Merge "CCodecBufferChannel: remove rendering depth" into main
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/camera/cameraserver/Android.bp b/camera/cameraserver/Android.bp
index 6862cb1..d0df90b 100644
--- a/camera/cameraserver/Android.bp
+++ b/camera/cameraserver/Android.bp
@@ -22,6 +22,11 @@
     default_applicable_licenses: ["frameworks_av_camera_license"],
 }
 
+vintf_fragment {
+    name: "manifest_android.frameworks.cameraservice.service.xml",
+    src: "manifest_android.frameworks.cameraservice.service.xml",
+}
+
 cc_binary {
     name: "cameraserver",
 
@@ -61,7 +66,7 @@
 
     init_rc: ["cameraserver.rc"],
 
-    vintf_fragments: [
+    vintf_fragment_modules: [
         "manifest_android.frameworks.cameraservice.service.xml",
     ],
 }
diff --git a/drm/libmediadrmrkp/include/DrmRemotelyProvisionedComponent.h b/drm/libmediadrmrkp/include/DrmRemotelyProvisionedComponent.h
index f046785..97a8cc4 100644
--- a/drm/libmediadrmrkp/include/DrmRemotelyProvisionedComponent.h
+++ b/drm/libmediadrmrkp/include/DrmRemotelyProvisionedComponent.h
@@ -35,7 +35,8 @@
 class DrmRemotelyProvisionedComponent : public BnRemotelyProvisionedComponent {
   public:
     DrmRemotelyProvisionedComponent(std::shared_ptr<IDrmPlugin> drm, std::string drmVendor,
-                                    std::string drmDesc, std::vector<uint8_t> bcc);
+                                    std::string drmDesc, std::vector<uint8_t> bcc,
+                                    std::vector<uint8_t> bcc_signature);
     ScopedAStatus getHardwareInfo(RpcHardwareInfo* info) override;
 
     ScopedAStatus generateEcdsaP256KeyPair(bool testMode, MacedPublicKey* macedPublicKey,
@@ -60,6 +61,7 @@
     std::string mDrmVendor;
     std::string mDrmDesc;
     std::vector<uint8_t> mBcc;
+    std::vector<uint8_t> mBccSignature;
 };
 }  // namespace android::mediadrm
 
diff --git a/drm/libmediadrmrkp/src/DrmRemotelyProvisionedComponent.cpp b/drm/libmediadrmrkp/src/DrmRemotelyProvisionedComponent.cpp
index 440be79..65054b0 100644
--- a/drm/libmediadrmrkp/src/DrmRemotelyProvisionedComponent.cpp
+++ b/drm/libmediadrmrkp/src/DrmRemotelyProvisionedComponent.cpp
@@ -28,11 +28,13 @@
 DrmRemotelyProvisionedComponent::DrmRemotelyProvisionedComponent(std::shared_ptr<IDrmPlugin> drm,
                                                                  std::string drmVendor,
                                                                  std::string drmDesc,
-                                                                 std::vector<uint8_t> bcc)
+                                                                 std::vector<uint8_t> bcc,
+                                                                 std::vector<uint8_t> bcc_signature)
     : mDrm(std::move(drm)),
       mDrmVendor(std::move(drmVendor)),
       mDrmDesc(std::move(drmDesc)),
-      mBcc(std::move(bcc)) {}
+      mBcc(std::move(bcc)),
+      mBccSignature(std::move(bcc_signature)) {}
 
 ScopedAStatus DrmRemotelyProvisionedComponent::getHardwareInfo(RpcHardwareInfo* info) {
     info->versionNumber = 3;
@@ -107,7 +109,7 @@
     for (auto i : keyToProp) {
         auto key = i.first;
         auto prop = i.second;
-        const auto& val= deviceInfoMap.get(key);
+        const auto& val = deviceInfoMap.get(key);
         if (val == nullptr || val->asTstr()->value().empty()) {
             std::string propValue = android::base::GetProperty(prop, "");
             if (propValue.empty()) {
@@ -161,12 +163,16 @@
     }
 
     // assemble AuthenticatedRequest (definition in IRemotelyProvisionedComponent.aidl)
-    *out = cppbor::Array()
-                   .add(1 /* version */)
-                   .add(cppbor::Map() /* UdsCerts */)
-                   .add(cppbor::EncodedItem(mBcc))
-                   .add(cppbor::EncodedItem(std::move(deviceSignedCsrPayload)))
-                   .encode();
+    cppbor::Array request_array = cppbor::Array().add(1 /* version */);
+    if (!mBccSignature.empty()) {
+        request_array.add(cppbor::EncodedItem(mBccSignature) /* UdsCerts */);
+    } else {
+        request_array.add(cppbor::Map() /* empty UdsCerts */);
+    }
+    request_array.add(cppbor::EncodedItem(mBcc))
+            .add(cppbor::EncodedItem(std::move(deviceSignedCsrPayload)));
+    *out = request_array.encode();
+
     return ScopedAStatus::ok();
 }
 }  // namespace android::mediadrm
\ No newline at end of file
diff --git a/drm/libmediadrmrkp/src/DrmRkpAdapter.cpp b/drm/libmediadrmrkp/src/DrmRkpAdapter.cpp
index 515d157..750b51e 100644
--- a/drm/libmediadrmrkp/src/DrmRkpAdapter.cpp
+++ b/drm/libmediadrmrkp/src/DrmRkpAdapter.cpp
@@ -87,13 +87,21 @@
                           status.getDescription().c_str());
                     return;
                 }
-
+                std::vector<uint8_t> bcc_signature;
+                status =
+                        mDrm->getPropertyByteArray("bootCertificateChainSignature", &bcc_signature);
+                if (!status.isOk()) {
+                    ALOGW("mDrm->getPropertyByteArray(\"bootCertificateChainSignature\") failed."
+                          "Detail: [%s].",
+                          status.getDescription().c_str());
+                    // bcc signature is optional, no need to return when it is unavailable.
+                }
                 std::string compName(instance);
                 auto comps = static_cast<
                         std::map<std::string, std::shared_ptr<IRemotelyProvisionedComponent>>*>(
                         context);
                 (*comps)[compName] = ::ndk::SharedRefBase::make<DrmRemotelyProvisionedComponent>(
-                        mDrm, drmVendor, drmDesc, bcc);
+                        mDrm, drmVendor, drmDesc, bcc, bcc_signature);
             });
     return comps;
 }
diff --git a/media/TEST_MAPPING b/media/TEST_MAPPING
index 1a637ac..695cad6 100644
--- a/media/TEST_MAPPING
+++ b/media/TEST_MAPPING
@@ -45,6 +45,32 @@
             "file_patterns": ["(?i)drm|crypto"]
         }
     ],
+    "postsubmit": [
+        {
+            "name": "MctsMediaCodecTestCases",
+            "options": [
+                {
+                    "include-filter": "android.media.codec.cts.EncodeDecodeTest"
+                }
+            ]
+        },
+        {
+            "name": "MctsMediaCodecTestCases",
+            "options": [
+                {
+                    "include-filter": "android.media.codec.cts.DecodeEditEncodeTest"
+                }
+            ]
+        },
+        {
+            "name": "MctsMediaCodecTestCases",
+            "options": [
+                {
+                    "include-filter": "android.media.codec.cts.ExtractDecodeEditEncodeMuxTest"
+                }
+            ]
+        }
+    ],
     // Postsubmit tests for TV devices
     "tv-postsubmit": [
         {
diff --git a/media/aconfig/codec_fwk.aconfig b/media/aconfig/codec_fwk.aconfig
index c0ebcd5..14540b7 100644
--- a/media/aconfig/codec_fwk.aconfig
+++ b/media/aconfig/codec_fwk.aconfig
@@ -13,6 +13,16 @@
 }
 
 flag {
+  name: "codec_buffer_state_cleanup"
+  namespace: "codec_fwk"
+  description: "Bugfix flag for more buffer state cleanup in MediaCodec"
+  bug: "343502509"
+  metadata {
+    purpose: PURPOSE_BUGFIX
+  }
+}
+
+flag {
   name: "dataspace_v0_partial"
   namespace: "codec_fwk"
   description: "Bugfix flag for using V0 dataspace in some cases"
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/audio/aconfig/audio_framework.aconfig b/media/audio/aconfig/audio_framework.aconfig
index 0209e28..ea5f26d 100644
--- a/media/audio/aconfig/audio_framework.aconfig
+++ b/media/audio/aconfig/audio_framework.aconfig
@@ -40,6 +40,14 @@
     bug: "316414750"
 }
 
+flag {
+    name: "dolby_ac4_level4_encoding_api"
+    namespace: "media_audio"
+    description: "Feature flag for Dolby AC-4 level 4 AudioFormat encoding."
+    is_fixed_read_only: true
+    bug: "266537650"
+}
+
 # TODO remove
 flag {
     name: "foreground_audio_control"
diff --git a/media/audioaidlconversion/AidlConversionCppNdk.cpp b/media/audioaidlconversion/AidlConversionCppNdk.cpp
index 01b6e42..90996a3 100644
--- a/media/audioaidlconversion/AidlConversionCppNdk.cpp
+++ b/media/audioaidlconversion/AidlConversionCppNdk.cpp
@@ -741,6 +741,8 @@
             {// Note: not in the IANA registry.
              AUDIO_FORMAT_APTX_HD, make_AudioFormatDescription("audio/vnd.qcom.aptx.hd")},
             {AUDIO_FORMAT_AC4, make_AudioFormatDescription(::android::MEDIA_MIMETYPE_AUDIO_AC4)},
+            {AUDIO_FORMAT_AC4_L4, make_AudioFormatDescription(
+                    std::string(::android::MEDIA_MIMETYPE_AUDIO_AC4) + ";version=02.01.04")},
             {// Note: not in the IANA registry.
              AUDIO_FORMAT_LDAC, make_AudioFormatDescription("audio/vnd.sony.ldac")},
             {AUDIO_FORMAT_MAT,
diff --git a/media/codec2/components/aom/C2SoftAomEnc.cpp b/media/codec2/components/aom/C2SoftAomEnc.cpp
index 722b13a..93009c4 100644
--- a/media/codec2/components/aom/C2SoftAomEnc.cpp
+++ b/media/codec2/components/aom/C2SoftAomEnc.cpp
@@ -466,6 +466,7 @@
 
 aom_codec_err_t C2SoftAomEnc::setupCodecParameters() {
     aom_codec_err_t codec_return = AOM_CODEC_OK;
+    const int maxIntraBitratePct = mBitrateControlMode == AOM_CBR ? 300 : 450;
 
     codec_return = aom_codec_control(mCodecContext, AV1E_SET_TARGET_SEQ_LEVEL_IDX, mAV1EncLevel);
     if (codec_return != AOM_CODEC_OK) goto BailOut;
@@ -492,6 +493,10 @@
     codec_return = aom_codec_control(mCodecContext, AV1E_SET_AQ_MODE, 3);
     if (codec_return != AOM_CODEC_OK) goto BailOut;
 
+    codec_return = aom_codec_control(mCodecContext, AOME_SET_MAX_INTRA_BITRATE_PCT,
+                                     maxIntraBitratePct);
+    if (codec_return != AOM_CODEC_OK) goto BailOut;
+
     codec_return = aom_codec_control(mCodecContext, AV1E_SET_COEFF_COST_UPD_FREQ, 3);
     if (codec_return != AOM_CODEC_OK) goto BailOut;
 
diff --git a/media/codec2/components/mp3/C2SoftMp3Dec.cpp b/media/codec2/components/mp3/C2SoftMp3Dec.cpp
index 149c6ee..aed5e68 100644
--- a/media/codec2/components/mp3/C2SoftMp3Dec.cpp
+++ b/media/codec2/components/mp3/C2SoftMp3Dec.cpp
@@ -114,7 +114,9 @@
 c2_status_t C2SoftMP3::onStop() {
     // Make sure that the next buffer output does not still
     // depend on fragments from the last one decoded.
-    pvmp3_InitDecoder(mConfig, mDecoderBuf);
+    if (mDecoderBuf) {
+        pvmp3_InitDecoder(mConfig, mDecoderBuf);
+    }
     mSignalledError = false;
     mIsFirst = true;
     mSignalledOutputEos = false;
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..bb4c596 100644
--- a/media/codec2/hal/aidl/include/codec2/aidl/ComponentStore.h
+++ b/media/codec2/hal/aidl/include/codec2/aidl/ComponentStore.h
@@ -52,6 +52,13 @@
 using ::aidl::android::hardware::media::bufferpool2::IClientManager;
 
 struct ComponentStore : public BnComponentStore {
+    /**
+     * Constructor for ComponentStore.
+     *
+     * IMPORTANT: SetPreferredCodec2ComponentStore() is called in the constructor.
+     * Be careful about the order of SetPreferredCodec2ComponentStore() and
+     * ComponentStore() in the code.
+     */
     ComponentStore(const std::shared_ptr<C2ComponentStore>& store);
     virtual ~ComponentStore();
 
@@ -120,6 +127,9 @@
     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/common/include/codec2/common/BqPoolInvalidateHelper.h b/media/codec2/hal/common/include/codec2/common/BqPoolInvalidateHelper.h
new file mode 100644
index 0000000..859f703
--- /dev/null
+++ b/media/codec2/hal/common/include/codec2/common/BqPoolInvalidateHelper.h
@@ -0,0 +1,50 @@
+/*
+ * Copyright 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.
+ */
+
+#pragma once
+
+#include <C2BqBufferPriv.h>
+#include <C2PlatformSupport.h>
+
+namespace android {
+
+// filter fn from component's blockpool container to bqpool conatainer
+static inline bool BqPoolFilterFn(
+        std::pair<const uint64_t, std::shared_ptr<C2BlockPool>> pool) {
+    return (pool.second->getAllocatorId() == C2PlatformAllocatorStore::BUFFERQUEUE);
+}
+
+// convert fn from component's blockpool container to bqpool container
+static inline std::shared_ptr<C2BufferQueueBlockPool> BqPoolConvertFn(
+        std::pair<const uint64_t, std::shared_ptr<C2BlockPool>> pool) {
+    return std::static_pointer_cast<C2BufferQueueBlockPool>(pool.second);
+}
+
+// This is similar to std::transform excpet there is \pred functor parameter.
+// The elements with \pred function value \true only will be transformed and
+// added to the dest container. (For portability std::ranges are not used.)
+template <class InputIt, class OutputIt, class Pred, class Fct>
+void transform_if(InputIt first, InputIt last, OutputIt dest, Pred pred, Fct transform)
+{
+   while (first != last) {
+      if (pred(*first)) {
+         *dest++ = transform(*first);
+      }
+      ++first;
+   }
+}
+
+}  // namespace android
diff --git a/media/codec2/hal/hidl/1.0/utils/Component.cpp b/media/codec2/hal/hidl/1.0/utils/Component.cpp
index 62f0e25..162a80e 100644
--- a/media/codec2/hal/hidl/1.0/utils/Component.cpp
+++ b/media/codec2/hal/hidl/1.0/utils/Component.cpp
@@ -18,6 +18,7 @@
 #define LOG_TAG "Codec2-Component"
 #include <android-base/logging.h>
 
