WithParamInterface<MixPortSelectionForAttr> {
+};
+
+// There is no easy way to create a flat tuple from tuples via ::testing::Combine.
+// Instead, just run the same selection twice while altering the deep buffer for media setting.
+TEST_P(AudioPolicyManagerOutputMixPortForAttrSelectionTest, SelectPortByFlags) {
+    mConfig->setUseDeepBufferForMediaOverrideForTests(false);
+    ASSERT_NO_FATAL_FAILURE(testOutputMixPortSelectionForAttr(
+                    std::get<MIX_PORT_ATTR_FLAGS_PARAMETER>(GetParam()),
+                    std::get<MIX_PORT_ATTR_FORMAT_PARAMETER>(GetParam()),
+                    std::get<MIX_PORT_ATTR_SAMPLING_RATE_PARAMETER>(GetParam()),
+                    false /*isMusic*/,
+                    std::get<MIX_PORT_ATTR_EXPECTED_NAME_PARAMETER>(GetParam())));
+}
+TEST_P(AudioPolicyManagerOutputMixPortForAttrSelectionTest, SelectPortByFlags_Music) {
+    mConfig->setUseDeepBufferForMediaOverrideForTests(false);
+    ASSERT_NO_FATAL_FAILURE(testOutputMixPortSelectionForAttr(
+                    std::get<MIX_PORT_ATTR_FLAGS_PARAMETER>(GetParam()),
+                    std::get<MIX_PORT_ATTR_FORMAT_PARAMETER>(GetParam()),
+                    std::get<MIX_PORT_ATTR_SAMPLING_RATE_PARAMETER>(GetParam()),
+                    true /*isMusic*/,
+                    std::get<MIX_PORT_ATTR_EXPECTED_NAME_PARAMETER>(GetParam())));
+}
+TEST_P(AudioPolicyManagerOutputMixPortForAttrSelectionTest, SelectPortByFlags_DeepMedia) {
+    mConfig->setUseDeepBufferForMediaOverrideForTests(true);
+    const char* fallbackName = std::get<MIX_PORT_ATTR_EXPECTED_NAME_PARAMETER>(GetParam());
+    ASSERT_NO_FATAL_FAILURE(
+            testOutputMixPortSelectionForAttr(std::get<MIX_PORT_ATTR_FLAGS_PARAMETER>(GetParam()),
+                                       std::get<MIX_PORT_ATTR_FORMAT_PARAMETER>(GetParam()),
+                                       std::get<MIX_PORT_ATTR_SAMPLING_RATE_PARAMETER>(GetParam()),
+                                       false /*isMusic*/,
+                                       std::get<MIX_PORT_ATTR_EXPECTED_NAME_WITH_DBFM_PARAMETER>(
+                                               GetParam()) ?: fallbackName));
+}
+TEST_P(AudioPolicyManagerOutputMixPortForAttrSelectionTest, SelectPortByFlags_DeepMedia_Music) {
+    mConfig->setUseDeepBufferForMediaOverrideForTests(true);
+    const char* fallbackName = std::get<MIX_PORT_ATTR_EXPECTED_NAME_PARAMETER>(GetParam());
+    ASSERT_NO_FATAL_FAILURE(
+            testOutputMixPortSelectionForAttr(std::get<MIX_PORT_ATTR_FLAGS_PARAMETER>(GetParam()),
+                                       std::get<MIX_PORT_ATTR_FORMAT_PARAMETER>(GetParam()),
+                                       std::get<MIX_PORT_ATTR_SAMPLING_RATE_PARAMETER>(GetParam()),
+                                       true /*isMusic*/,
+                                       std::get<MIX_PORT_ATTR_EXPECTED_NAME_WITH_DBFM_PARAMETER>(
+                                               GetParam()) ?: fallbackName));
+}
+
+INSTANTIATE_TEST_CASE_P(AudioPolicyManagerOutputMixPortForAttrSelection,
+        AudioPolicyManagerOutputMixPortForAttrSelectionTest,
+        ::testing::Values(
+                std::make_tuple("primary output", "deep buffer", AUDIO_OUTPUT_FLAG_NONE,
+                        AUDIO_FORMAT_PCM_16_BIT, AudioPolicyManagerTest::k48000SamplingRate),
+                std::make_tuple("primary output", "deep buffer", AUDIO_OUTPUT_FLAG_NONE,
+                        AUDIO_FORMAT_PCM_FLOAT, AudioPolicyManagerTest::k48000SamplingRate),
+                // Note: this goes to "direct" because 384000 > SAMPLE_RATE_HZ_MAX (192000)
+                std::make_tuple("direct", "deep buffer", AUDIO_OUTPUT_FLAG_NONE,
+                        AUDIO_FORMAT_PCM_FLOAT, AudioPolicyManagerTest::k384000SamplingRate),
+                std::make_tuple("primary output", nullptr, AUDIO_OUTPUT_FLAG_FAST,
+                        AUDIO_FORMAT_PCM_16_BIT, AudioPolicyManagerTest::k48000SamplingRate),
+                std::make_tuple("direct", nullptr, AUDIO_OUTPUT_FLAG_DIRECT,
+                        AUDIO_FORMAT_PCM_FLOAT, AudioPolicyManagerTest::k96000SamplingRate),
+                std::make_tuple("direct", nullptr, AUDIO_OUTPUT_FLAG_DIRECT,
+                        AUDIO_FORMAT_PCM_FLOAT, AudioPolicyManagerTest::k384000SamplingRate),
+                std::make_tuple("deep buffer", nullptr, AUDIO_OUTPUT_FLAG_DEEP_BUFFER,
+                        AUDIO_FORMAT_PCM_16_BIT, AudioPolicyManagerTest::k48000SamplingRate),
+                std::make_tuple("deep buffer", nullptr, AUDIO_OUTPUT_FLAG_DEEP_BUFFER,
+                        AUDIO_FORMAT_PCM_FLOAT, AudioPolicyManagerTest::k384000SamplingRate),
+                std::make_tuple("compressed_offload", nullptr,
+                        (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD |
+                                AUDIO_OUTPUT_FLAG_NON_BLOCKING),
+                        AUDIO_FORMAT_MP3, AudioPolicyManagerTest::k48000SamplingRate),
+                std::make_tuple("raw", nullptr,
+                        AUDIO_OUTPUT_FLAG_RAW, AUDIO_FORMAT_PCM_32_BIT,
+                        AudioPolicyManagerTest::k48000SamplingRate),
+                std::make_tuple("mmap_no_irq_out", nullptr,
+                        (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DIRECT |
+                                AUDIO_OUTPUT_FLAG_MMAP_NOIRQ),
+                        AUDIO_FORMAT_PCM_FLOAT, AudioPolicyManagerTest::k48000SamplingRate),
+                std::make_tuple("mmap_no_irq_out", nullptr,
+                        (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DIRECT |
+                                AUDIO_OUTPUT_FLAG_MMAP_NOIRQ),
+                        AUDIO_FORMAT_PCM_FLOAT, AudioPolicyManagerTest::k384000SamplingRate),
+                std::make_tuple("voip_rx", nullptr, AUDIO_OUTPUT_FLAG_VOIP_RX,
+                        AUDIO_FORMAT_PCM_16_BIT, AudioPolicyManagerTest::k48000SamplingRate)),
+        [](const ::testing::TestParamInfo<MixPortSelectionForAttr>& info) {
+            static const std::string flagPrefix = "AUDIO_OUTPUT_FLAG_";
+            static const std::string formatPrefix = "AUDIO_FORMAT_";
+            std::string flags;
+            TypeConverter<OutputFlagTraits>::maskToString(
+                    std::get<MIX_PORT_ATTR_FLAGS_PARAMETER>(info.param), flags, "__");
+            size_t index = 0;
+            while (true) {
+                index = flags.rfind(flagPrefix);
+                if (index == std::string::npos) break;
+                flags.erase(index, flagPrefix.length());
+            }
+            std::string format;
+            TypeConverter<FormatTraits>::toString(
+                    std::get<MIX_PORT_ATTR_FORMAT_PARAMETER>(info.param), format);
+            if (size_t index = format.find(formatPrefix); index != std::string::npos) {
+                format.erase(index, formatPrefix.length());
+            }
+            return flags + "__" + format + "__" +
+                    std::to_string(std::get<MIX_PORT_ATTR_SAMPLING_RATE_PARAMETER>(info.param));
+        }
+);
+
+
+enum {
+    MIX_PORT_STRM_EXPECTED_NAME_PARAMETER,
+    MIX_PORT_STRM_EXPECTED_NAME_WITH_DBFM_PARAMETER,
+    MIX_PORT_STRM_STREAM_PARAMETER,
+};
+using MixPortSelectionForStream =
+        std::tuple<const char*, const char*, audio_stream_type_t>;
+
+class AudioPolicyManagerOutputMixPortForStreamSelectionTest
+    : public AudioPolicyManagerPhoneTest,
+      public testing::WithParamInterface<MixPortSelectionForStream> {
+};
+
+// There is no easy way to create a flat tuple from tuples via ::testing::Combine.