+#include <codec2/common/BqPoolInvalidateHelper.h>
 #include <codec2/hidl/1.0/Component.h>
 #include <codec2/hidl/1.0/ComponentStore.h>
 #include <codec2/hidl/1.0/InputBufferManager.h>
@@ -30,6 +31,7 @@
 #include <utils/Timers.h>
 
 #include <C2BqBufferPriv.h>
+#include <C2BqPoolInvalidator.h>
 #include <C2Debug.h>
 #include <C2PlatformSupport.h>
 
@@ -270,16 +272,17 @@
 }
 
 void Component::onDeathReceived() {
+    std::list<std::shared_ptr<C2BufferQueueBlockPool>> bqPools;
     {
         std::lock_guard<std::mutex> lock(mBlockPoolsMutex);
         mClientDied = true;
-        for (auto it = mBlockPools.begin(); it != mBlockPools.end(); ++it) {
-            if (it->second->getAllocatorId() == C2PlatformAllocatorStore::BUFFERQUEUE) {
-                std::shared_ptr<C2BufferQueueBlockPool> bqPool =
-                        std::static_pointer_cast<C2BufferQueueBlockPool>(it->second);
-                bqPool->invalidate();
-            }
-        }
+        transform_if(mBlockPools.begin(), mBlockPools.end(), std::back_inserter(bqPools),
+                BqPoolFilterFn, BqPoolConvertFn);
+    }
+    if (!bqPools.empty()) {
+        std::shared_ptr<C2BqPoolInvalidateItem> bqInvalidateItem =
+                std::make_shared<C2BqPoolInvalidateItem>(std::move(bqPools));
+        bqInvalidateItem->invalidate();
     }
     release();
 }
@@ -549,7 +552,26 @@
 }
 
 Return<Status> Component::release() {
+    std::list<std::shared_ptr<C2BufferQueueBlockPool>> bqPools;
+    {
+        std::lock_guard<std::mutex> lock(mBlockPoolsMutex);
+        if (!mClientDied) {
+            transform_if(mBlockPools.begin(), mBlockPools.end(), std::back_inserter(bqPools),
+                     BqPoolFilterFn, BqPoolConvertFn);
+        }
+    }
+    std::shared_ptr<C2BqPoolInvalidateItem> bqInvalidateItem;
+    if (!bqPools.empty()) {
+        // handling rare cases of process death just after release() called.
+        bqInvalidateItem = std::make_shared<C2BqPoolInvalidateItem>(std::move(bqPools));
+        C2BqPoolInvalidator::getInstance().queue(bqInvalidateItem);
+    }
     Status status = static_cast<Status>(mComponent->release());
+    if (bqInvalidateItem) {
+        // If release is not blocked,
+        // skip invalidation and finish ASAP.
+        bqInvalidateItem->skip();
+    }
     {
         std::lock_guard<std::mutex> lock(mBlockPoolsMutex);
         mBlockPools.clear();
@@ -637,6 +659,18 @@
 }
 
 Component::~Component() {
+    std::list<std::shared_ptr<C2BufferQueueBlockPool>> bqPools;
+    {
+        std::lock_guard<std::mutex> lock(mBlockPoolsMutex);
+        transform_if(mBlockPools.begin(), mBlockPools.end(), std::back_inserter(bqPools),
+                     BqPoolFilterFn, BqPoolConvertFn);
+    }
+    if (!bqPools.empty()) {
+        LOG(ERROR) << "blockpools are not cleared yet at dtor";
+        std::shared_ptr<C2BqPoolInvalidateItem> bqInvalidateItem =
+                std::make_shared<C2BqPoolInvalidateItem>(std::move(bqPools));
+        C2BqPoolInvalidator::getInstance().queue(bqInvalidateItem);
+    }
     InputBufferManager::unregisterFrameData(mListener);
     mStore->reportComponentDeath(this);
 }
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..028238b 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
@@ -55,6 +55,13 @@
 using ::android::sp;
 
 struct ComponentStore : public IComponentStore {
+    /**
+     * Constructor for ComponentStore.
+     *
+     * IMPORTANT: SetPreferredCodec2ComponentStore() is called in the constructor.
+     * Be careful about the order of SetPreferredCodec2ComponentStore() and
+     * ComponentStore() in the code.
+     */
     ComponentStore(const std::shared_ptr<C2ComponentStore>& store);
     virtual ~ComponentStore();
 
@@ -124,6 +131,9 @@
     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/Component.cpp b/media/codec2/hal/hidl/1.1/utils/Component.cpp
index 7f2c4dd..1c2a49a 100644
--- a/media/codec2/hal/hidl/1.1/utils/Component.cpp
+++ b/media/codec2/hal/hidl/1.1/utils/Component.cpp
@@ -18,6 +18,7 @@
 #define LOG_TAG "Codec2-Component@1.1"
 #include <android-base/logging.h>
 
+#include <codec2/common/BqPoolInvalidateHelper.h>
 #include <codec2/hidl/1.1/Component.h>
 #include <codec2/hidl/1.1/ComponentStore.h>
 #include <codec2/hidl/1.1/InputBufferManager.h>
@@ -32,6 +33,7 @@
 #include <codec2/common/MultiAccessUnitHelper.h>
 
 #include <C2BqBufferPriv.h>
+#include <C2BqPoolInvalidator.h>
 #include <C2Debug.h>
 #include <C2PlatformSupport.h>
 
@@ -274,16 +276,17 @@
 }
 
 void Component::onDeathReceived() {
+    std::list<std::shared_ptr<C2BufferQueueBlockPool>> bqPools;
     {
         std::lock_guard<std::mutex> lock(mBlockPoolsMutex);
         mClientDied = true;
-        for (auto it = mBlockPools.begin(); it != mBlockPools.end(); ++it) {
-            if (it->second->getAllocatorId() == C2PlatformAllocatorStore::BUFFERQUEUE) {
-                std::shared_ptr<C2BufferQueueBlockPool> bqPool =
-                        std::static_pointer_cast<C2BufferQueueBlockPool>(it->second);
-                bqPool->invalidate();
-            }
-        }
+        transform_if(mBlockPools.begin(), mBlockPools.end(), std::back_inserter(bqPools),
+                BqPoolFilterFn, BqPoolConvertFn);
+    }
+    if (!bqPools.empty()) {
+        std::shared_ptr<C2BqPoolInvalidateItem> bqInvalidateItem =
+                std::make_shared<C2BqPoolInvalidateItem>(std::move(bqPools));
+        bqInvalidateItem->invalidate();
     }
     release();
 }
@@ -555,7 +558,26 @@
 }
 
 Return<Status> Component::release() {
+    std::list<std::shared_ptr<C2BufferQueueBlockPool>> bqPools;
+    {
+        std::lock_guard<std::mutex> lock(mBlockPoolsMutex);
+        if (!mClientDied) {
+            transform_if(mBlockPools.begin(), mBlockPools.end(), std::back_inserter(bqPools),
+                    BqPoolFilterFn, BqPoolConvertFn);
+        }
+    }
+    std::shared_ptr<C2BqPoolInvalidateItem> bqInvalidateItem;
+    if (!bqPools.empty()) {
+        // handling rare cases of process death just after release() called.
+        bqInvalidateItem = std::make_shared<C2BqPoolInvalidateItem>(std::move(bqPools));
+        C2BqPoolInvalidator::getInstance().queue(bqInvalidateItem);
+    }
     Status status = static_cast<Status>(mComponent->release());
+    if (bqInvalidateItem) {
+        // If release is not blocked,
+        // skip invalidation and finish ASAP.
+        bqInvalidateItem->skip();
+    }
     {
         std::lock_guard<std::mutex> lock(mBlockPoolsMutex);
         mBlockPools.clear();
@@ -649,6 +671,18 @@
 }
 
 Component::~Component() {
+    std::list<std::shared_ptr<C2BufferQueueBlockPool>> bqPools;
+    {
+        std::lock_guard<std::mutex> lock(mBlockPoolsMutex);
+        transform_if(mBlockPools.begin(), mBlockPools.end(), std::back_inserter(bqPools),
+                BqPoolFilterFn, BqPoolConvertFn);
+    }
+    if (!bqPools.empty()) {
+        LOG(ERROR) << "blockpools are not cleared yet at dtor";
+        std::shared_ptr<C2BqPoolInvalidateItem> bqInvalidateItem =
+                std::make_shared<C2BqPoolInvalidateItem>(std::move(bqPools));
+        C2BqPoolInvalidator::getInstance().queue(bqInvalidateItem);
+    }
     InputBufferManager::unregisterFrameData(mListener);
     mStore->reportComponentDeath(this);
 }
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..b023115 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
@@ -56,6 +56,13 @@
 using ::android::sp;
 
 struct ComponentStore : public IComponentStore {
+    /**
+     * Constructor for ComponentStore.
+     *
+     * IMPORTANT: SetPreferredCodec2ComponentStore() is called in the constructor.
+     * Be careful about the order of SetPreferredCodec2ComponentStore() and
+     * ComponentStore() in the code.
+     */
     ComponentStore(const std::shared_ptr<C2ComponentStore>& store);
     virtual ~ComponentStore();
 
@@ -132,6 +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/hidl/1.2/utils/Component.cpp b/media/codec2/hal/hidl/1.2/utils/Component.cpp
index 7b0aa9b..a15febe 100644
--- a/media/codec2/hal/hidl/1.2/utils/Component.cpp
+++ b/media/codec2/hal/hidl/1.2/utils/Component.cpp
@@ -18,6 +18,7 @@
 #define LOG_TAG "Codec2-Component@1.2"
 #include <android-base/logging.h>
 
+#include <codec2/common/BqPoolInvalidateHelper.h>
 #include <codec2/hidl/1.2/Component.h>
 #include <codec2/hidl/1.2/ComponentStore.h>
 #include <codec2/hidl/1.2/InputBufferManager.h>
@@ -30,6 +31,7 @@
 #include <utils/Timers.h>
 
 #include <C2BqBufferPriv.h>
+#include <C2BqPoolInvalidator.h>
 #include <C2Debug.h>
 #include <C2PlatformSupport.h>
 
@@ -272,16 +274,17 @@
 }
 
 void Component::onDeathReceived() {
+    std::list<std::shared_ptr<C2BufferQueueBlockPool>> bqPools;
     {
         std::lock_guard<std::mutex> lock(mBlockPoolsMutex);
         mClientDied = true;
-        for (auto it = mBlockPools.begin(); it != mBlockPools.end(); ++it) {
-            if (it->second->getAllocatorId() == C2PlatformAllocatorStore::BUFFERQUEUE) {
-                std::shared_ptr<C2BufferQueueBlockPool> bqPool =
-                        std::static_pointer_cast<C2BufferQueueBlockPool>(it->second);
-                bqPool->invalidate();
-            }
-        }
+        transform_if(mBlockPools.begin(), mBlockPools.end(), std::back_inserter(bqPools),
+                BqPoolFilterFn, BqPoolConvertFn);
+    }
+    if (!bqPools.empty()) {
+        std::shared_ptr<C2BqPoolInvalidateItem> bqInvalidateItem =
+                std::make_shared<C2BqPoolInvalidateItem>(std::move(bqPools));
+        bqInvalidateItem->invalidate();
     }
     release();
 }
@@ -551,7 +554,26 @@
 }
 
 Return<Status> Component::release() {
+    std::list<std::shared_ptr<C2BufferQueueBlockPool>> bqPools;
+    {
+        std::lock_guard<std::mutex> lock(mBlockPoolsMutex);
+        if (!mClientDied) {
+            transform_if(mBlockPools.begin(), mBlockPools.end(), std::back_inserter(bqPools),
+                    BqPoolFilterFn, BqPoolConvertFn);
+        }
+    }
+    std::shared_ptr<C2BqPoolInvalidateItem> bqInvalidateItem;
+    if (!bqPools.empty()) {
+        // handling rare cases of process death just after release() called.
+        bqInvalidateItem = std::make_shared<C2BqPoolInvalidateItem>(std::move(bqPools));
+        C2BqPoolInvalidator::getInstance().queue(bqInvalidateItem);
+    }
     Status status = static_cast<Status>(mComponent->release());
+    if (bqInvalidateItem) {
+        // If release is not blocked,
+        // skip invalidation and finish ASAP.
+        bqInvalidateItem->skip();
+    }
     {
         std::lock_guard<std::mutex> lock(mBlockPoolsMutex);
         mBlockPools.clear();
@@ -676,6 +698,18 @@
 }
 
 Component::~Component() {
+    std::list<std::shared_ptr<C2BufferQueueBlockPool>> bqPools;
+    {
+        std::lock_guard<std::mutex> lock(mBlockPoolsMutex);
+        transform_if(mBlockPools.begin(), mBlockPools.end(), std::back_inserter(bqPools),
+                BqPoolFilterFn, BqPoolConvertFn);
+    }
+    if (!bqPools.empty()) {
+        LOG(ERROR) << "blockpools are not cleared yet at dtor";
+        std::shared_ptr<C2BqPoolInvalidateItem> bqInvalidateItem =
+                std::make_shared<C2BqPoolInvalidateItem>(std::move(bqPools));
+        C2BqPoolInvalidator::getInstance().queue(bqInvalidateItem);
+    }
     InputBufferManager::unregisterFrameData(mListener);
     mStore->reportComponentDeath(this);
 }
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..a7e043b 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
@@ -56,6 +56,13 @@
 using ::android::sp;
 
 struct ComponentStore : public IComponentStore {
+    /**
+     * Constructor for ComponentStore.
+     *
+     * IMPORTANT: SetPreferredCodec2ComponentStore() is called in the constructor.
+     * Be careful about the order of SetPreferredCodec2ComponentStore() and
+     * ComponentStore() in the code.
+     */
     ComponentStore(const std::shared_ptr<C2ComponentStore>& store);
     virtual ~ComponentStore();
 
@@ -139,6 +146,9 @@
     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 f90363b..5c46d99 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -2063,6 +2063,14 @@
 
 status_t CCodecBufferChannel::requestInitialInputBuffers(
         std::map<size_t, sp<MediaCodecBuffer>> &&clientInputBuffers) {
+    std::optional<QueueGuard> guard;
+    if (android::media::codec::provider_->codec_buffer_state_cleanup()) {
+        guard.emplace(mSync);
+        if (!guard->isRunning()) {
+            ALOGD("[%s] skip requestInitialInputBuffers when not running", mName);
+            return OK;
+        }
+    }
     C2StreamBufferTypeSetting::output oStreamFormat(0u);
     C2PrependHeaderModeSetting prepend(PREPEND_HEADER_TO_NONE);
     c2_status_t err = mComponent->query({ &oStreamFormat, &prepend }, {}, C2_DONT_BLOCK, nullptr);
@@ -2639,8 +2647,6 @@
         switch (action) {
         case OutputBuffers::SKIP:
             return;
-        case OutputBuffers::DISCARD:
-            break;
         case OutputBuffers::NOTIFY_CLIENT:
         {
             // TRICKY: we want popped buffers reported in order, so sending
@@ -2667,13 +2673,16 @@
                     outBuffer->meta()->setObject("accessUnitInfo", obj);
                 }
             }
+            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] onOutputBufferAvailable: numProcessingBuffersBalance = %lld",
-                      mName, static_cast<long long>(inputSurface->numProcessingBuffersBalance));
+                ALOGV("[%s] onWorkDone: numProcessingBuffersBalance = %lld",
+                        mName, static_cast<long long>(inputSurface->numProcessingBuffersBalance));
             }
-            mCallback->onOutputBufferAvailable(index, outBuffer);
             break;
         }
         case OutputBuffers::REALLOCATE:
diff --git a/media/codec2/vndk/Android.bp b/media/codec2/vndk/Android.bp
index dc06ee6..9d1cbff 100644
--- a/media/codec2/vndk/Android.bp
+++ b/media/codec2/vndk/Android.bp
@@ -53,7 +53,7 @@
     ],
 
     defaults: [
-	"aconfig_lib_cc_static_link.defaults",
+        "aconfig_lib_cc_static_link.defaults",
         "libcodec2_hal_selection",
     ],
 
@@ -68,6 +68,7 @@
         "C2PlatformStorePluginLoader.cpp",
         "C2Store.cpp",
         "platform/C2BqBuffer.cpp",
+        "platform/C2BqPoolInvalidator.cpp",
         "platform/C2SurfaceSyncObj.cpp",
         "platform/C2IgbaBuffer.cpp",
         "types.cpp",
diff --git a/media/codec2/vndk/include/C2BqPoolInvalidator.h b/media/codec2/vndk/include/C2BqPoolInvalidator.h
new file mode 100644
index 0000000..612d023
--- /dev/null
+++ b/media/codec2/vndk/include/C2BqPoolInvalidator.h
@@ -0,0 +1,105 @@
+/*
+ * 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.
+ */
+
+#pragma once
+
+#include <android-base/no_destructor.h>
+#include <media/stagefright/foundation/ABase.h>
+
+#include <condition_variable>
+#include <deque>
+#include <list>
+#include <memory>
+#include <thread>
+
+class C2BufferQueueBlockPool;
+
+namespace android {
+
+/**
+ * Container class in order to invalidate C2BufferQueueBlockPool(s) and their resources
+ * when the client process is dead abruptly.
+ */
+class C2BqPoolInvalidateItem {
+public:
+
+    /**
+     * invalidate contained C2BufferQueueBlockPool(s) and their resources
+     */
+    void invalidate();
+
+    /**
+     * skip invalidate(), if it is scheduled and not yet invalidated.
+     */
+    void skip();
+
+    /**
+     * returns whether invalidate() is reuqired or not.
+     */
+    bool needsInvalidate();
+
+    C2BqPoolInvalidateItem(std::list<std::shared_ptr<C2BufferQueueBlockPool>> &&pools);
+
+    ~C2BqPoolInvalidateItem() = default;
+private:
+
+    std::list<std::shared_ptr<C2BufferQueueBlockPool>>  mPools;
+    bool mNeedsInvalidate;
+    std::mutex mLock;
+
+    DISALLOW_EVIL_CONSTRUCTORS(C2BqPoolInvalidateItem);
+};
+
+/**
+ * Asynchronous C2BufferQueueBlockPool invalidator.
+ *
+ * this has C2BqPoolInvalidateItem inside. and call invalidate() from a separate
+ * thread asynchronously.
+ */
+class C2BqPoolInvalidator {
+public:
+    /**
+     * This gets the singleton instance of the class.
+     */
+    static C2BqPoolInvalidator &getInstance();
+
+    /**
+     * queue invalidation items. the item will be invalidated after certain
+     * amount of delay from a separate thread.
+     */
+    void queue(std::shared_ptr<C2BqPoolInvalidateItem> &item);
+
+    ~C2BqPoolInvalidator();
+private:
+
+    C2BqPoolInvalidator();
+
+    void run();
+
+    std::thread mThread;
+    bool mDone;
+
+    std::mutex mMutex;
+    std::condition_variable mCv;
+
+    std::deque<std::pair<int64_t, std::shared_ptr<C2BqPoolInvalidateItem>>> mItems;
+
+    friend class ::android::base::NoDestructor<C2BqPoolInvalidator>;
+
+    DISALLOW_EVIL_CONSTRUCTORS(C2BqPoolInvalidator);
+};
+
+}  // namespace android
diff --git a/media/codec2/vndk/platform/C2BqPoolInvalidator.cpp b/media/codec2/vndk/platform/C2BqPoolInvalidator.cpp
new file mode 100644
index 0000000..2666cd3
--- /dev/null
+++ b/media/codec2/vndk/platform/C2BqPoolInvalidator.cpp
@@ -0,0 +1,130 @@
+/*
+ * 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.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "C2BqPoolInvalidator"
+#include <utils/Log.h>
+#include <utils/SystemClock.h>
+
+#include <C2BqBufferPriv.h>
+#include <C2BqPoolInvalidator.h>
+
+namespace android {
+
+namespace {
+    static constexpr int64_t kBqPoolInvalidateDelayMs = 1000;
+} // anonymous namespace
+
+C2BqPoolInvalidateItem::C2BqPoolInvalidateItem(
+        std::list<std::shared_ptr<C2BufferQueueBlockPool>> &&pools) : mPools(std::move(pools)) {
+    if (!mPools.empty()) {
+        mNeedsInvalidate = true;
+    } else {
+        mNeedsInvalidate = false;
+    }
+}
+
+void C2BqPoolInvalidateItem::invalidate() {
+    std::list<std::shared_ptr<C2BufferQueueBlockPool>> pools;
+    {
+        std::unique_lock<std::mutex> l(mLock);
+        if (!mNeedsInvalidate) {
+            return;
+        }
+        pools = std::move(mPools);
+        mNeedsInvalidate = false;
+    }
+    for(auto it = pools.begin(); it != pools.end(); ++it) {
+        (*it)->invalidate();
+    }
+}
+
+void C2BqPoolInvalidateItem::skip() {
+    std::unique_lock<std::mutex> l(mLock);
+    mNeedsInvalidate = false;
+    mPools.clear();
+}
+
+bool C2BqPoolInvalidateItem::needsInvalidate() {
+    std::unique_lock<std::mutex> l(mLock);
+    return mNeedsInvalidate;
+}
+
+C2BqPoolInvalidator &C2BqPoolInvalidator::getInstance() {
+    static android::base::NoDestructor<C2BqPoolInvalidator> sInvalidator;
+    return *sInvalidator;
+}
+
+C2BqPoolInvalidator::C2BqPoolInvalidator() : mDone(false) {
+    mThread = std::thread(&C2BqPoolInvalidator::run, this);
+}
+
+C2BqPoolInvalidator::~C2BqPoolInvalidator() {
+    {
+        std::unique_lock<std::mutex> l(mMutex);
+        mDone = true;
+        mCv.notify_one();
+    }
+    if (mThread.joinable()) {
+        mThread.join();
+    }
+}
+
+void C2BqPoolInvalidator::queue(std::shared_ptr<C2BqPoolInvalidateItem> &item) {
+    std::unique_lock<std::mutex> l(mMutex);
+    std::pair<int64_t, std::shared_ptr<C2BqPoolInvalidateItem>> p =
+            std::make_pair(::android::elapsedRealtime() + kBqPoolInvalidateDelayMs, item);
+    mItems.push_back(p);
+    mCv.notify_one();
+}
+
+void C2BqPoolInvalidator::run() {
+    while(true) {
+        int64_t nowMs = ::android::elapsedRealtime();
+        std::unique_lock<std::mutex> l(mMutex);
+        if (mDone) {
+            break;
+        }
+        std::list<std::shared_ptr<C2BqPoolInvalidateItem>> items;
+        while (!mItems.empty()) {
+            if (mItems.front().first <= nowMs) {
+                items.push_back(mItems.front().second);
+                mItems.pop_front();
+            } else {
+                break;
+            }
+        }
+        if (items.empty()) {
+            if (mItems.empty()) {
+                mCv.wait(l);
+            } else {
+                int64_t nextMs = mItems.front().first;
+                if (nextMs > nowMs) {
+                    mCv.wait_for(l, std::chrono::milliseconds(nextMs - nowMs));
+                }
+            }
+        } else {
+            l.unlock();
+            int invalidated = 0;
+            for (auto it = items.begin(); it != items.end(); ++it, ++invalidated) {
+                (*it)->invalidate();
+            }
+            ALOGD("invalidated %d bqpool items", invalidated);
+        }
+    }
+}
+
+} // android
diff --git a/media/libaaudio/fuzzer/Android.bp b/media/libaaudio/fuzzer/Android.bp
index ba231c1..b369a62 100644
--- a/media/libaaudio/fuzzer/Android.bp
+++ b/media/libaaudio/fuzzer/Android.bp
@@ -66,6 +66,7 @@
         "libmedia_helper",
         "libmediametrics",
         "libprocessgroup",
+        "libprocessgroup_util",
         "mediametricsservice-aidl-cpp",
         "shared-file-region-aidl-cpp",
     ],
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/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index f92103b..ecf7436 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -2209,11 +2209,10 @@
         {   // start of lock scope
             AutoMutex lock(mLock);
 
-            uint32_t newSequence = mSequence;
             // did previous obtainBuffer() fail due to media server death or voluntary invalidation?
             if (status == DEAD_OBJECT) {
                 // re-create track, unless someone else has already done so
-                if (newSequence == oldSequence) {
+                if (mSequence == oldSequence) {
                     status = restoreTrack_l("obtainBuffer");
                     if (status != NO_ERROR) {
                         buffer.mFrameCount = 0;
@@ -2223,7 +2222,7 @@
                     }
                 }
             }
-            oldSequence = newSequence;
+            oldSequence = mSequence;
 
             if (status == NOT_ENOUGH_DATA) {
                 restartIfDisabled();
diff --git a/media/libaudioclient/TEST_MAPPING b/media/libaudioclient/TEST_MAPPING
index 68dba34..29b876c 100644
--- a/media/libaudioclient/TEST_MAPPING
+++ b/media/libaudioclient/TEST_MAPPING
@@ -47,12 +47,7 @@
       "name": "audioeffect_analysis"
     },
     {
-      "name": "CtsVirtualDevicesTestCases",
-      "options" : [
-        {
-          "include-filter": "android.virtualdevice.cts.VirtualAudioTest"
-        }
-      ]
+      "name": "CtsVirtualDevicesAudioTestCases"
     }
   ]
 }
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/aidl/fuzzer/Android.bp b/media/libaudioclient/aidl/fuzzer/Android.bp
index 61d5ccd..a215c0b 100644
--- a/media/libaudioclient/aidl/fuzzer/Android.bp
+++ b/media/libaudioclient/aidl/fuzzer/Android.bp
@@ -30,6 +30,7 @@
         "libjsoncpp",
         "libmediametricsservice",
         "libprocessgroup",
+        "libprocessgroup_util",
         "shared-file-region-aidl-cpp",
     ],
     shared_libs: [
diff --git a/media/libaudioclient/fuzzer/Android.bp b/media/libaudioclient/fuzzer/Android.bp
index a95c700..8bca8df 100644
--- a/media/libaudioclient/fuzzer/Android.bp
+++ b/media/libaudioclient/fuzzer/Android.bp
@@ -46,6 +46,7 @@
         "libmediametrics",
         "libmediametricsservice",
         "libprocessgroup",
+        "libprocessgroup_util",
         "shared-file-region-aidl-cpp",
     ],
     shared_libs: [
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/audiorouting_tests.cpp b/media/libaudioclient/tests/audiorouting_tests.cpp
index 8151d39..a3ab9d2 100644
--- a/media/libaudioclient/tests/audiorouting_tests.cpp
+++ b/media/libaudioclient/tests/audiorouting_tests.cpp
@@ -86,7 +86,18 @@
     }
 }
 