+// Instead, just run the same selection twice while altering the deep buffer for media setting.
+TEST_P(AudioPolicyManagerOutputMixPortForStreamSelectionTest, SelectPort_NoDBFM) {
+    mConfig->setUseDeepBufferForMediaOverrideForTests(false);
+    ASSERT_NO_FATAL_FAILURE(testOutputMixPortSelectionForStream(
+                    std::get<MIX_PORT_STRM_STREAM_PARAMETER>(GetParam()),
+                    std::get<MIX_PORT_STRM_EXPECTED_NAME_PARAMETER>(GetParam())));
+}
+TEST_P(AudioPolicyManagerOutputMixPortForStreamSelectionTest, SelectPort_WithDBFM) {
+    mConfig->setUseDeepBufferForMediaOverrideForTests(true);
+    const char* fallbackName = std::get<MIX_PORT_STRM_EXPECTED_NAME_PARAMETER>(GetParam());
+    ASSERT_NO_FATAL_FAILURE(testOutputMixPortSelectionForStream(
+                    std::get<MIX_PORT_STRM_STREAM_PARAMETER>(GetParam()),
+                    std::get<MIX_PORT_STRM_EXPECTED_NAME_WITH_DBFM_PARAMETER>(
+                            GetParam()) ?: fallbackName));
+}
+
+INSTANTIATE_TEST_CASE_P(
+        AudioPolicyManagerOutputMixPortForStreamSelection,
+        AudioPolicyManagerOutputMixPortForStreamSelectionTest,
+        ::testing::Values(std::make_tuple("primary output", nullptr, AUDIO_STREAM_DEFAULT),
+                          std::make_tuple("primary output", nullptr, AUDIO_STREAM_SYSTEM),
+                          std::make_tuple("primary output", nullptr, AUDIO_STREAM_RING),
+                          std::make_tuple("primary output", "deep buffer", AUDIO_STREAM_MUSIC),
+                          std::make_tuple("primary output", nullptr, AUDIO_STREAM_ALARM),
+                          std::make_tuple("primary output", nullptr, AUDIO_STREAM_NOTIFICATION),
+                          std::make_tuple("primary output", nullptr, AUDIO_STREAM_BLUETOOTH_SCO),
+                          std::make_tuple("primary output", nullptr, AUDIO_STREAM_ENFORCED_AUDIBLE),
+                          std::make_tuple("primary output", nullptr, AUDIO_STREAM_DTMF),
+                          std::make_tuple("primary output", nullptr, AUDIO_STREAM_TTS),
+                          std::make_tuple("primary output", nullptr, AUDIO_STREAM_ACCESSIBILITY),
+                          std::make_tuple("primary output", nullptr, AUDIO_STREAM_ASSISTANT)),
+        [](const ::testing::TestParamInfo<MixPortSelectionForStream>& info) {
+            static const std::string streamPrefix = "AUDIO_STREAM_";
+            std::string stream;
+            TypeConverter<StreamTraits>::toString(
+                    std::get<MIX_PORT_STRM_STREAM_PARAMETER>(info.param), stream);
+            if (size_t index = stream.find(streamPrefix); index != std::string::npos) {
+                stream.erase(index, streamPrefix.length());
+            }
+            return stream;
+        }
+);
+
 class AudioPolicyManagerDynamicHwModulesTest : public AudioPolicyManagerTestWithConfigurationFile {
 protected:
     void SetUpManagerConfig() override;
@@ -3331,7 +3593,7 @@
     audio_io_handle_t input = AUDIO_PORT_HANDLE_NONE;
     ASSERT_NO_FATAL_FAILURE(getInputForAttr(attr, &input, AUDIO_SESSION_NONE, 1, &selectedDeviceId,
                                             AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO,
-                                            48000));
+                                            k48000SamplingRate));
     auto selectedDevice = availableDevices.getDeviceFromId(selectedDeviceId);
     ASSERT_NE(nullptr, selectedDevice);
 
@@ -3352,7 +3614,7 @@
     input = AUDIO_PORT_HANDLE_NONE;
     ASSERT_NO_FATAL_FAILURE(getInputForAttr(attr, &input, AUDIO_SESSION_NONE, 1, &selectedDeviceId,
                                             AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO,
-                                            48000));
+                                            k48000SamplingRate));
     ASSERT_EQ(preferredDevice, availableDevices.getDeviceFromId(selectedDeviceId));
 
     // After clearing preferred device for capture preset, the selected device for input should be
@@ -3363,7 +3625,7 @@
     input = AUDIO_PORT_HANDLE_NONE;
     ASSERT_NO_FATAL_FAILURE(getInputForAttr(attr, &input, AUDIO_SESSION_NONE, 1, &selectedDeviceId,
                                             AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO,
-                                            48000));
+                                            k48000SamplingRate));
     ASSERT_EQ(selectedDevice, availableDevices.getDeviceFromId(selectedDeviceId));
 
     ASSERT_EQ(NO_ERROR, mManager->setDeviceConnectionState(
@@ -3389,7 +3651,7 @@
     audio_io_handle_t input = AUDIO_PORT_HANDLE_NONE;
     ASSERT_NO_FATAL_FAILURE(getInputForAttr(attr, &input, AUDIO_SESSION_NONE, 1, &selectedDeviceId,
                                             AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO,
-                                            48000));
+                                            k48000SamplingRate));
     auto selectedDevice = availableDevices.getDeviceFromId(selectedDeviceId);
     ASSERT_NE(nullptr, selectedDevice);
 
@@ -3402,7 +3664,7 @@
     input = AUDIO_PORT_HANDLE_NONE;
     ASSERT_NO_FATAL_FAILURE(getInputForAttr(attr, &input, AUDIO_SESSION_NONE, 1,
                                             &selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT,
-                                            AUDIO_CHANNEL_IN_STEREO, 48000));
+                                            AUDIO_CHANNEL_IN_STEREO, k48000SamplingRate));
     ASSERT_NE(selectedDevice, availableDevices.getDeviceFromId(selectedDeviceId));
 
     // After clearing disabled device for capture preset, the selected device for input should be
@@ -3413,7 +3675,7 @@
     input = AUDIO_PORT_HANDLE_NONE;
     ASSERT_NO_FATAL_FAILURE(getInputForAttr(attr, &input, AUDIO_SESSION_NONE, 1, &selectedDeviceId,
                                             AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO,
-                                            48000));
+                                            k48000SamplingRate));
     ASSERT_EQ(selectedDevice, availableDevices.getDeviceFromId(selectedDeviceId));
 
     ASSERT_EQ(NO_ERROR, mManager->setDeviceConnectionState(
@@ -3487,7 +3749,7 @@
     audio_port_handle_t routedPortId = devicePort.id;
     ASSERT_NO_FATAL_FAILURE(getInputForAttr(attr, &inputClientHandle, session, 1, &routedPortId,
                                             AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO,
-                                            48000, AUDIO_INPUT_FLAG_NONE, &portId));