-TEST(AudioTrackTest, DefaultRoutingTest) {
+class AudioTrackTest
+        : public ::testing::TestWithParam<int> {
+
+public:
+    AudioTrackTest()
+            : mSampleRate(GetParam()){};
+
+    const uint32_t mSampleRate;
+
+};
+
+TEST_P(AudioTrackTest, DefaultRoutingTest) {
     audio_port_v7 port;
     if (OK != getPortByAttributes(AUDIO_PORT_ROLE_SOURCE, AUDIO_PORT_TYPE_DEVICE,
                                   AUDIO_DEVICE_IN_REMOTE_SUBMIX, "0", port)) {
@@ -95,7 +106,8 @@
 
     // create record instance
     sp<AudioCapture> capture = sp<AudioCapture>::make(
-            AUDIO_SOURCE_REMOTE_SUBMIX, 48000, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO);
+            AUDIO_SOURCE_REMOTE_SUBMIX, mSampleRate, AUDIO_FORMAT_PCM_16_BIT,
+            AUDIO_CHANNEL_IN_STEREO);
     ASSERT_NE(nullptr, capture);
     ASSERT_EQ(OK, capture->create()) << "record creation failed";
     sp<OnAudioDeviceUpdateNotifier> cbCapture = sp<OnAudioDeviceUpdateNotifier>::make();
@@ -103,7 +115,7 @@
 
     // create playback instance
     sp<AudioPlayback> playback = sp<AudioPlayback>::make(
-            48000 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+            mSampleRate, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
             AUDIO_OUTPUT_FLAG_NONE, AUDIO_SESSION_NONE);
     ASSERT_NE(nullptr, playback);
     ASSERT_EQ(OK, playback->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
@@ -133,6 +145,12 @@
     playback->stop();
 }
 
+INSTANTIATE_TEST_SUITE_P(
+        AudioTrackParameterizedTest,
+        AudioTrackTest,
+        ::testing::Values(44100, 48000)
+);
+
 class AudioRoutingTest : public ::testing::Test {
   public:
     void SetUp() override {
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/DeviceHalHidl.cpp b/media/libaudiohal/impl/DeviceHalHidl.cpp
index 0a262e4..263ef96 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.cpp
+++ b/media/libaudiohal/impl/DeviceHalHidl.cpp
@@ -619,7 +619,14 @@
             result != NO_ERROR) {
         return result;
     }
-    return processReturn("setConnectedState", mDevice->setConnectedState(hidlAddress, connected));
+    Return<Result> ret = mDevice->setConnectedState(hidlAddress, connected);
+    if (ret.isOk() || ret == Result::NOT_SUPPORTED) {
+        // The framework is only interested in errors occurring due to connection state handling,
+        // so it can decide whether retrying is needed. If the HAL does not support this operation,
+        // it's not an error.
+        return NO_ERROR;
+    }
+    return processReturn("setConnectedState", ret);
 }
 
 error::Result<audio_hw_sync_t> DeviceHalHidl::getHwAvSync() {
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/Hal2AidlMapper.cpp b/media/libaudiohal/impl/Hal2AidlMapper.cpp
index f352849..0cdf0f2 100644
--- a/media/libaudiohal/impl/Hal2AidlMapper.cpp
+++ b/media/libaudiohal/impl/Hal2AidlMapper.cpp
@@ -368,16 +368,21 @@
         const AudioConfig& config, const std::optional<AudioIoFlags>& flags, int32_t ioHandle,
         AudioSource source, const std::set<int32_t>& destinationPortIds,
         AudioPortConfig* portConfig, bool* created) {
-    // These flags get removed one by one in this order when retrying port finding.
-    static const std::vector<AudioInputFlags> kOptionalInputFlags{
-        AudioInputFlags::FAST, AudioInputFlags::RAW, AudioInputFlags::VOIP_TX };
     if (auto portConfigIt = findPortConfig(config, flags, ioHandle);
             portConfigIt == mPortConfigs.end() && flags.has_value()) {
-        auto optionalInputFlagsIt = kOptionalInputFlags.begin();
+        // These input flags get removed one by one in this order when retrying port finding.
+        std::vector<AudioInputFlags> optionalInputFlags {
+            AudioInputFlags::FAST, AudioInputFlags::RAW, AudioInputFlags::VOIP_TX };
+        // For remote submix input, retry with direct input flag removed as the remote submix
+        // input is not expected to manipulate the contents of the audio stream.
+        if (mRemoteSubmixIn.has_value()) {
+            optionalInputFlags.push_back(AudioInputFlags::DIRECT);
+        }
+        auto optionalInputFlagsIt = optionalInputFlags.begin();
         AudioIoFlags matchFlags = flags.value();
         auto portsIt = findPort(config, matchFlags, destinationPortIds);
         while (portsIt == mPorts.end() && matchFlags.getTag() == AudioIoFlags::Tag::input
-                && optionalInputFlagsIt != kOptionalInputFlags.end()) {
+                && optionalInputFlagsIt != optionalInputFlags.end()) {
             if (!isBitPositionFlagSet(
                             matchFlags.get<AudioIoFlags::Tag::input>(), *optionalInputFlagsIt)) {
                 ++optionalInputFlagsIt;
@@ -392,6 +397,36 @@
                         config.toString().c_str(), flags.value().toString().c_str(),
                         matchFlags.toString().c_str());
         }
+        // These output flags get removed one by one in this order when retrying port finding.
+        std::vector<AudioOutputFlags> optionalOutputFlags { };
+        // For remote submix output, retry with these output flags removed one by one:
+        // 1. DIRECT: remote submix outputs are expected not to manipulate the contents of the
+        //            audio stream.
+        // 2. IEC958_NONAUDIO: remote submix outputs are not connected to ALSA and do not require
+        //                     non audio signalling.
+        if (mRemoteSubmixOut.has_value()) {
+            optionalOutputFlags.push_back(AudioOutputFlags::DIRECT);
+            optionalOutputFlags.push_back(AudioOutputFlags::IEC958_NONAUDIO);
+        }
+        auto optionalOutputFlagsIt = optionalOutputFlags.begin();
+        matchFlags = flags.value();
+        while (portsIt == mPorts.end() && matchFlags.getTag() == AudioIoFlags::Tag::output
+                && optionalOutputFlagsIt != optionalOutputFlags.end()) {
+            if (!isBitPositionFlagSet(
+                            matchFlags.get<AudioIoFlags::Tag::output>(),*optionalOutputFlagsIt)) {
+                ++optionalOutputFlagsIt;
+                continue;
+            }
+            matchFlags.set<AudioIoFlags::Tag::output>(matchFlags.get<AudioIoFlags::Tag::output>() &
+                    ~makeBitPositionFlagMask(*optionalOutputFlagsIt++));
+            portsIt = findPort(config, matchFlags, destinationPortIds);
+            AUGMENT_LOG(I,
+                        "mix port for config %s, flags %s was not found"
+                        "retried with flags %s",
+                        config.toString().c_str(), flags.value().toString().c_str(),
+                        matchFlags.toString().c_str());
+        }
+
         if (portsIt == mPorts.end()) {
             AUGMENT_LOG(E, "mix port for config %s, flags %s is not found",
                         config.toString().c_str(), matchFlags.toString().c_str());
@@ -792,7 +827,8 @@
     status_t status = prepareToOpenStreamHelper(ioHandle, devicePortConfig.portId,
             devicePortConfig.id, flags, source, initialConfig, cleanups, config,
             mixPortConfig, patch);
-    if (status != OK) {
+    if (status != OK && !(mRemoteSubmixOut.has_value() &&
+                initialConfig.base.format.type != AudioFormatType::PCM)) {
         // If using the client-provided config did not work out for establishing a mix port config
         // or patching, try with the device port config. Note that in general device port config and
         // mix port config are not required to be the same, however they must match if the HAL
diff --git a/media/libaudiohal/impl/StreamHalAidl.cpp b/media/libaudiohal/impl/StreamHalAidl.cpp
index 63be1bc..c4e4ae8 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;
 }
@@ -427,9 +443,29 @@
     if (!mStream) return NO_INIT;
 
     if (const auto state = getState(); isInPlayOrRecordState(state)) {
-        return sendCommand(
-                makeHalCommand<HalCommand::Tag::pause>(), reply,
+        StreamDescriptor::Reply localReply{};
+        StreamDescriptor::Reply* innerReply = reply ?: &localReply;
+        auto status = sendCommand(
+                makeHalCommand<HalCommand::Tag::pause>(), innerReply,
                 true /*safeFromNonWorkerThread*/);  // The workers stops its I/O activity first.
+        if (status == STATUS_INVALID_OPERATION &&
+                !isInPlayOrRecordState(innerReply->state)) {
+            /**
+             * In case of transient states like DRAINING, the HAL may change its
+             * StreamDescriptor::State on its own and may not be in synchronization with client.
+             * Thus, client can send the unexpected command and HAL returns failure. such failure is
+             * natural. The client handles it gracefully.
+             * Example where HAL change its state,
+             * 1) DRAINING -> IDLE (on empty buffer)
+             * 2) DRAINING -> IDLE (on IStreamCallback::onDrainReady)
+             **/
+            AUGMENT_LOG(D,
+                        "HAL failed to handle the 'pause' command, but stream state is in one of"
+                        " the PAUSED kind of states, current state: %s",
+                        toString(state).c_str());
+            return OK;
+        }
+        return status;
     } else {
         AUGMENT_LOG(D, "already stream in one of the PAUSED kind of states, current state: %s",
                 toString(state).c_str());
@@ -457,13 +493,9 @@
                 return INVALID_OPERATION;
             }
             return OK;
-        } else if (state == StreamDescriptor::State::PAUSED ||
-                   state == StreamDescriptor::State::TRANSFER_PAUSED ||
-                   state == StreamDescriptor::State::DRAIN_PAUSED) {
+        } else if (isInPausedState(state)) {
             return sendCommand(makeHalCommand<HalCommand::Tag::start>(), reply);
-        } else if (state == StreamDescriptor::State::ACTIVE ||
-                   state == StreamDescriptor::State::TRANSFERRING ||
-                   state == StreamDescriptor::State::DRAINING) {
+        } else if (isInPlayOrRecordState(state)) {
             AUGMENT_LOG(D, "already in stream state: %s", toString(state).c_str());
             return OK;
         } else {
@@ -512,7 +544,14 @@
 }
 
 void StreamHalAidl::onAsyncTransferReady() {
-    if (auto state = getState(); state == StreamDescriptor::State::TRANSFERRING) {
+    StreamDescriptor::State state;
+    {
+        // Use 'mCommandReplyLock' to ensure that 'sendCommand' has finished updating the state
+        // after the reply from the 'burst' command.
+        std::lock_guard l(mCommandReplyLock);
+        state = getState();
+    }
+    if (state == StreamDescriptor::State::TRANSFERRING) {
         // Retrieve the current state together with position counters unconditionally
         // to ensure that the state on our side gets updated.
         sendCommand(makeHalCommand<HalCommand::Tag::getStatus>(),
@@ -523,7 +562,14 @@
 }
 
 void StreamHalAidl::onAsyncDrainReady() {
-    if (auto state = getState(); state == StreamDescriptor::State::DRAINING) {
+    StreamDescriptor::State state;
+    {
+        // Use 'mCommandReplyLock' to ensure that 'sendCommand' has finished updating the state
+        // after the reply from the 'drain' command.
+        std::lock_guard l(mCommandReplyLock);
+        state = getState();
+    }
+    if (state == StreamDescriptor::State::DRAINING) {
         // Retrieve the current state together with position counters unconditionally
         // to ensure that the state on our side gets updated.
         sendCommand(makeHalCommand<HalCommand::Tag::getStatus>(), nullptr,
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/libmedia/include/media/mediametadataretriever.h b/media/libmedia/include/media/mediametadataretriever.h
index 116ed9a..d76ed25 100644
--- a/media/libmedia/include/media/mediametadataretriever.h
+++ b/media/libmedia/include/media/mediametadataretriever.h
@@ -122,6 +122,10 @@
     static sp<IMediaPlayerService>            sService;
 
     Mutex                                     mLock;
+    // Static lock was added to the client in order to consume at most
+    // one service thread from image extraction requests of the same
+    // client process(See also b/21277449).
+    static Mutex                              sLock;
     sp<IMediaMetadataRetriever>               mRetriever;
 
 };
diff --git a/media/libmedia/mediametadataretriever.cpp b/media/libmedia/mediametadataretriever.cpp
index 40fd022..9196f9f 100644
--- a/media/libmedia/mediametadataretriever.cpp
+++ b/media/libmedia/mediametadataretriever.cpp
@@ -35,6 +35,8 @@
 sp<IMediaPlayerService> MediaMetadataRetriever::sService;
 sp<MediaMetadataRetriever::DeathNotifier> MediaMetadataRetriever::sDeathNotifier;
 
+Mutex MediaMetadataRetriever::sLock;
+
 const sp<IMediaPlayerService> MediaMetadataRetriever::getService()
 {
     Mutex::Autolock lock(sServiceLock);
@@ -143,6 +145,7 @@
     ALOGV("getFrameAtTime: time(%" PRId64 " us) option(%d) colorFormat(%d) metaOnly(%d)",
             timeUs, option, colorFormat, metaOnly);
     Mutex::Autolock _l(mLock);
+    Mutex::Autolock _gLock(sLock);
     if (mRetriever == 0) {
         ALOGE("retriever is not initialized");
         return NULL;
@@ -155,6 +158,7 @@
     ALOGV("getImageAtIndex: index(%d) colorFormat(%d) metaOnly(%d) thumbnail(%d)",
             index, colorFormat, metaOnly, thumbnail);
     Mutex::Autolock _l(mLock);
+    Mutex::Autolock _gLock(sLock);
     if (mRetriever == 0) {
         ALOGE("retriever is not initialized");
         return NULL;
@@ -167,6 +171,7 @@
     ALOGV("getImageRectAtIndex: index(%d) colorFormat(%d) rect {%d, %d, %d, %d}",
             index, colorFormat, left, top, right, bottom);
     Mutex::Autolock _l(mLock);
+    Mutex::Autolock _gLock(sLock);
     if (mRetriever == 0) {
         ALOGE("retriever is not initialized");
         return NULL;
@@ -180,6 +185,7 @@
     ALOGV("getFrameAtIndex: index(%d), colorFormat(%d) metaOnly(%d)",
             index, colorFormat, metaOnly);
     Mutex::Autolock _l(mLock);
+    Mutex::Autolock _gLock(sLock);
     if (mRetriever == 0) {
         ALOGE("retriever is not initialized");
         return NULL;
diff --git a/media/libmedia/xsd/vts/Android.bp b/media/libmedia/xsd/vts/Android.bp
index 83ab977..add7b51 100644
--- a/media/libmedia/xsd/vts/Android.bp
+++ b/media/libmedia/xsd/vts/Android.bp
@@ -15,6 +15,7 @@
 //
 
 package {
+    default_team: "trendy_team_android_kernel",
     // See: http://go/android-license-faq
     // A large-scale-change added 'default_applicable_licenses' to import
     // all of the 'license_kinds' from "frameworks_av_media_libmedia_license"
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/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 10a1da7..761137e 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -2514,6 +2514,15 @@
     {
         Mutex::Autolock lock(mLock);
         track = mTrack;
+    }
+
+    // do not hold lock while joining.
+    if (track) {
+        track->stopAndJoinCallbacks();
+    }
+
+    {
+        Mutex::Autolock lock(mLock);
         close_l(); // clears mTrack
     }
     // destruction of the track occurs outside of mutex.
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/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index eea5242..7d47837 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -2015,6 +2015,7 @@
     int32_t flags;
     CHECK(buffer->meta()->findInt32("flags", &flags));
     if (flags & BUFFER_FLAG_DECODE_ONLY) {
+        ALOGV("discardDecodeOnlyOutputBuffer: mPortBuffers[out][%zu] NOT owned by client", index);
         info->mOwnedByClient = false;
         info->mData.clear();
         mBufferChannel->discardBuffer(buffer);
@@ -4494,9 +4495,16 @@
                 {
                     /* size_t index = */updateBuffers(kPortIndexInput, msg);
 
-                    if (mState == FLUSHING
-                            || mState == STOPPING
-                            || mState == RELEASING) {
+                    bool inStateToReturnBuffers =
+                        mState == FLUSHING || mState == STOPPING || mState == RELEASING;
+                    if (android::media::codec::provider_->codec_buffer_state_cleanup()) {
+                        // Late callbacks from the codec could arrive here
+                        // after the codec is already stopped or released.
+                        inStateToReturnBuffers = mState == FLUSHING ||
+                                                 mState == STOPPING || mState == INITIALIZED ||
+                                                 mState == RELEASING || mState == UNINITIALIZED;
+                    }
+                    if (inStateToReturnBuffers) {
                         returnBuffersToCodecOnPort(kPortIndexInput);
                         break;
                     }
@@ -4575,9 +4583,16 @@
 
                     /* size_t index = */updateBuffers(kPortIndexOutput, msg);
 
-                    if (mState == FLUSHING
-                            || mState == STOPPING
-                            || mState == RELEASING) {
+                    bool inStateToReturnBuffers =
+                        mState == FLUSHING || mState == STOPPING || mState == RELEASING;
+                    if (android::media::codec::provider_->codec_buffer_state_cleanup()) {
+                        // Late callbacks from the codec could arrive here
+                        // after the codec is already stopped or released.
+                        inStateToReturnBuffers = mState == FLUSHING ||
+                                                 mState == STOPPING || mState == INITIALIZED ||
+                                                 mState == RELEASING || mState == UNINITIALIZED;
+                    }
+                    if (inStateToReturnBuffers) {
                         returnBuffersToCodecOnPort(kPortIndexOutput);
                         break;
                     }
@@ -5943,7 +5958,7 @@
     }
 
     updateHdrMetrics(false /* isConfig */);
- }
+}
 
 void MediaCodec::extractCSD(const sp<AMessage> &format) {
     mCSD.clear();
@@ -6022,7 +6037,6 @@
             return -EINVAL;
         }
         if (codecInputData->data() == NULL) {
-            ALOGV("Input buffer %zu is not properly allocated", bufferIndex);
             mErrorLog.log(LOG_TAG, base::StringPrintf(
                     "Fatal error: input buffer %zu is not properly allocated", bufferIndex));
             return -EINVAL;
@@ -6068,6 +6082,10 @@
 
         mInputFormat.clear();
         mOutputFormat.clear();
+        if (android::media::codec::provider_->codec_buffer_state_cleanup()) {
+            mCSD.clear();
+            mLeftover.clear();
+        }
         mFlags &= ~kFlagOutputFormatChanged;
         mFlags &= ~kFlagOutputBuffersChanged;
         mFlags &= ~kFlagStickyError;
@@ -6126,6 +6144,8 @@
                 ALOGD("port %d buffer %zu still owned by client when codec is reclaimed",
                         portIndex, i);
             } else {
+                ALOGV("returnBuffersToCodecOnPort: mPortBuffers[%s][%zu] NOT owned by client",
+                      portIndex == kPortIndexInput ? "in" : "out", i);
                 info->mOwnedByClient = false;
                 info->mData.clear();
             }
@@ -6478,6 +6498,7 @@
 
         // synchronization boundary for getBufferAndFormat
         Mutex::Autolock al(mBufferLock);
+        ALOGV("onQueueInputBuffer: mPortBuffers[in][%zu] NOT owned by client", index);
         info->mOwnedByClient = false;
         info->mData.clear();
 
@@ -6494,6 +6515,7 @@
     sp<AMessage> msg = mLeftover.front();
     mLeftover.pop_front();
     msg->setSize("index", index);
+    ALOGV("handleLeftover(%zu)", index);
     return onQueueInputBuffer(msg);
 }
 
@@ -6562,6 +6584,7 @@
     sp<MediaCodecBuffer> buffer;
     {
         Mutex::Autolock al(mBufferLock);
+        ALOGV("onReleaseOutputBuffer: mPortBuffers[out][%zu] NOT owned by client", index);
         info->mOwnedByClient = false;
         buffer = info->mData;
         info->mData.clear();
@@ -6674,6 +6697,8 @@
 
     {
         Mutex::Autolock al(mBufferLock);
+        ALOGV("dequeuePortBuffer: mPortBuffers[%s][%zu] checking if not owned by client",
+              portIndex == kPortIndexInput ? "in" : "out", index);
         CHECK(!info->mOwnedByClient);
         info->mOwnedByClient = true;
 
diff --git a/media/libstagefright/TEST_MAPPING b/media/libstagefright/TEST_MAPPING
index b7efbce..354fab0 100644
--- a/media/libstagefright/TEST_MAPPING
+++ b/media/libstagefright/TEST_MAPPING
@@ -85,13 +85,37 @@
     // writerTest fails about 5 out of 66
     // { "name": "writerTest" },
     {
-       "name": "BatteryChecker_test"
+        "name": "BatteryChecker_test"
     },
     {
         "name": "ExtractorFactoryTest"
     },
     {
         "name": "HEVCUtilsUnitTest"
+    },
+    {
+      "name": "MctsMediaDecoderTestCases",
+      "options": [
+        {
+          "include-annotation": "android.platform.test.annotations.Presubmit"
+        }
+      ]
+    },
+    {
+      "name": "MctsMediaEncoderTestCases",
+      "options": [
+        {
+          "include-annotation": "android.platform.test.annotations.Presubmit"
+        }
+      ]
+    },
+    {
+      "name": "MctsMediaCodecTestCases",
+      "options": [
+        {
+          "include-annotation": "android.platform.test.annotations.Presubmit"
+        }
+      ]
     }
   ]
 }
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/libstagefright/xmlparser/vts/Android.bp b/media/libstagefright/xmlparser/vts/Android.bp
index 1e36c8f..527230c 100644
--- a/media/libstagefright/xmlparser/vts/Android.bp
+++ b/media/libstagefright/xmlparser/vts/Android.bp
@@ -15,6 +15,7 @@
 //
 
 package {
+    default_team: "trendy_team_android_media_codec_framework",
     // See: http://go/android-license-faq
     // A large-scale-change added 'default_applicable_licenses' to import
     // all of the 'license_kinds' from "frameworks_av_media_libstagefright_license"
diff --git a/media/mediaserver/Android.bp b/media/mediaserver/Android.bp
index 6ea40e3..b5124d0 100644
--- a/media/mediaserver/Android.bp
+++ b/media/mediaserver/Android.bp
@@ -51,9 +51,16 @@
     ],
 }
 
+vintf_fragment {
+    name: "manifest_media_c2_software.xml",
+    src: "manifest_media_c2_software.xml",
+}
+
 mediaserver_cc_binary {
     name: "mediaserver",
 
+    defaults: ["libcodec2_hal_selection"],
+
     srcs: ["main_mediaserver.cpp"],
 
     shared_libs: [
@@ -61,6 +68,7 @@
         "libicu",
         "libfmq",
         "libbinder",
+        "libbinder_ndk",
         "libhidlbase",
         "liblog",
         "libmediaplayerservice",
@@ -85,7 +93,7 @@
         "-Wall",
     ],
 
-    vintf_fragments: ["manifest_media_c2_software.xml"],
+    vintf_fragment_modules: ["manifest_media_c2_software.xml"],
 
     soong_config_variables: {
         TARGET_DYNAMIC_64_32_MEDIASERVER: {
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/common/include/basic_op_c_equivalent.h b/media/module/codecs/amrnb/common/include/basic_op_c_equivalent.h
index 8817621..64fdfb9 100644
--- a/media/module/codecs/amrnb/common/include/basic_op_c_equivalent.h
+++ b/media/module/codecs/amrnb/common/include/basic_op_c_equivalent.h
@@ -115,7 +115,6 @@
      Returns:
         L_sum = 32-bit sum of L_var1 and L_var2 (Word32)
     */
-    __attribute__((no_sanitize("integer")))
     static inline Word32 L_add(Word32 L_var1, Word32 L_var2, Flag *pOverflow)
     {
         Word32 L_sum;
@@ -454,7 +453,8 @@
     {
         Word32 result;
 
-        result = L_var3 + L_var1 * L_var2;
+        __builtin_mul_overflow(L_var1, L_var2, &result);
+        __builtin_add_overflow(L_var3, result, &result);
 
         return result;
     }
@@ -463,7 +463,8 @@
     {
         Word32 result;
 
-        result = L_var3 - L_var1 * L_var2;
+        __builtin_mul_overflow(L_var1, L_var2, &result);
+        __builtin_sub_overflow(L_var3, result, &result);
 
         return result;
     }
diff --git a/media/module/codecs/amrnb/common/src/az_lsp.cpp b/media/module/codecs/amrnb/common/src/az_lsp.cpp
index f3098f5..a19ddbf 100644
--- a/media/module/codecs/amrnb/common/src/az_lsp.cpp
+++ b/media/module/codecs/amrnb/common/src/az_lsp.cpp
@@ -237,9 +237,6 @@
 
 ------------------------------------------------------------------------------
 */
-#ifdef __clang__
-__attribute__((no_sanitize("integer")))
-#endif
 static Word16 Chebps(Word16 x,
                      Word16 f[], /* (n) */
                      Word16 n,
diff --git a/media/module/codecs/amrnb/common/src/l_abs.cpp b/media/module/codecs/amrnb/common/src/l_abs.cpp
index 7e0ae99..b13a40a 100644
--- a/media/module/codecs/amrnb/common/src/l_abs.cpp
+++ b/media/module/codecs/amrnb/common/src/l_abs.cpp
@@ -186,8 +186,12 @@
     ; Function body here
     ----------------------------------------------------------------------------*/
 
-    Word32 y = L_var1 - (L_var1 < 0);
-    y = y ^(y >> 31);
-    return (y);
+    if (L_var1 >= 0) return L_var1;
+    if (L_var1 != 0x80000000) return -L_var1;
+    // abs(0x80000000) can not be represented in Word32.
+    // we choose to return the closest value we can -- 0x7fffffff
+    // This is acceptable because it keeps the result within the valid 32-bit signed integer range,
+    // consistent with other overflow handling in the code. such as amrnb/enc/src/l_negate.cpp.
+    return 0x7FFFFFFF;
 
 }
diff --git a/media/module/codecs/amrnb/common/src/lsp_az.cpp b/media/module/codecs/amrnb/common/src/lsp_az.cpp
index 495359f..bb8a34d 100644
--- a/media/module/codecs/amrnb/common/src/lsp_az.cpp
+++ b/media/module/codecs/amrnb/common/src/lsp_az.cpp
@@ -281,8 +281,8 @@
             t0 += ((Word32)lo * *lsp) >> 15;
 
             *(f) +=  *(f - 2);          /*      *f += f[-2]      */
-            *(f--) -=  t0 << 2;         /*      *f -= t0         */
-
+            __builtin_sub_overflow(*(f), (t0 << 2), f);   /*      *f -= t0         */
+            f--;
         }
 
         *f -= (Word32)(*lsp++) << 10;
diff --git a/media/module/codecs/amrnb/common/src/norm_l.cpp b/media/module/codecs/amrnb/common/src/norm_l.cpp
index d8d1259..b24ebda 100644
--- a/media/module/codecs/amrnb/common/src/norm_l.cpp
+++ b/media/module/codecs/amrnb/common/src/norm_l.cpp
@@ -211,8 +211,7 @@
     if (L_var1)
     {
 
-        Word32 y = L_var1 - (L_var1 < 0);
-        L_var1 = y ^(y >> 31);
+        L_var1 = L_abs(L_var1);
 
 
         while (!(0x40000000L & L_var1))
diff --git a/media/module/codecs/amrnb/common/src/residu.cpp b/media/module/codecs/amrnb/common/src/residu.cpp
index 2ad132f..9b077e2 100644
--- a/media/module/codecs/amrnb/common/src/residu.cpp
+++ b/media/module/codecs/amrnb/common/src/residu.cpp
@@ -227,22 +227,35 @@
         p_input3 = p_input_ptr--;
         p_input4 = p_input_ptr--;
 