+                                            k48000SamplingRate, AUDIO_INPUT_FLAG_NONE, &portId));
     ASSERT_EQ(devicePort.id, routedPortId);
     auto selectedDevice = availableDevices.getDeviceFromId(routedPortId);
     ASSERT_NE(nullptr, selectedDevice);
@@ -3516,7 +3778,7 @@
     // effect attached again
     ASSERT_NO_FATAL_FAILURE(getInputForAttr(attr, &inputClientHandle, session, 1, &routedPortId,
                                             AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO,
-                                            48000));
+                                            k48000SamplingRate));
 
     // unregister effect should succeed since effect shall have been restore on the client session
     ASSERT_EQ(NO_ERROR, mManager->unregisterEffect(effectId));
diff --git a/services/audiopolicy/tests/resources/Android.bp b/services/audiopolicy/tests/resources/Android.bp
index 1c191f5..8e7a697 100644
--- a/services/audiopolicy/tests/resources/Android.bp
+++ b/services/audiopolicy/tests/resources/Android.bp
@@ -11,11 +11,16 @@
 filegroup {
     name: "audiopolicytest_configuration_files",
     srcs: [
+        "engine/test_audio_policy_engine_configuration.xml",
+        "engine/test_audio_policy_engine_default_stream_volumes.xml",
+        "engine/test_audio_policy_engine_product_strategies.xml",
+        "engine/test_audio_policy_engine_stream_volumes.xml",
         "test_audio_policy_configuration.xml",
         "test_audio_policy_configuration_bluetooth.xml",
         "test_audio_policy_primary_only_configuration.xml",
         "test_car_ap_atmos_offload_configuration.xml",
         "test_invalid_audio_policy_configuration.xml",
+        "test_phone_apm_configuration.xml",
         "test_settop_box_surround_configuration.xml",
         "test_tv_apm_configuration.xml",
     ],
diff --git a/services/audiopolicy/tests/resources/engine/test_audio_policy_engine_configuration.xml b/services/audiopolicy/tests/resources/engine/test_audio_policy_engine_configuration.xml
new file mode 100644
index 0000000..dc2e7af
--- /dev/null
+++ b/services/audiopolicy/tests/resources/engine/test_audio_policy_engine_configuration.xml
@@ -0,0 +1,24 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (C) 2024 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.
+     -->
+
+<configuration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+
+    <xi:include href="test_audio_policy_engine_product_strategies.xml"/>
+    <xi:include href="test_audio_policy_engine_stream_volumes.xml"/>
+    <xi:include href="test_audio_policy_engine_default_stream_volumes.xml"/>
+
+</configuration>
+
diff --git a/services/audiopolicy/tests/resources/engine/test_audio_policy_engine_default_stream_volumes.xml b/services/audiopolicy/tests/resources/engine/test_audio_policy_engine_default_stream_volumes.xml
new file mode 100644
index 0000000..d184cb5
--- /dev/null
+++ b/services/audiopolicy/tests/resources/engine/test_audio_policy_engine_default_stream_volumes.xml
@@ -0,0 +1,136 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- Copyright (C) 2024 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.
+-->
+<!-- Default Volume Tables included by Audio Policy Configuration file -->
+<!-- Full Default Volume table for all device category -->
+<volumes>
+    <reference name="FULL_SCALE_VOLUME_CURVE">
+    <!-- Full Scale reference Volume Curve -->
+        <point>0,0</point>
+        <point>100,0</point>
+    </reference>
+    <reference name="SILENT_VOLUME_CURVE">
+        <point>0,-9600</point>
+        <point>100,-9600</point>
+    </reference>
+    <reference name="DEFAULT_SYSTEM_VOLUME_CURVE">
+    <!-- Default System reference Volume Curve -->
+        <point>1,-2400</point>
+        <point>33,-1800</point>
+        <point>66,-1200</point>
+        <point>100,-600</point>
+    </reference>
+    <reference name="DEFAULT_MEDIA_VOLUME_CURVE">
+    <!-- Default Media reference Volume Curve -->
+        <point>1,-5800</point>
+        <point>20,-4000</point>
+        <point>60,-1700</point>
+        <point>100,0</point>
+    </reference>
+    <reference name="DEFAULT_DEVICE_CATEGORY_HEADSET_VOLUME_CURVE">
+    <!--Default Volume Curve -->
+        <point>1,-4950</point>
+        <point>33,-3350</point>
+        <point>66,-1700</point>
+        <point>100,0</point>
+    </reference>
+    <reference name="DEFAULT_DEVICE_CATEGORY_SPEAKER_VOLUME_CURVE">
+    <!-- Default is Speaker Media Volume Curve -->
+        <point>1,-5800</point>
+        <point>20,-4000</point>
+        <point>60,-1700</point>
+        <point>100,0</point>
+    </reference>
+    <reference name="DEFAULT_DEVICE_CATEGORY_SPEAKER_SYSTEM_VOLUME_CURVE">
+    <!-- Default is Speaker System Volume Curve -->
+        <point>1,-4680</point>
+        <point>42,-2070</point>
+        <point>85,-540</point>
+        <point>100,0</point>
+    </reference>
+    <reference name="DEFAULT_DEVICE_CATEGORY_EARPIECE_VOLUME_CURVE">
+    <!--Default Volume Curve -->
+        <point>1,-4950</point>
+        <point>33,-3350</point>
+        <point>66,-1700</point>
+        <point>100,0</point>
+    </reference>
+    <reference name="DEFAULT_DEVICE_CATEGORY_EXT_MEDIA_VOLUME_CURVE">
+    <!-- Default is Ext Media System Volume Curve -->
+        <point>1,-5800</point>
+        <point>20,-4000</point>
+        <point>60,-2100</point>
+        <point>100,-1000</point>
+    </reference>
+    <reference name="DEFAULT_HEARING_AID_VOLUME_CURVE">
+    <!-- Default Hearing Aid Volume Curve -->
+        <point>1,-12700</point>
+        <point>20,-8000</point>
+        <point>60,-4000</point>
+        <point>100,0</point>
+    </reference>
+    <!-- **************************************************************** -->
+    <!-- Non-mutable default volume curves:                               -->
+    <!--     * first point is always for index 0                          -->
+    <!--     * attenuation is small enough that stream can still be heard -->
+    <reference name="DEFAULT_NON_MUTABLE_VOLUME_CURVE">
+    <!-- Default non-mutable reference Volume Curve -->
+    <!--        based on DEFAULT_MEDIA_VOLUME_CURVE -->
+        <point>0,-5800</point>
+        <point>20,-4000</point>
+        <point>60,-1700</point>
+        <point>100,0</point>
+    </reference>
+    <reference name="DEFAULT_NON_MUTABLE_HEADSET_VOLUME_CURVE">
+    <!--Default non-mutable Volume Curve for headset -->
+    <!--    based on DEFAULT_DEVICE_CATEGORY_HEADSET_VOLUME_CURVE -->
+        <point>0,-4950</point>
+        <point>33,-3350</point>
+        <point>66,-1700</point>
+        <point>100,0</point>
+    </reference>
+    <reference name="DEFAULT_NON_MUTABLE_SPEAKER_VOLUME_CURVE">
+    <!-- Default non-mutable Speaker Volume Curve -->
+    <!--    based on DEFAULT_DEVICE_CATEGORY_SPEAKER_VOLUME_CURVE -->
+        <point>0,-5800</point>
+        <point>20,-4000</point>
+        <point>60,-1700</point>
+        <point>100,0</point>
+    </reference>
+    <reference name="DEFAULT_NON_MUTABLE_EARPIECE_VOLUME_CURVE">
+    <!--Default non-mutable Volume Curve -->
+    <!--    based on DEFAULT_DEVICE_CATEGORY_EARPIECE_VOLUME_CURVE -->
+        <point>0,-4950</point>
+        <point>33,-3350</point>
+        <point>66,-1700</point>
+        <point>100,0</point>
+    </reference>
+    <reference name="DEFAULT_NON_MUTABLE_EXT_VOLUME_CURVE">
+    <!-- Default non-mutable Ext Media System Volume Curve -->
+    <!--     based on DEFAULT_DEVICE_CATEGORY_EXT_MEDIA_VOLUME_CURVE -->
+        <point>0,-5800</point>
+        <point>20,-4000</point>
+        <point>60,-2100</point>
+        <point>100,-1000</point>
+    </reference>
+    <reference name="DEFAULT_NON_MUTABLE_HEARING_AID_VOLUME_CURVE">
+    <!-- Default non-mutable Hearing Aid Volume Curve -->
+    <!--     based on DEFAULT_HEARING_AID_VOLUME_CURVE -->
+        <point>0,-12700</point>
+        <point>20,-8000</point>
+        <point>60,-4000</point>
+        <point>100,0</point>
+    </reference>
+</volumes>
diff --git a/services/audiopolicy/tests/resources/engine/test_audio_policy_engine_product_strategies.xml b/services/audiopolicy/tests/resources/engine/test_audio_policy_engine_product_strategies.xml
new file mode 100644
index 0000000..58e7152
--- /dev/null
+++ b/services/audiopolicy/tests/resources/engine/test_audio_policy_engine_product_strategies.xml
@@ -0,0 +1,98 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (C) 2024 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.
+     -->
+
+<ProductStrategies>
+
+    <!-- "hidden strategies" like TTS, enforced audible:
+            Shall we expose them here or keep it hard coded -->
+
+    <!-- Used to identify the volume of audio streams for enforced system sounds in certain
+         countries (e.g. camera in Japan)
+         This strategy will only have higher priority than phone if force for system is set to
+         enforced. -->
+
+    <ProductStrategy name="STRATEGY_PHONE">
+        <AttributesGroup streamType="AUDIO_STREAM_VOICE_CALL" volumeGroup="voice_call">
+            <Attributes> <Usage value="AUDIO_USAGE_VOICE_COMMUNICATION"/> </Attributes>
+        </AttributesGroup>
+        <AttributesGroup streamType="AUDIO_STREAM_BLUETOOTH_SCO" volumeGroup="bluetooth_sco">
+            <Attributes> <Flags value="AUDIO_FLAG_SCO"/> </Attributes>
+        </AttributesGroup>
+    </ProductStrategy>
+
+    <ProductStrategy name="STRATEGY_SONIFICATION">
+        <AttributesGroup streamType="AUDIO_STREAM_RING" volumeGroup="ring">
+            <Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE"/> </Attributes>
+        </AttributesGroup>
+        <AttributesGroup streamType="AUDIO_STREAM_ALARM" volumeGroup="alarm">
+            <Attributes> <Usage value="AUDIO_USAGE_ALARM"/> </Attributes>
+        </AttributesGroup>
+    </ProductStrategy>
+
+    <ProductStrategy name="STRATEGY_ENFORCED_AUDIBLE">
+        <AttributesGroup streamType="AUDIO_STREAM_ENFORCED_AUDIBLE" volumeGroup="enforced_audible">
+            <Attributes> <Flags value="AUDIO_FLAG_AUDIBILITY_ENFORCED"/> </Attributes>
+        </AttributesGroup>
+    </ProductStrategy>
+
+    <ProductStrategy name="STRATEGY_ACCESSIBILITY">
+        <AttributesGroup streamType="AUDIO_STREAM_ACCESSIBILITY" volumeGroup="accessibility">
+            <Attributes> <Usage value="AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY"/> </Attributes>
+        </AttributesGroup>
+    </ProductStrategy>
+
+    <ProductStrategy name="STRATEGY_SONIFICATION_RESPECTFUL">
+        <AttributesGroup streamType="AUDIO_STREAM_NOTIFICATION" volumeGroup="notification">
+            <Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION"/> </Attributes>
+            <Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION_EVENT"/> </Attributes>
+        </AttributesGroup>
+    </ProductStrategy>
+
+    <ProductStrategy name="STRATEGY_MEDIA">
+        <AttributesGroup streamType="AUDIO_STREAM_ASSISTANT" volumeGroup="assistant">
+            <Attributes>
+                <ContentType value="AUDIO_CONTENT_TYPE_SPEECH"/>
+                <Usage value="AUDIO_USAGE_ASSISTANT"/>
+            </Attributes>
+        </AttributesGroup>
+         <AttributesGroup streamType="AUDIO_STREAM_MUSIC" volumeGroup="music">
+            <Attributes> <Usage value="AUDIO_USAGE_MEDIA"/> </Attributes>
+            <Attributes> <Usage value="AUDIO_USAGE_GAME"/> </Attributes>
+            <Attributes> <Usage value="AUDIO_USAGE_ASSISTANT"/> </Attributes>
+            <Attributes> <Usage value="AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE"/> </Attributes>
+            <Attributes></Attributes>
+        </AttributesGroup>
+        <AttributesGroup streamType="AUDIO_STREAM_SYSTEM" volumeGroup="system">
+            <Attributes> <Usage value="AUDIO_USAGE_ASSISTANCE_SONIFICATION"/> </Attributes>
+        </AttributesGroup>
+    </ProductStrategy>
+
+    <ProductStrategy name="STRATEGY_DTMF">
+        <AttributesGroup streamType="AUDIO_STREAM_DTMF" volumeGroup="dtmf">
+            <Attributes> <Usage value="AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING"/> </Attributes>
+        </AttributesGroup>
+    </ProductStrategy>
+
+    <!-- Used to identify the volume of audio streams exclusively transmitted through the  speaker
+         (TTS) of the device -->
+    <ProductStrategy name="STRATEGY_TRANSMITTED_THROUGH_SPEAKER">
+        <AttributesGroup streamType="AUDIO_STREAM_TTS" volumeGroup="tts">
+            <Attributes> <Flags value="AUDIO_FLAG_BEACON"/> </Attributes>
+        </AttributesGroup>
+    </ProductStrategy>
+
+</ProductStrategies>
+
diff --git a/services/audiopolicy/tests/resources/engine/test_audio_policy_engine_stream_volumes.xml b/services/audiopolicy/tests/resources/engine/test_audio_policy_engine_stream_volumes.xml
new file mode 100644
index 0000000..af517cf
--- /dev/null
+++ b/services/audiopolicy/tests/resources/engine/test_audio_policy_engine_stream_volumes.xml
@@ -0,0 +1,221 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- Copyright (C) 2024 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.