+        Word32 tmp;
         for (j = M >> 1; j != 0; j--)
         {
-            s1 += ((Word32) * (p_coef) * *(p_input1++));
-            s2 += ((Word32) * (p_coef) * *(p_input2++));
-            s3 += ((Word32) * (p_coef) * *(p_input3++));
-            s4 += ((Word32) * (p_coef--) * *(p_input4++));
-            s1 += ((Word32) * (p_coef) * *(p_input1++));
-            s2 += ((Word32) * (p_coef) * *(p_input2++));
-            s3 += ((Word32) * (p_coef) * *(p_input3++));
-            s4 += ((Word32) * (p_coef--) * *(p_input4++));
+            __builtin_mul_overflow(*p_coef, *(p_input1++), &tmp);
+            __builtin_add_overflow(s1, tmp, &s1);
+            __builtin_mul_overflow(*p_coef, *(p_input2++), &tmp);
+            __builtin_add_overflow(s2, tmp, &s2);
+            __builtin_mul_overflow(*p_coef, *(p_input3++), &tmp);
+            __builtin_add_overflow(s3, tmp, &s3);
+            __builtin_mul_overflow(*(p_coef--), *(p_input4++), &tmp);
+            __builtin_add_overflow(s4, tmp, &s4);
+            __builtin_mul_overflow(*p_coef, *(p_input1++), &tmp);
+            __builtin_add_overflow(s1, tmp, &s1);
+            __builtin_mul_overflow(*p_coef, *(p_input2++), &tmp);
+            __builtin_add_overflow(s2, tmp, &s2);
+            __builtin_mul_overflow(*p_coef, *(p_input3++), &tmp);
+            __builtin_add_overflow(s3, tmp, &s3);
+            __builtin_mul_overflow(*(p_coef--), *(p_input4++), &tmp);
+            __builtin_add_overflow(s4, tmp, &s4);
         }
 
-        s1 += (((Word32) * (p_coef)) * *(p_input1));
-        s2 += (((Word32) * (p_coef)) * *(p_input2));
-        s3 += (((Word32) * (p_coef)) * *(p_input3));
-        s4 += (((Word32) * (p_coef)) * *(p_input4));
+        __builtin_mul_overflow(*p_coef, *(p_input1), &tmp);
+        __builtin_add_overflow(s1, tmp, &s1);
+        __builtin_mul_overflow(*p_coef, *(p_input2), &tmp);
+        __builtin_add_overflow(s2, tmp, &s2);
+        __builtin_mul_overflow(*p_coef, *(p_input3), &tmp);
+        __builtin_add_overflow(s3, tmp, &s3);
+        __builtin_mul_overflow(*p_coef, *(p_input4), &tmp);
+        __builtin_add_overflow(s4, tmp, &s4);
 
         *(p_residual_ptr--) = (Word16)(s1 >> 12);
         *(p_residual_ptr--) = (Word16)(s2 >> 12);
diff --git a/media/module/codecs/amrnb/common/src/sub.cpp b/media/module/codecs/amrnb/common/src/sub.cpp
index b956912..d936128 100644
--- a/media/module/codecs/amrnb/common/src/sub.cpp
+++ b/media/module/codecs/amrnb/common/src/sub.cpp
@@ -187,9 +187,6 @@
 ; FUNCTION CODE
 ----------------------------------------------------------------------------*/
 
-#ifdef __clang__
-__attribute__((no_sanitize("integer")))
-#endif
 Word16 sub(Word16 var1, Word16 var2, Flag *pOverflow)
 {
 
diff --git a/media/module/codecs/amrnb/common/src/syn_filt.cpp b/media/module/codecs/amrnb/common/src/syn_filt.cpp
index 36c1d84..82770f1 100644
--- a/media/module/codecs/amrnb/common/src/syn_filt.cpp
+++ b/media/module/codecs/amrnb/common/src/syn_filt.cpp
@@ -245,9 +245,6 @@
 
 ------------------------------------------------------------------------------
 */
-#ifdef __clang__
-__attribute__((no_sanitize("integer")))
-#endif
 void Syn_filt(
     Word16 a[],     /* (i)   : a[M+1] prediction coefficients   (M=10)  */
     Word16 x[],     /* (i)   : input signal                             */
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/src/autocorr.cpp b/media/module/codecs/amrnb/enc/src/autocorr.cpp
index c71811d..a078f5a 100644
--- a/media/module/codecs/amrnb/enc/src/autocorr.cpp
+++ b/media/module/codecs/amrnb/enc/src/autocorr.cpp
@@ -312,6 +312,7 @@
 
     Word16 y[L_WINDOW];
     Word32 sum;
+    Word32 mul;
     Word16 overfl_shft;
 
 
@@ -343,7 +344,8 @@
         temp = (amrnb_fxp_mac_16_by_16bb((Word32) * (p_x++), (Word32) * (p_wind++), 0x04000)) >> 15;
         *(p_y++) = temp;
 
-        sum += ((Word32)temp * temp) << 1;
+        __builtin_mul_overflow(temp, temp, &mul);
+        __builtin_add_overflow(sum, mul << 1, &sum);
         if (sum < 0)
         {
             /*
@@ -395,10 +397,12 @@
         {
             temp = *p_y >> 2;
             *(p_y++) = temp;
-            sum += ((Word32)temp * temp) << 1;
+            __builtin_mul_overflow(temp, temp, &mul);
+            __builtin_add_overflow(sum, mul << 1, &sum);
             temp = *p_y >> 2;
             *(p_y++) = temp;
-            sum += ((Word32)temp * temp) << 1;
+            __builtin_mul_overflow(temp, temp, &mul);
+            __builtin_add_overflow(sum, mul << 1, &sum);
         }
         if (sum > 0)
         {
diff --git a/media/module/codecs/amrnb/enc/src/c2_9pf.cpp b/media/module/codecs/amrnb/enc/src/c2_9pf.cpp
index b211032..56b4fb8 100644
--- a/media/module/codecs/amrnb/enc/src/c2_9pf.cpp
+++ b/media/module/codecs/amrnb/enc/src/c2_9pf.cpp
@@ -610,6 +610,7 @@
         Word32 alp1;
         Word16 i;
         Word32 L_temp;
+        Word32 mul;
         Word16 *p_codvec = &codvec[0];
 
         OSCL_UNUSED_ARG(pOverflow);
@@ -693,7 +694,8 @@
                     L_temp = ((Word32) alp * sq1) << 1;
 
                     /* s = L_msu(L_temp, sq, alp_16, pOverflow); */
-                    s = L_temp - (((Word32) sq * alp_16) << 1);
+                    __builtin_mul_overflow(sq, alp_16, &mul);
+                    __builtin_sub_overflow(L_temp, (mul << 1), &s);
 
                     if (s > 0)
                     {
diff --git a/media/module/codecs/amrnb/enc/src/c3_14pf.cpp b/media/module/codecs/amrnb/enc/src/c3_14pf.cpp
index 58ab2fa..bb4fe36 100644
--- a/media/module/codecs/amrnb/enc/src/c3_14pf.cpp
+++ b/media/module/codecs/amrnb/enc/src/c3_14pf.cpp
@@ -403,6 +403,7 @@
     Word16 *p_codvec = &codvec[0];
 
     Word32 s;
+    Word32 mul;
     Word32 alp0;
     Word32 alp1;
 
@@ -487,7 +488,8 @@
                             s = ((Word32) alp * sq1) << 1;
 
                             /* s = L_msu(s, sq, alp_16, pOverflow); */
-                            s -= (((Word32) sq * alp_16) << 1);
+                            __builtin_mul_overflow(sq, alp_16, &mul);
+                            __builtin_sub_overflow(s, (mul << 1), &s);
 
                             if (s > 0)
                             {
diff --git a/media/module/codecs/amrnb/enc/src/c4_17pf.cpp b/media/module/codecs/amrnb/enc/src/c4_17pf.cpp
index d52b43b..062ee5a 100644
--- a/media/module/codecs/amrnb/enc/src/c4_17pf.cpp
+++ b/media/module/codecs/amrnb/enc/src/c4_17pf.cpp
@@ -416,6 +416,7 @@
         Word16 *p_codvec = &codvec[0];
 
         Word32 s;
+        Word32 mul;
         Word32 alp0;
         Word32 alp1;
 
@@ -497,7 +498,8 @@
                             s = ((Word32) alp * sq1) << 1;
 
                             /* s = L_msu(s, sq, alp_16, pOverflow); */
-                            s -= (((Word32) sq * alp_16) << 1);
+                            __builtin_mul_overflow(sq, alp_16, &mul);
+                            __builtin_sub_overflow(s, (mul << 1), &s);
 
                             if (s > 0)
                             {
@@ -610,7 +612,8 @@
                             s = ((Word32) alp * sq1) << 1;
 
                             /* s = L_msu(s, sq, alp_16, pOverflow); */
-                            s -= (((Word32) sq * alp_16) << 1);
+                            __builtin_mul_overflow(sq, alp_16, &mul);
+                            __builtin_sub_overflow(s, (mul << 1), &s);
 
                             if (s > 0)
                             {
@@ -630,7 +633,8 @@
                         s = ((Word32) alpk * sq) << 1;
 
                         /* s = L_msu(s, psk, alp, pOverflow); */
-                        s -= (((Word32) psk * alp) << 1);
+                        __builtin_mul_overflow(psk, alp, &mul);
+                        __builtin_sub_overflow(s, (mul << 1), &s);
 
                         if (s > 0)
                         {
diff --git a/media/module/codecs/amrnb/enc/src/cor_h_x.cpp b/media/module/codecs/amrnb/enc/src/cor_h_x.cpp
index c25c026..398c71f 100644
--- a/media/module/codecs/amrnb/enc/src/cor_h_x.cpp
+++ b/media/module/codecs/amrnb/enc/src/cor_h_x.cpp
@@ -254,6 +254,7 @@
     Word16 k;
 
     Word32 s;
+    Word32 mul;
     Word32 y32[L_CODE];
     Word32 max;
     Word32 tot;
@@ -275,15 +276,19 @@
 
             for (j = (L_CODE - i - 1) >> 1; j != 0; j--)
             {
-                s += ((Word32) * (p_x++) * *(p_ptr++)) << 1;
-                s += ((Word32) * (p_x++) * *(p_ptr++)) << 1;
+                __builtin_mul_overflow(*(p_x++), *(p_ptr++), &mul);
+                __builtin_add_overflow(s, mul << 1, &s);
+                __builtin_mul_overflow(*(p_x++), *(p_ptr++), &mul);
+                __builtin_add_overflow(s, mul << 1, &s);
             }
 
-            s += ((Word32) * (p_x++) * *(p_ptr++)) << 1;
+            __builtin_mul_overflow(*(p_x++), *(p_ptr++), &mul);
+            __builtin_add_overflow(s, mul << 1, &s);
 
             if (!((L_CODE - i) & 1))    /* if even number of iterations */
             {
-                s += ((Word32) * (p_x++) * *(p_ptr++)) << 1;
+                __builtin_mul_overflow(*(p_x++), *(p_ptr++), &mul);
+                __builtin_add_overflow(s, mul << 1, &s);
             }
 
             y32[i] = s;
@@ -299,7 +304,7 @@
             }
         }
 
-        tot += (max >> 1);
+        __builtin_add_overflow(tot, (max >> 1), &tot);
     }
 
 
@@ -310,10 +315,13 @@
 
     for (i = L_CODE >> 1; i != 0; i--)
     {
+        Word32 result;
         s = L_shl(*(p_y32++), j, pOverflow);
-        *(p_ptr++) = (s + 0x00008000) >> 16;
+        __builtin_add_overflow(s, 0x00008000, &result);
+        *(p_ptr++) = result >> 16;
         s = L_shl(*(p_y32++), j, pOverflow);
-        *(p_ptr++) = (s + 0x00008000) >> 16;
+        __builtin_add_overflow(s, 0x00008000, &result);
+        *(p_ptr++) = result >> 16;
     }
 
     return;
diff --git a/media/module/codecs/amrnb/enc/src/cor_h_x2.cpp b/media/module/codecs/amrnb/enc/src/cor_h_x2.cpp
index e32eb4a..80ebb73 100644
--- a/media/module/codecs/amrnb/enc/src/cor_h_x2.cpp
+++ b/media/module/codecs/amrnb/enc/src/cor_h_x2.cpp
@@ -268,7 +268,7 @@
                 max = s;
             }
         }
-        tot = (tot + (max >> 1));
+        __builtin_add_overflow(tot, (max >> 1), &tot);
     }
 
     j = sub(norm_l(tot), sf, pOverflow);
diff --git a/media/module/codecs/amrnb/enc/src/dtx_enc.cpp b/media/module/codecs/amrnb/enc/src/dtx_enc.cpp
index 2ccb777..0d56c9b 100644
--- a/media/module/codecs/amrnb/enc/src/dtx_enc.cpp
+++ b/media/module/codecs/amrnb/enc/src/dtx_enc.cpp
@@ -945,6 +945,7 @@
 
     Word16 i;
     Word32 L_frame_en;
+    Word32 mul;
     Word32 L_temp;
     Word16 log_en_e;
     Word16 log_en_m;
@@ -967,7 +968,8 @@
 
     for (i = L_FRAME; i != 0; i--)
     {
-        L_frame_en += (((Word32) * p_speech) * *(p_speech)) << 1;
+        __builtin_mul_overflow(*p_speech, *p_speech, &mul);
+        __builtin_add_overflow(L_frame_en, mul << 1, &L_frame_en);
         p_speech++;
         if (L_frame_en < 0)
         {
diff --git a/media/module/codecs/amrnb/enc/src/levinson.cpp b/media/module/codecs/amrnb/enc/src/levinson.cpp
index 29cdac6..83dd81e 100644
--- a/media/module/codecs/amrnb/enc/src/levinson.cpp
+++ b/media/module/codecs/amrnb/enc/src/levinson.cpp
@@ -731,7 +731,7 @@
         t0 = t0 << 5;
 
         t1 = ((Word32) * (Rh + i) << 16) + ((Word32)(*(Rl + i)) << 1);
-        t0 += t1;
+        __builtin_add_overflow(t0, t1, &t0);
 