+-->
+<!-- Volume section defines a volume curve for a given use case and device category.
+It contains a list of points of this curve expressing the attenuation in Millibels for a given
+volume index from 0 to 100.
+<volume deviceCategory=””>
+<point>0,-9600</point>
+<point>100,0</point>
+</volume>
+-->
+
+<volumeGroups>
+    <volumeGroup>
+        <name>voice_call</name>
+        <indexMin>1</indexMin>
+        <indexMax>7</indexMax>
+        <volume deviceCategory="DEVICE_CATEGORY_HEADSET">
+            <point>0,-4200</point>
+            <point>33,-2800</point>
+            <point>66,-1400</point>
+            <point>100,0</point>
+        </volume>
+        <volume deviceCategory="DEVICE_CATEGORY_SPEAKER">
+            <point>0,-2400</point>
+            <point>33,-1600</point>
+            <point>66,-800</point>
+            <point>100,0</point>
+        </volume>
+        <volume deviceCategory="DEVICE_CATEGORY_EARPIECE">
+            <point>0,-2700</point>
+            <point>33,-1800</point>
+            <point>66,-900</point>
+            <point>100,0</point>
+        </volume>
+        <volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_HEARING_AID" ref="DEFAULT_HEARING_AID_VOLUME_CURVE"/>
+    </volumeGroup>
+
+    <volumeGroup>
+        <name>system</name>
+        <indexMin>0</indexMin>
+        <indexMax>7</indexMax>
+        <volume deviceCategory="DEVICE_CATEGORY_HEADSET">
+            <point>1,-3000</point>
+            <point>33,-2600</point>
+            <point>66,-2200</point>
+            <point>100,-1800</point>
+        </volume>
+        <volume deviceCategory="DEVICE_CATEGORY_SPEAKER">
+            <point>1,-5100</point>
+            <point>57,-2800</point>
+            <point>71,-2500</point>
+            <point>85,-2300</point>
+            <point>100,-2100</point>
+        </volume>
+        <!--volume deviceCategory="DEVICE_CATEGORY_SPEAKER" ref="DEFAULT_SYSTEM_VOLUME_CURVE"/-->
+        <volume deviceCategory="DEVICE_CATEGORY_EARPIECE" ref="DEFAULT_SYSTEM_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="DEFAULT_DEVICE_CATEGORY_EXT_MEDIA_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_HEARING_AID" ref="DEFAULT_HEARING_AID_VOLUME_CURVE"/>
+    </volumeGroup>
+
+    <volumeGroup>
+        <name>ring</name>
+        <indexMin>0</indexMin>
+        <indexMax>7</indexMax>
+        <volume deviceCategory="DEVICE_CATEGORY_HEADSET" ref="DEFAULT_DEVICE_CATEGORY_HEADSET_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_SPEAKER" ref="DEFAULT_DEVICE_CATEGORY_SPEAKER_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_EARPIECE" ref="DEFAULT_DEVICE_CATEGORY_EARPIECE_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="DEFAULT_DEVICE_CATEGORY_EXT_MEDIA_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_HEARING_AID" ref="DEFAULT_HEARING_AID_VOLUME_CURVE"/>
+    </volumeGroup>
+
+    <volumeGroup>
+        <name>music</name>
+        <indexMin>0</indexMin>
+        <indexMax>25</indexMax>
+        <volume deviceCategory="DEVICE_CATEGORY_HEADSET" ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_SPEAKER" ref="DEFAULT_DEVICE_CATEGORY_SPEAKER_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_EARPIECE" ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_HEARING_AID"  ref="DEFAULT_HEARING_AID_VOLUME_CURVE"/>
+    </volumeGroup>
+
+    <volumeGroup>
+        <name>alarm</name>
+        <indexMin>1</indexMin>
+        <indexMax>7</indexMax>
+        <volume deviceCategory="DEVICE_CATEGORY_HEADSET" ref="DEFAULT_NON_MUTABLE_HEADSET_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_SPEAKER" ref="DEFAULT_NON_MUTABLE_SPEAKER_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_EARPIECE" ref="DEFAULT_NON_MUTABLE_EARPIECE_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="DEFAULT_NON_MUTABLE_EXT_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_HEARING_AID" ref="DEFAULT_NON_MUTABLE_HEARING_AID_VOLUME_CURVE"/>
+    </volumeGroup>
+
+    <volumeGroup>
+        <name>notification</name>
+        <indexMin>0</indexMin>
+        <indexMax>7</indexMax>
+        <volume deviceCategory="DEVICE_CATEGORY_HEADSET" ref="DEFAULT_DEVICE_CATEGORY_HEADSET_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_SPEAKER" ref="DEFAULT_DEVICE_CATEGORY_SPEAKER_SYSTEM_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_EARPIECE" ref="DEFAULT_DEVICE_CATEGORY_EARPIECE_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="DEFAULT_DEVICE_CATEGORY_EXT_MEDIA_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_HEARING_AID" ref="DEFAULT_DEVICE_CATEGORY_HEADSET_VOLUME_CURVE"/>
+    </volumeGroup>
+
+    <volumeGroup>
+        <name>bluetooth_sco</name>
+        <indexMin>0</indexMin>
+        <indexMax>15</indexMax>
+        <volume deviceCategory="DEVICE_CATEGORY_HEADSET">
+            <point>0,-4200</point>
+            <point>33,-2800</point>
+            <point>66,-1400</point>
+            <point>100,0</point>
+        </volume>
+        <volume deviceCategory="DEVICE_CATEGORY_SPEAKER">
+            <point>0,-2400</point>
+            <point>33,-1600</point>
+            <point>66,-800</point>
+            <point>100,0</point>
+        </volume>
+        <volume deviceCategory="DEVICE_CATEGORY_EARPIECE">
+            <point>0,-4200</point>
+            <point>33,-2800</point>
+            <point>66,-1400</point>
+            <point>100,0</point>
+        </volume>
+        <volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_HEARING_AID" ref="DEFAULT_HEARING_AID_VOLUME_CURVE"/>
+    </volumeGroup>
+
+    <volumeGroup>
+        <name>enforced_audible</name>
+        <indexMin>0</indexMin>
+        <indexMax>7</indexMax>
+        <volume deviceCategory="DEVICE_CATEGORY_HEADSET">
+            <point>1,-3000</point>
+            <point>33,-2600</point>
+            <point>66,-2200</point>
+            <point>100,-1800</point>
+        </volume>
+        <volume deviceCategory="DEVICE_CATEGORY_SPEAKER">
+            <point>1,-3400</point>
+            <point>71,-2400</point>
+            <point>100,-2000</point>
+        </volume>
+        <!--volume deviceCategory="DEVICE_CATEGORY_SPEAKER" ref="DEFAULT_SYSTEM_VOLUME_CURVE"/-->
+        <volume deviceCategory="DEVICE_CATEGORY_EARPIECE" ref="DEFAULT_SYSTEM_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="DEFAULT_DEVICE_CATEGORY_EXT_MEDIA_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_HEARING_AID" ref="DEFAULT_HEARING_AID_VOLUME_CURVE"/>
+    </volumeGroup>
+
+    <volumeGroup>
+        <name>dtmf</name>
+        <indexMin>0</indexMin>
+        <indexMax>15</indexMax>
+        <volume deviceCategory="DEVICE_CATEGORY_HEADSET">
+            <point>1,-3000</point>
+            <point>33,-2600</point>
+            <point>66,-2200</point>
+            <point>100,-1800</point>
+        </volume>
+        <volume deviceCategory="DEVICE_CATEGORY_SPEAKER">
+            <point>1,-4000</point>
+            <point>71,-2400</point>
+            <point>100,-1400</point>
+        </volume>
+        <!--volume deviceCategory="DEVICE_CATEGORY_SPEAKER" ref="DEFAULT_SYSTEM_VOLUME_CURVE"/-->
+        <volume deviceCategory="DEVICE_CATEGORY_EARPIECE" ref="DEFAULT_SYSTEM_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="DEFAULT_DEVICE_CATEGORY_EXT_MEDIA_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_HEARING_AID" ref="DEFAULT_HEARING_AID_VOLUME_CURVE"/>
+    </volumeGroup>
+
+    <volumeGroup>
+        <name>tts</name>
+        <indexMin>0</indexMin>
+        <indexMax>15</indexMax>
+        <volume deviceCategory="DEVICE_CATEGORY_HEADSET" ref="SILENT_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_SPEAKER" ref="FULL_SCALE_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_EARPIECE" ref="SILENT_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="SILENT_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_HEARING_AID" ref="SILENT_VOLUME_CURVE"/>
+    </volumeGroup>
+
+    <volumeGroup>
+        <name>accessibility</name>
+        <indexMin>1</indexMin>
+        <indexMax>15</indexMax>
+        <volume deviceCategory="DEVICE_CATEGORY_HEADSET" ref="DEFAULT_NON_MUTABLE_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_SPEAKER" ref="DEFAULT_NON_MUTABLE_SPEAKER_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_EARPIECE" ref="DEFAULT_NON_MUTABLE_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="DEFAULT_NON_MUTABLE_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_HEARING_AID" ref="DEFAULT_NON_MUTABLE_HEARING_AID_VOLUME_CURVE"/>
+    </volumeGroup>
+
+    <volumeGroup>
+        <name>assistant</name>
+        <indexMin>0</indexMin>
+        <indexMax>15</indexMax>
+        <volume deviceCategory="DEVICE_CATEGORY_HEADSET" ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_SPEAKER" ref="DEFAULT_DEVICE_CATEGORY_SPEAKER_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_EARPIECE" ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
+        <volume deviceCategory="DEVICE_CATEGORY_HEARING_AID"  ref="DEFAULT_HEARING_AID_VOLUME_CURVE"/>
+    </volumeGroup>
+
+</volumeGroups>
+
diff --git a/services/audiopolicy/tests/resources/test_phone_apm_configuration.xml b/services/audiopolicy/tests/resources/test_phone_apm_configuration.xml
new file mode 100644
index 0000000..efe1400
--- /dev/null
+++ b/services/audiopolicy/tests/resources/test_phone_apm_configuration.xml
@@ -0,0 +1,279 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (C) 2024 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.
+-->
+<audioPolicyConfiguration version="7.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+    <globalConfiguration speaker_drc_enabled="false" call_screen_mode_supported="true" />
+    <modules>
+        <!-- Primary Audio HAL -->
+        <module name="primary" halVersion="2.0">
+            <attachedDevices>
+                <item>Speaker</item>
+                <item>Speaker Safe</item>
+                <item>Earpiece</item>
+                <item>Built-In Mic</item>
+                <item>Built-In Back Mic</item>
+                <item>Telephony Tx</item>
+                <item>Voice Call And Telephony Rx</item>
+                <item>Echo Ref In</item>
+            </attachedDevices>
+            <defaultOutputDevice>Speaker</defaultOutputDevice>
+            <mixPorts>
+                <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_PRIMARY AUDIO_OUTPUT_FLAG_FAST"
+                         recommendedMuteDurationMs="40">
+                    <profile name="" format="AUDIO_FORMAT_PCM_FLOAT"
+                             samplingRates="48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="direct" role="source" flags="AUDIO_OUTPUT_FLAG_DIRECT"
+                         recommendedMuteDurationMs="40">
+                    <profile name="" format="AUDIO_FORMAT_PCM_FLOAT"
+                             samplingRates="48000 96000 384000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="deep buffer" role="source" flags="AUDIO_OUTPUT_FLAG_DEEP_BUFFER">
+                    <profile name="" format="AUDIO_FORMAT_PCM_FLOAT"
+                             samplingRates="48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="compressed_offload" role="source"
+                         flags="AUDIO_OUTPUT_FLAG_DIRECT AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD AUDIO_OUTPUT_FLAG_NON_BLOCKING AUDIO_OUTPUT_FLAG_GAPLESS_OFFLOAD">
+                    <profile name="" format="AUDIO_FORMAT_MP3"
+                             samplingRates="8000 16000 24000 32000 44100 48000 96000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_MONO"/>
+                    <profile name="" format="AUDIO_FORMAT_AAC_LC"
+                             samplingRates="8000 16000 24000 32000 44100 48000 96000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_MONO"/>
+                    <profile name="" format="AUDIO_FORMAT_AAC_HE_V1"
+                             samplingRates="8000 16000 24000 32000 44100 48000 96000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_MONO"/>
+                    <profile name="" format="AUDIO_FORMAT_AAC_HE_V2"
+                             samplingRates="8000 16000 24000 32000 44100 48000 96000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_MONO"/>
+                    <profile name="" format="AUDIO_FORMAT_OPUS"
+                             samplingRates="48000 96000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="haptic" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000 96000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO_HAPTIC_AB" />
+                </mixPort>
+                <mixPort name="raw" role="source" flags="AUDIO_OUTPUT_FLAG_RAW AUDIO_OUTPUT_FLAG_FAST">
+                    <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
+                             samplingRates="48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="mmap_no_irq_out" role="source" flags="AUDIO_OUTPUT_FLAG_DIRECT AUDIO_OUTPUT_FLAG_MMAP_NOIRQ">
+                    <profile name="" format="AUDIO_FORMAT_PCM_FLOAT"
+                             samplingRates="48000 96000 384000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="immersive_out" role="source" flags="AUDIO_OUTPUT_FLAG_SPATIALIZER">
+                    <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
+                             samplingRates="48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="incall playback" role="source"
+                         flags="AUDIO_OUTPUT_FLAG_INCALL_MUSIC">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000 96000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO" />
+                </mixPort>
+                <mixPort name="voice call tx" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000 96000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO" />
+                </mixPort>
+                <mixPort name="voip_rx" role="source"
+                         flags="AUDIO_OUTPUT_FLAG_VOIP_RX">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                           samplingRates="48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="primary input" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000 96000"
+                             channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO AUDIO_CHANNEL_INDEX_MASK_3"/>
+                </mixPort>
+                <mixPort name="hotword input" role="sink" flags="AUDIO_INPUT_FLAG_HW_HOTWORD" maxActiveCount="0" >
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000 11025 12000 16000 22050 24000 32000 44100 48000 96000"