         /* K = -t0 / Alpha */
 
diff --git a/media/module/codecs/amrnb/enc/src/pitch_fr.cpp b/media/module/codecs/amrnb/enc/src/pitch_fr.cpp
index 584f79b..ab0a221 100644
--- a/media/module/codecs/amrnb/enc/src/pitch_fr.cpp
+++ b/media/module/codecs/amrnb/enc/src/pitch_fr.cpp
@@ -326,6 +326,7 @@
     Word16 norm_h;
     Word16 norm_l;
     Word32 s;
+    Word32 mul;
     Word32 s2;
     Word16 excf[L_SUBFR];
     Word16 scaling;
@@ -353,10 +354,12 @@
     {
         temp = *(p_excf++);
         *(p_s_excf++) = temp >> 2;
-        s += (Word32) temp * temp;
+        __builtin_mul_overflow(temp, temp, &mul);
+        __builtin_add_overflow(s, mul, &s);
         temp = *(p_excf++);
         *(p_s_excf++) = temp >> 2;
-        s += (Word32) temp * temp;
+        __builtin_mul_overflow(temp, temp, &mul);
+        __builtin_add_overflow(s, mul, &s);
     }
 
 
@@ -387,20 +390,24 @@
 
         while (j--)
         {
-            s  += (Word32) * (p_x++) * *(p_s_excf);
-            s2 += ((Word32)(*(p_s_excf)) * (*(p_s_excf)));
+            __builtin_mul_overflow(*(p_x++), *p_s_excf, &mul);
+            __builtin_add_overflow(s, mul, &s);
+            __builtin_mul_overflow(*p_s_excf, *p_s_excf, &mul);
+            __builtin_add_overflow(s2, mul, &s2);
             p_s_excf++;
-            s  += (Word32) * (p_x++) * *(p_s_excf);
-            s2 += ((Word32)(*(p_s_excf)) * (*(p_s_excf)));
+            __builtin_mul_overflow(*(p_x++), *p_s_excf, &mul);
+            __builtin_add_overflow(s, mul, &s);
+            __builtin_mul_overflow(*p_s_excf, *p_s_excf, &mul);
+            __builtin_add_overflow(s2, mul, &s2);
             p_s_excf++;
         }
 
         s2     = s2 << 1;
         s2     = Inv_sqrt(s2, pOverflow);
         norm_h = (Word16)(s2 >> 16);
-        norm_l = (Word16)((s2 >> 1) - (norm_h << 15));
+        __builtin_sub_overflow((s2 >> 1), (norm_h << 15), &norm_l);
         corr_h = (Word16)(s >> 15);
-        corr_l = (Word16)((s) - (corr_h << 15));
+        __builtin_sub_overflow(s, (corr_h << 15), &corr_l);
 
         /* Normalize correlation = correlation * (1/sqrt(energy)) */
 
diff --git a/media/module/codecs/amrnb/enc/src/pitch_ol.cpp b/media/module/codecs/amrnb/enc/src/pitch_ol.cpp
index c039bb0..0e4b74b 100644
--- a/media/module/codecs/amrnb/enc/src/pitch_ol.cpp
+++ b/media/module/codecs/amrnb/enc/src/pitch_ol.cpp
@@ -959,6 +959,7 @@
     Word16 p_max3;
     Word16 scal_flag = 0;
     Word32 t0;
+    Word32 mul;
 
 #ifdef VAD2
     Word32 r01;
@@ -1002,7 +1003,8 @@
 
     for (i = -pit_max; i < L_frame; i++)
     {
-        t0 += (((Word32) * (p_signal)) * *(p_signal)) << 1;
+        __builtin_mul_overflow(*p_signal, *p_signal, &mul);
+        __builtin_add_overflow(t0, mul << 1, &t0);
         p_signal++;
         if (t0 < 0)
         {
diff --git a/media/module/codecs/amrnb/enc/src/pre_proc.cpp b/media/module/codecs/amrnb/enc/src/pre_proc.cpp
index 042920e..0e2be41 100644
--- a/media/module/codecs/amrnb/enc/src/pre_proc.cpp
+++ b/media/module/codecs/amrnb/enc/src/pre_proc.cpp
@@ -576,7 +576,7 @@
         *(p_signal++) = (Word16)((L_tmp + 0x0000800L) >> 12);
 
         st->y1_hi = (Word16)(L_tmp >> 12);
-        st->y1_lo = (Word16)((L_tmp << 3) - ((Word32)(st->y1_hi) << 15));
+        __builtin_sub_overflow((Word16)(L_tmp << 3), (st->y1_hi) << 15, &st->y1_lo);
 
     }
 
diff --git a/media/module/codecs/amrnb/enc/src/s10_8pf.cpp b/media/module/codecs/amrnb/enc/src/s10_8pf.cpp
index 352b611..97d0318 100644
--- a/media/module/codecs/amrnb/enc/src/s10_8pf.cpp
+++ b/media/module/codecs/amrnb/enc/src/s10_8pf.cpp
@@ -746,11 +746,13 @@
 
             for (i5 = ipos[5]; i5 < L_CODE; i5 += step)
             {
-                ps2 = ps1 + *(p_temp1++);
+                __builtin_add_overflow(ps1, *(p_temp1++), &ps2);
 
-                alp2 = alp1 + ((Word32) * (p_temp2 + i5) << 12);
+                __builtin_add_overflow(alp1, *(p_temp2 + i5) << 12, &alp2);
 
-                alp_16 = (Word16)((alp2 + ((Word32) * (p_temp1++) << 14)) >> 16);
+                Word32 result;
+                __builtin_add_overflow(alp2, *(p_temp1++) << 14, &result);
+                alp_16 = (Word16)(result >> 16);
                 sq2 = (Word16)(((Word32) ps2 * ps2) >> 15);
 
                 if (((Word32) sq2 * alp) > ((Word32) sq * alp_16))
diff --git a/media/module/codecs/amrnb/enc/src/set_sign.cpp b/media/module/codecs/amrnb/enc/src/set_sign.cpp
index fa43f78..55658a4 100644
--- a/media/module/codecs/amrnb/enc/src/set_sign.cpp
+++ b/media/module/codecs/amrnb/enc/src/set_sign.cpp
@@ -505,6 +505,7 @@
     Word16 en[L_CODE];                  /* correlation vector */
     Word32 s;
     Word32 t;
+    Word32 mul;
     Word32 L_temp;
     Word16 *p_cn;
     Word16 *p_dn;
@@ -525,7 +526,8 @@
         val = *(p_cn++);
         s = L_mac(s, val, val, pOverflow);
         val = *(p_dn++);
-        t += ((Word32) val * val) << 1;
+        __builtin_mul_overflow(val, val, &mul);
+        __builtin_add_overflow(t, mul << 1, &t);
     }
     s = Inv_sqrt(s, pOverflow);
     k_cn = (Word16)((L_shl(s, 5, pOverflow)) >> 16);
diff --git a/media/module/codecs/amrnb/enc/src/spstproc.cpp b/media/module/codecs/amrnb/enc/src/spstproc.cpp
index b9574aa..5210a39 100644
--- a/media/module/codecs/amrnb/enc/src/spstproc.cpp
+++ b/media/module/codecs/amrnb/enc/src/spstproc.cpp
@@ -192,6 +192,7 @@
     Word16 i;
     Word16 j;
     Word16 temp;
+    Word32 mul;
     Word32 L_temp;
     Word32 L_temp2;
     Word16 tempShift;
@@ -262,8 +263,10 @@
          */
         L_temp     = ((Word32) * (p_exc++) * pitch_fac) << 1;
         L_temp2    = ((Word32) * (p_exc--) * pitch_fac) << 1;
-        L_temp    += ((Word32) * (p_code++) * gain_code) << 1;
-        L_temp2   += ((Word32) * (p_code++) * gain_code) << 1;
+        __builtin_mul_overflow(*(p_code++), gain_code, &mul);
+        __builtin_add_overflow(L_temp, mul << 1, &L_temp);
+        __builtin_mul_overflow(*(p_code++), gain_code, &mul);
+        __builtin_add_overflow(L_temp2, mul << 1, &L_temp2);
         L_temp   <<=  tempShift;
         L_temp2  <<=  tempShift;
         *(p_exc++) = (Word16)((L_temp  + 0x08000L) >> 16);
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/Android.bp b/media/module/codecs/amrwb/enc/Android.bp
index 04f36b5..6ca3b6e 100644
--- a/media/module/codecs/amrwb/enc/Android.bp
+++ b/media/module/codecs/amrwb/enc/Android.bp
@@ -96,8 +96,6 @@
                 "-DARM",
                 "-DARMV7",
                 "-DASM_OPT",
-                // don't actually generate neon instructions, see bug 26932980
-                "-mfpu=vfpv3",
             ],
             local_include_dirs: [
                 "src/asm/ARMV7",
diff --git a/media/module/codecs/amrwb/enc/src/az_isp.c b/media/module/codecs/amrwb/enc/src/az_isp.c
index d7074f0..22a5c25 100644
--- a/media/module/codecs/amrwb/enc/src/az_isp.c
+++ b/media/module/codecs/amrwb/enc/src/az_isp.c
@@ -248,10 +248,10 @@
         b1_h = b0_h;
     }
 
-    t0 = ((b1_h * x)<<1) + (((b1_l * x)>>15)<<1);
-    t0 += (b2_h * (-32768))<<1;             /* t0 = x*b1 - b2          */
-    t0 -= (b2_l << 1);
-    t0 += (f[n] << 12);                     /* t0 = x*b1 - b2 + f[i]/2 */
+    __builtin_add_overflow(((b1_h * x)<<1), (((b1_l * x)>>15)<<1), &t0);
+    __builtin_add_overflow(t0, (b2_h * (-32768))<<1, &t0);   /* t0 = x*b1 - b2          */
+    __builtin_sub_overflow(t0, (b2_l << 1), &t0);
+    __builtin_add_overflow(t0, (f[n] << 12), &t0);     /* t0 = x*b1 - b2 + f[i]/2 */
 
     t0 = L_shl2(t0, 6);                     /* Q24 to Q30 with saturation */
 
diff --git a/media/module/codecs/amrwb/enc/src/syn_filt.c b/media/module/codecs/amrwb/enc/src/syn_filt.c
index 7eba12f..40398f5 100644
--- a/media/module/codecs/amrwb/enc/src/syn_filt.c
+++ b/media/module/codecs/amrwb/enc/src/syn_filt.c
@@ -109,38 +109,38 @@
         p2 = &sig_lo[i - 1];
         p3 = &sig_hi[i - 1];
 
-        L_tmp  -= vo_mult32((*p2--), (*p1));
-        L_tmp1 -= vo_mult32((*p3--), (*p1++));
-        L_tmp  -= vo_mult32((*p2--), (*p1));
-        L_tmp1 -= vo_mult32((*p3--), (*p1++));
-        L_tmp  -= vo_mult32((*p2--), (*p1));
-        L_tmp1 -= vo_mult32((*p3--), (*p1++));
-        L_tmp  -= vo_mult32((*p2--), (*p1));
-        L_tmp1 -= vo_mult32((*p3--), (*p1++));
-        L_tmp  -= vo_mult32((*p2--), (*p1));
-        L_tmp1 -= vo_mult32((*p3--), (*p1++));
-        L_tmp  -= vo_mult32((*p2--), (*p1));
-        L_tmp1 -= vo_mult32((*p3--), (*p1++));
-        L_tmp  -= vo_mult32((*p2--), (*p1));
-        L_tmp1 -= vo_mult32((*p3--), (*p1++));
-        L_tmp  -= vo_mult32((*p2--), (*p1));
-        L_tmp1 -= vo_mult32((*p3--), (*p1++));
-        L_tmp  -= vo_mult32((*p2--), (*p1));
-        L_tmp1 -= vo_mult32((*p3--), (*p1++));
-        L_tmp  -= vo_mult32((*p2--), (*p1));
-        L_tmp1 -= vo_mult32((*p3--), (*p1++));
-        L_tmp  -= vo_mult32((*p2--), (*p1));
-        L_tmp1 -= vo_mult32((*p3--), (*p1++));
-        L_tmp  -= vo_mult32((*p2--), (*p1));
-        L_tmp1 -= vo_mult32((*p3--), (*p1++));
-        L_tmp  -= vo_mult32((*p2--), (*p1));
-        L_tmp1 -= vo_mult32((*p3--), (*p1++));
-        L_tmp  -= vo_mult32((*p2--), (*p1));
-        L_tmp1 -= vo_mult32((*p3--), (*p1++));
-        L_tmp  -= vo_mult32((*p2--), (*p1));
-        L_tmp1 -= vo_mult32((*p3--), (*p1++));
-        L_tmp  -= vo_mult32((*p2--), (*p1));
-        L_tmp1 -= vo_mult32((*p3--), (*p1++));
+        __builtin_sub_overflow(L_tmp, vo_mult32((*p2--), (*p1)), &L_tmp);
+        __builtin_sub_overflow(L_tmp1, vo_mult32((*p3--), (*p1++)), &L_tmp1);
+        __builtin_sub_overflow(L_tmp, vo_mult32((*p2--), (*p1)), &L_tmp);
+        __builtin_sub_overflow(L_tmp1, vo_mult32((*p3--), (*p1++)), &L_tmp1);
+        __builtin_sub_overflow(L_tmp, vo_mult32((*p2--), (*p1)), &L_tmp);
+        __builtin_sub_overflow(L_tmp1, vo_mult32((*p3--), (*p1++)), &L_tmp1);
+        __builtin_sub_overflow(L_tmp, vo_mult32((*p2--), (*p1)), &L_tmp);
+        __builtin_sub_overflow(L_tmp1, vo_mult32((*p3--), (*p1++)), &L_tmp1);
+        __builtin_sub_overflow(L_tmp, vo_mult32((*p2--), (*p1)), &L_tmp);
+        __builtin_sub_overflow(L_tmp1, vo_mult32((*p3--), (*p1++)), &L_tmp1);
+        __builtin_sub_overflow(L_tmp, vo_mult32((*p2--), (*p1)), &L_tmp);
+        __builtin_sub_overflow(L_tmp1, vo_mult32((*p3--), (*p1++)), &L_tmp1);
+        __builtin_sub_overflow(L_tmp, vo_mult32((*p2--), (*p1)), &L_tmp);
+        __builtin_sub_overflow(L_tmp1, vo_mult32((*p3--), (*p1++)), &L_tmp1);
+        __builtin_sub_overflow(L_tmp, vo_mult32((*p2--), (*p1)), &L_tmp);
+        __builtin_sub_overflow(L_tmp1, vo_mult32((*p3--), (*p1++)), &L_tmp1);
+        __builtin_sub_overflow(L_tmp, vo_mult32((*p2--), (*p1)), &L_tmp);
+        __builtin_sub_overflow(L_tmp1, vo_mult32((*p3--), (*p1++)), &L_tmp1);
+        __builtin_sub_overflow(L_tmp, vo_mult32((*p2--), (*p1)), &L_tmp);
+        __builtin_sub_overflow(L_tmp1, vo_mult32((*p3--), (*p1++)), &L_tmp1);
+        __builtin_sub_overflow(L_tmp, vo_mult32((*p2--), (*p1)), &L_tmp);
+        __builtin_sub_overflow(L_tmp1, vo_mult32((*p3--), (*p1++)), &L_tmp1);
+        __builtin_sub_overflow(L_tmp, vo_mult32((*p2--), (*p1)), &L_tmp);
+        __builtin_sub_overflow(L_tmp1, vo_mult32((*p3--), (*p1++)), &L_tmp1);
+        __builtin_sub_overflow(L_tmp, vo_mult32((*p2--), (*p1)), &L_tmp);
+        __builtin_sub_overflow(L_tmp1, vo_mult32((*p3--), (*p1++)), &L_tmp1);
+        __builtin_sub_overflow(L_tmp, vo_mult32((*p2--), (*p1)), &L_tmp);
+        __builtin_sub_overflow(L_tmp1, vo_mult32((*p3--), (*p1++)), &L_tmp1);
+        __builtin_sub_overflow(L_tmp, vo_mult32((*p2--), (*p1)), &L_tmp);
+        __builtin_sub_overflow(L_tmp1, vo_mult32((*p3--), (*p1++)), &L_tmp1);
+        __builtin_sub_overflow(L_tmp, vo_mult32((*p2--), (*p1)), &L_tmp);
+        __builtin_sub_overflow(L_tmp1, vo_mult32((*p3--), (*p1++)), &L_tmp1);
 