+                             channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO"/>
+                </mixPort>
+                <mixPort name="incall capture" role="sink"  maxActiveCount="2" maxOpenCount="2">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000 96000"
+                             channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+                </mixPort>
+                <mixPort name="voice call rx" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000 96000"
+                             channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+                </mixPort>
+                <mixPort name="voip_tx" role="sink"
+                         flags="AUDIO_INPUT_FLAG_VOIP_TX">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000 96000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+                </mixPort>
+                <mixPort name="fast input" role="sink" flags="AUDIO_INPUT_FLAG_RAW AUDIO_INPUT_FLAG_FAST">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000 96000"
+                             channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO"/>
+                </mixPort>
+                <mixPort name="mmap_no_irq_in" role="sink" flags="AUDIO_INPUT_FLAG_MMAP_NOIRQ">
+                    <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
+                             samplingRates="48000 96000"
+                             channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO"/>
+                </mixPort>
+                <mixPort name="hifi_playback" role="source" />
+                <mixPort name="hifi_input" role="sink" />
+                <mixPort name="echo_ref_input" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
+                             samplingRates="48000 96000"
+                             channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+                </mixPort>
+            </mixPorts>
+            <devicePorts>
+                <!-- Output devices declaration, i.e. Sink DEVICE PORT -->
+                <devicePort tagName="Earpiece" type="AUDIO_DEVICE_OUT_EARPIECE" role="sink">
+                </devicePort>
+                <devicePort tagName="Speaker" type="AUDIO_DEVICE_OUT_SPEAKER" role="sink">
+                </devicePort>
+                <devicePort tagName="Speaker Safe" type="AUDIO_DEVICE_OUT_SPEAKER_SAFE" role="sink">
+                </devicePort>
+                <devicePort tagName="BT SCO" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO" role="sink">
+                </devicePort>
+                <devicePort tagName="BT SCO Headset" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET" role="sink">
+                </devicePort>
+                <devicePort tagName="BT SCO Car Kit" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT" role="sink">
+                </devicePort>
+                <devicePort tagName="USB Device Out" type="AUDIO_DEVICE_OUT_USB_DEVICE" role="sink">
+                </devicePort>
+                <devicePort tagName="USB Headset Out" type="AUDIO_DEVICE_OUT_USB_HEADSET" role="sink">
+                </devicePort>
+                <devicePort tagName="HDMI Out" type="AUDIO_DEVICE_OUT_HDMI" role="sink">
+                </devicePort>
+                <devicePort tagName="Telephony Tx" type="AUDIO_DEVICE_OUT_TELEPHONY_TX" role="sink">
+                </devicePort>
+                <!-- Input devices declaration, i.e. Source DEVICE PORT -->
+                <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
+                </devicePort>
+                <devicePort tagName="Built-In Back Mic" type="AUDIO_DEVICE_IN_BACK_MIC" role="source">
+                </devicePort>
+                <devicePort tagName="BT SCO Headset Mic" type="AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET" role="source">
+                </devicePort>
+                <devicePort tagName="BT A2DP Out" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP" role="sink"
+                            encodedFormats="AUDIO_FORMAT_OPUS AUDIO_FORMAT_AAC AUDIO_FORMAT_SBC">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="44100 48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
+                <devicePort tagName="BT A2DP Headphones" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES" role="sink"
+                            encodedFormats="AUDIO_FORMAT_OPUS AUDIO_FORMAT_AAC AUDIO_FORMAT_SBC">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="44100 48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
+                <devicePort tagName="BT A2DP Speaker" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER" role="sink"
+                            encodedFormats="AUDIO_FORMAT_OPUS AUDIO_FORMAT_AAC AUDIO_FORMAT_SBC">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="44100 48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
+                <devicePort tagName="BT BLE Headset" type="AUDIO_DEVICE_OUT_BLE_HEADSET" role="sink"
+                            encodedFormats="AUDIO_FORMAT_LC3">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
+                <devicePort tagName="BT BLE Speaker" type="AUDIO_DEVICE_OUT_BLE_SPEAKER" role="sink"
+                            encodedFormats="AUDIO_FORMAT_LC3">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
+                <devicePort tagName="BT BLE Broadcast" type="AUDIO_DEVICE_OUT_BLE_BROADCAST" role="sink"
+                            encodedFormats="AUDIO_FORMAT_LC3">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000 96000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
+                <devicePort tagName="BLE Headset Mic" type="AUDIO_DEVICE_IN_BLE_HEADSET" role="source">
+                </devicePort>
+                <devicePort tagName="USB Device In" type="AUDIO_DEVICE_IN_USB_DEVICE" role="source">
+                </devicePort>
+                <devicePort tagName="USB Headset In" type="AUDIO_DEVICE_IN_USB_HEADSET" role="source">
+                </devicePort>
+                <!-- AUDIO_DEVICE_IN_VOICE_CALL and AUDIO_DEVICE_IN_TELEPHONY_RX are in the same value -->
+                <devicePort tagName="Voice Call And Telephony Rx" type="AUDIO_DEVICE_IN_VOICE_CALL" role="source">
+                </devicePort>
+                <devicePort tagName="Echo Ref In" type="AUDIO_DEVICE_IN_ECHO_REFERENCE" role="source">
+                </devicePort>
+            </devicePorts>
+            <!-- route declaration, i.e. list all available sources for a given sink -->
+            <routes>
+                <route type="mix" sink="Speaker"
+                       sources="primary output,direct,deep buffer,haptic,raw,mmap_no_irq_out,voip_rx,compressed_offload"/>
+                <route type="mix" sink="Speaker Safe"
+                       sources="primary output,direct,deep buffer,haptic,raw,mmap_no_irq_out,voip_rx,compressed_offload"/>
+                <route type="mix" sink="Earpiece"
+                       sources="primary output,direct,deep buffer,haptic,raw,mmap_no_irq_out,voip_rx,compressed_offload"/>
+                <route type="mix" sink="BT A2DP Out"
+                       sources="primary output,direct,deep buffer,haptic,voip_rx,compressed_offload,raw,mmap_no_irq_out,immersive_out"/>
+                <route type="mix" sink="BT A2DP Headphones"
+                       sources="primary output,direct,deep buffer,haptic,voip_rx,compressed_offload,raw,mmap_no_irq_out,immersive_out"/>
+                <route type="mix" sink="BT A2DP Speaker"
+                       sources="primary output,direct,deep buffer,haptic,voip_rx,compressed_offload,raw,mmap_no_irq_out,immersive_out"/>
+                <route type="mix" sink="BT BLE Headset"
+                       sources="primary output,direct,deep buffer,haptic,voip_rx,compressed_offload,raw,mmap_no_irq_out,immersive_out"/>
+                <route type="mix" sink="BT BLE Speaker"
+                       sources="primary output,direct,deep buffer,haptic,voip_rx,compressed_offload,raw,mmap_no_irq_out,immersive_out"/>
+                <route type="mix" sink="BT BLE Broadcast"
+                       sources="primary output,direct,deep buffer,haptic,voip_rx,compressed_offload,raw,mmap_no_irq_out,immersive_out"/>
+                <route type="mix" sink="USB Device Out"
+                       sources="primary output,direct,deep buffer,haptic,raw,mmap_no_irq_out,voip_rx,hifi_playback,compressed_offload,immersive_out"/>
+                <route type="mix" sink="USB Headset Out"
+                       sources="primary output,direct,deep buffer,haptic,raw,mmap_no_irq_out,voip_rx,hifi_playback,compressed_offload,immersive_out"/>
+                <route type="mix" sink="HDMI Out"
+                       sources="primary output,direct,deep buffer,haptic,raw,mmap_no_irq_out,voip_rx,compressed_offload"/>
+                <route type="mix" sink="BT SCO"
+                       sources="primary output,direct,deep buffer,haptic,voip_rx,compressed_offload,raw,mmap_no_irq_out"/>
+                <route type="mix" sink="BT SCO Headset"
+                       sources="primary output,direct,deep buffer,haptic,voip_rx,compressed_offload,raw,mmap_no_irq_out"/>
+                <route type="mix" sink="BT SCO Car Kit"
+                       sources="primary output,direct,deep buffer,haptic,voip_rx,compressed_offload,raw,mmap_no_irq_out"/>
+                <route type="mix" sink="Telephony Tx" sources="incall playback,voice call tx" />
+                <route type="mix" sink="primary input"
+                       sources="Built-In Mic,Built-In Back Mic,USB Device In,USB Headset In,BT SCO Headset Mic,BLE Headset Mic"/>
+                <route type="mix" sink="hotword input"
+                       sources="Built-In Mic,Built-In Back Mic,USB Device In,USB Headset In,BT SCO Headset Mic,BLE Headset Mic"/>
+                <route type="mix" sink="incall capture" sources="Voice Call And Telephony Rx" />
+                <route type="mix" sink="voice call rx" sources="Voice Call And Telephony Rx" />
+                <route type="mix" sink="voip_tx"
+                       sources="Built-In Mic,Built-In Back Mic,USB Device In,USB Headset In,BT SCO Headset Mic,BLE Headset Mic"/>
+                <route type="mix" sink="fast input"
+                       sources="Built-In Mic,Built-In Back Mic,USB Device In,USB Headset In,BT SCO Headset Mic,BLE Headset Mic"/>
+                <route type="mix" sink="mmap_no_irq_in"
+                       sources="Built-In Mic,Built-In Back Mic,USB Device In,USB Headset In,BT SCO Headset Mic,BLE Headset Mic"/>
+                <route type="mix" sink="hifi_input" sources="USB Device In,USB Headset In" />
+                <route type="mix" sink="echo_ref_input" sources="Echo Ref In"/>
+            </routes>
+        </module>
+        <!-- Usb Audio HAL -->
+        <module name="usbv2" halVersion="2.0">
+            <mixPorts>
+                <mixPort name="usb_accessory output" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="44100" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+            </mixPorts>
+            <devicePorts>
+                <devicePort tagName="USB Host Out" type="AUDIO_DEVICE_OUT_USB_ACCESSORY" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="44100" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
+            </devicePorts>
+            <routes>
+                <route type="mix" sink="USB Host Out"
+                       sources="usb_accessory output"/>
+            </routes>
+        </module>
+    </modules>
+    <!-- End of Modules section -->
+</audioPolicyConfiguration>
diff --git a/services/camera/libcameraservice/Android.bp b/services/camera/libcameraservice/Android.bp
index 38476a4..0c4bfcb 100644
--- a/services/camera/libcameraservice/Android.bp
+++ b/services/camera/libcameraservice/Android.bp
@@ -265,9 +265,21 @@
         "liblog",
         "libutils",
         "libxml2",
-        "camera_platform_flags_c_lib",
     ],
 
+    target: {
+        android: {
+            shared_libs: [
+                "camera_platform_flags_c_lib",
+            ],
+        },
+        host: {
+            shared_libs: [
+                "camera_platform_flags_c_lib_for_test",
+            ],
+        },
+    },
+
     include_dirs: [
         "frameworks/av/camera/include",
         "frameworks/av/camera/include/camera",
diff --git a/services/camera/libcameraservice/tests/Android.bp b/services/camera/libcameraservice/tests/Android.bp
index 55e2c9d..d49aad6 100644
--- a/services/camera/libcameraservice/tests/Android.bp
+++ b/services/camera/libcameraservice/tests/Android.bp
@@ -44,13 +44,25 @@
         "libjpeg",
         "liblog",
         "libutils",
-        "camera_platform_flags_c_lib",
     ],
 
     static_libs: [
         "libgmock",
     ],
 
+    target: {
+        android: {
+            shared_libs: [
+                "camera_platform_flags_c_lib",
+            ],
+        },
+        host: {
+            shared_libs: [
+                "camera_platform_flags_c_lib_for_test",
+            ],
+        },
+    },
+
     cflags: [
         "-Wall",
         "-Wextra",
diff --git a/services/mediametrics/fuzzer/mediametrics_aidl_fuzzer.cpp b/services/mediametrics/fuzzer/mediametrics_aidl_fuzzer.cpp
index c7468c7..572e969 100644
--- a/services/mediametrics/fuzzer/mediametrics_aidl_fuzzer.cpp
+++ b/services/mediametrics/fuzzer/mediametrics_aidl_fuzzer.cpp
@@ -22,6 +22,7 @@
 using ::android::MediaMetricsService;
 
 extern "C" int LLVMFuzzerTestOneInput(const uint8_t* data, size_t size) {
+    signal(SIGPIPE, SIG_IGN);
     auto service = sp<MediaMetricsService>::make();
     fuzzService(service, FuzzedDataProvider(data, size));
     return 0;
diff --git a/services/mediaresourcemanager/fuzzer/resourcemanager_service_fuzzer.cpp b/services/mediaresourcemanager/fuzzer/resourcemanager_service_fuzzer.cpp
index 6253df7..1cad482 100644
--- a/services/mediaresourcemanager/fuzzer/resourcemanager_service_fuzzer.cpp
+++ b/services/mediaresourcemanager/fuzzer/resourcemanager_service_fuzzer.cpp
@@ -26,6 +26,7 @@
 using ndk::SharedRefBase;
 
 extern "C" int LLVMFuzzerTestOneInput(const uint8_t* data, size_t size) {
+   signal(SIGPIPE, SIG_IGN);
    std::shared_ptr<ResourceManagerService> service = ResourceManagerService::Create();
    fuzzService(service->asBinder().get(), FuzzedDataProvider(data, size));
    return 0;