         L_tmp = L_tmp >> 11;
         L_tmp += vo_L_mult(exc[i], a0);
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/module/extractors/Android.bp b/media/module/extractors/Android.bp
index f654ecd..e29d3e6 100644
--- a/media/module/extractors/Android.bp
+++ b/media/module/extractors/Android.bp
@@ -28,6 +28,10 @@
         "liblog",
     ],
 
+    static_libs: [
+        "libstagefright_metadatautils",
+    ],
+
     // extractors are expected to run on Q(29)
     min_sdk_version: "29",
     apex_available: [
@@ -56,6 +60,7 @@
                 "libutils",
                 "libmediandk_format",
                 "libmedia_ndkformatpriv",
+                "libstagefright_metadatautils",
             ],
         },
     },
@@ -68,3 +73,21 @@
         ],
     },
 }
+
+aconfig_declarations {
+    name: "android.media.extractor.flags-aconfig",
+    package: "com.android.media.extractor.flags",
+    container: "com.android.media",
+    srcs: ["extractor.aconfig"],
+}
+
+cc_aconfig_library {
+    name: "android.media.extractor.flags-aconfig-cc",
+    aconfig_declarations: "android.media.extractor.flags-aconfig",
+    host_supported: true,
+    min_sdk_version: "29",
+    apex_available: [
+        "//apex_available:platform",
+        "com.android.media",
+    ],
+}
diff --git a/media/module/extractors/extractor.aconfig b/media/module/extractors/extractor.aconfig
new file mode 100644
index 0000000..c9bf694
--- /dev/null
+++ b/media/module/extractors/extractor.aconfig
@@ -0,0 +1,14 @@
+# Media Extractor flags.
+#
+# !!! Please add flags in alphabetical order. !!!
+package: "com.android.media.extractor.flags"
+container: "com.android.media"
+
+flag {
+    name: "extractor_sniff_midi_optimizations"
+    is_exported: true
+    is_fixed_read_only: true
+    namespace: "media_extractor"
+    description: "Enable SniffMidi optimizations."
+    bug: "359920208"
+}
diff --git a/media/module/extractors/fuzzers/Android.bp b/media/module/extractors/fuzzers/Android.bp
index 7a49d8e..3da1589 100644
--- a/media/module/extractors/fuzzers/Android.bp
+++ b/media/module/extractors/fuzzers/Android.bp
@@ -24,6 +24,7 @@
     // to get the below license kinds:
     //   SPDX-license-identifier-Apache-2.0
     default_applicable_licenses: ["frameworks_av_license"],
+    default_team: "trendy_team_android_media_solutions_playback",
 }
 
 cc_defaults {
@@ -131,6 +132,7 @@
         "libstagefright_id3",
         "libstagefright_esds",
         "libmp4extractor",
+        "libstagefright_metadatautils",
     ],
 
     dictionary: "mp4_extractor_fuzzer.dict",
@@ -301,12 +303,18 @@
     ],
 
     static_libs: [
+        "android.media.extractor.flags-aconfig-cc",
+        "libaconfig_storage_read_api_cc",
         "libsonivox",
         "libmedia_midiiowrapper",
         "libmidiextractor",
         "libwatchdog",
     ],
 
+    shared_libs: [
+        "server_configurable_flags",
+    ],
+
     dictionary: "midi_extractor_fuzzer.dict",
 
     host_supported: true,
diff --git a/media/module/extractors/midi/Android.bp b/media/module/extractors/midi/Android.bp
index feabf9e..0eb34fc 100644
--- a/media/module/extractors/midi/Android.bp
+++ b/media/module/extractors/midi/Android.bp
@@ -32,6 +32,8 @@
     ],
 
     static_libs: [
+        "android.media.extractor.flags-aconfig-cc",
+        "libaconfig_storage_read_api_cc",
         "libmedia_midiiowrapper",
         "libsonivoxwithoutjet",
         "libstagefright_foundation",
@@ -40,6 +42,7 @@
 
     shared_libs: [
         "libbase",
+        "server_configurable_flags",
     ],
 
     host_supported: true,
diff --git a/media/module/extractors/midi/MidiExtractor.cpp b/media/module/extractors/midi/MidiExtractor.cpp
index 167cc40..98d7716 100644
--- a/media/module/extractors/midi/MidiExtractor.cpp
+++ b/media/module/extractors/midi/MidiExtractor.cpp
@@ -20,6 +20,7 @@
 
 #include "MidiExtractor.h"
 
+#include <com_android_media_extractor_flags.h>
 #include <media/MidiIoWrapper.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/MediaBufferGroup.h>
@@ -323,10 +324,97 @@
     return AMediaFormat_copy(meta, mFileMetadata);
 }
 
-// Sniffer
+static bool startsWith(const uint8_t *buf, size_t size, const char *pattern, size_t patternSize) {
+    if (size < patternSize) {
+        return false;
+    }
+    return (memcmp(buf, pattern, patternSize) == 0);
+}
 
-bool SniffMidi(CDataSource *source, float *confidence)
-{
+static bool isValidMThd(const uint8_t *buf, size_t size) {
+    return startsWith(buf, size, "MThd", 4);
+}
+
+static bool isValidXmf(const uint8_t *buf, size_t size) {
+    return startsWith(buf, size, "XMF_", 4);
+}
+
+static bool isValidImelody(const uint8_t *buf, size_t size) {
+    return startsWith(buf, size, "BEGIN:IMELODY", 13);
+}
+
+static bool isValidRtttl(const uint8_t *buf, size_t size) {
+    #define RTTTL_MAX_TITLE_LEN 32
+    // rtttl starts with the following:
+    // <title>:<type>=<value>
+    //
+    // Where:
+    // - <title>: Up to 32 characters
+    // - <type>: Single character indicating:
+    //     'd' for duration
+    //     'o' for octave
+    //     'b' for beats per minute
+    // - <value>: Corresponding value for the type
+    if (size < 4) {
+        return false;
+    }
+    for (size_t i = 0; i < RTTTL_MAX_TITLE_LEN && i < size; i++) {
+        if (buf[i] == ':') {
+            if (i < (size - 3) && buf[i + 2] == '=') {
+                return true;
+            }
+            break;
+        }
+    }
+    return false;
+}
+
+static bool isValidOta(const uint8_t *buf, size_t size) {
+    #define OTA_RINGTONE 0x25
+    #define OTA_SOUND 0x1d
+    #define OTA_UNICODE 0x22
+
+    // ota starts with the following:
+    // <cmdLen><cmd1><cmd2>..<cmdN>
+    //
+    // Where:
+    // - <cmdLen>: Single character command length
+    // - <cmd1>: Single character (OTA_RINGTONE << 1)
+    // - <cmd2>: Single character (OTA_SOUND << 1) or (OTA_UNICODE << 1)
+    //           and so on with last cmd being (0x1d << 1)
+
+    if (size < 3) {
+        return false;
+    }
+
+    uint8_t cmdLen = buf[0];
+    if (cmdLen < 2) {
+        return false;
+    }
+
+    if ((buf[1] >> 1) != OTA_RINGTONE) {
+        return false;
+    }
+    cmdLen--;
+
+    size_t i = 2;
+    while(cmdLen && i < size) {
+        switch(buf[i] >> 1) {
+            case OTA_SOUND:
+                return true;
+            case OTA_UNICODE:
+                break;
+            default:
+                return false;
+        }
+        cmdLen--;
+        i++;
+    }
+
+    return false;
+}
+
+bool SniffMidiLegacy(CDataSource *source, float *confidence) {
     MidiEngine p(source, NULL, NULL);
     if (p.initCheck() == OK) {
         *confidence = 0.8;
@@ -335,7 +423,47 @@
     }
     ALOGV("SniffMidi: no");
     return false;
+}
 
+bool SniffMidiEfficiently(CDataSource *source, float *confidence) {
+    uint8_t header[128];
+    int filled = source->readAt(source->handle, 0, header, sizeof(header));
+
+    if (isValidMThd(header, filled)) {
+        *confidence = 0.80;
+        ALOGV("SniffMidi: yes, MThd");
+        return true;
+    }
+    if (isValidXmf(header, filled)) {
+        *confidence = 0.80;
+        ALOGV("SniffMidi: yes, XMF_");
+        return true;
+    }
+    if (isValidImelody(header, filled)) {
+        *confidence = 0.80;
+        ALOGV("SniffMidi: yes, imelody");
+        return true;
+    }
+    if (isValidRtttl(header, filled)) {
+        *confidence = 0.80;
+        ALOGV("SniffMidi: yes, rtttl");
+        return true;
+    }
+    if (isValidOta(header, filled)) {
+        *confidence = 0.80;
+        ALOGV("SniffMidi: yes, ota");
+        return true;
+    }
+    ALOGV("SniffMidi: no");
+    return false;
+}
+
+// Sniffer
+bool SniffMidi(CDataSource *source, float *confidence) {
+    if(com::android::media::extractor::flags::extractor_sniff_midi_optimizations()) {
+        return SniffMidiEfficiently(source, confidence);
+    }
+    return SniffMidiLegacy(source, confidence);
 }
 
 static const char *extensions[] = {
diff --git a/media/module/extractors/mkv/MatroskaExtractor.cpp b/media/module/extractors/mkv/MatroskaExtractor.cpp
index f326db1..10ae07a 100644
--- a/media/module/extractors/mkv/MatroskaExtractor.cpp
+++ b/media/module/extractors/mkv/MatroskaExtractor.cpp
@@ -1787,7 +1787,7 @@
         return ERROR_MALFORMED;
     }
 
-    if (!MakeVP9CodecSpecificData(trackInfo->mMeta, tmpData.get(), frame.len)) {
+    if (!MakeVP9CodecSpecificDataFromFirstFrame(trackInfo->mMeta, tmpData.get(), frame.len)) {
         return ERROR_MALFORMED;
     }
 
diff --git a/media/module/extractors/mp4/MPEG4Extractor.cpp b/media/module/extractors/mp4/MPEG4Extractor.cpp
index f247f8c..12c0aaf 100644
--- a/media/module/extractors/mp4/MPEG4Extractor.cpp
+++ b/media/module/extractors/mp4/MPEG4Extractor.cpp
@@ -51,6 +51,7 @@
 #include <media/stagefright/MediaBufferGroup.h>
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/MetaDataBase.h>
+#include <media/stagefright/MetaDataUtils.h>
 #include <utils/String8.h>
 
 #include <byteswap.h>
@@ -2596,8 +2597,32 @@
             *offset += chunk_size;
             break;
         }
-
         case FOURCC("vpcC"):
+        {
+            if (mLastTrack == NULL) {
+                return ERROR_MALFORMED;
+            }
+
+            auto buffer = heapbuffer<uint8_t>(chunk_data_size);
+
+            if (buffer.get() == NULL) {
+                ALOGE("b/28471206");
+                return NO_MEMORY;
+            }
+
+            if (mDataSource->readAt(data_offset, buffer.get(), chunk_data_size) < chunk_data_size) {
+                return ERROR_IO;
+            }
+
+            if (!MakeVP9CodecPrivateFromVpcC(mLastTrack->meta, buffer.get(), chunk_data_size)) {
+                ALOGE("Failed to create VP9 CodecPrivate from vpcC.");
+                return ERROR_MALFORMED;
+            }
+
+            *offset += chunk_size;
+            break;
+        }
+
         case FOURCC("av1C"):
         {
             auto buffer = heapbuffer<uint8_t>(chunk_data_size);
diff --git a/media/module/metadatautils/MetaDataUtils.cpp b/media/module/metadatautils/MetaDataUtils.cpp
index 0895bb5..177438a 100644
--- a/media/module/metadatautils/MetaDataUtils.cpp
+++ b/media/module/metadatautils/MetaDataUtils.cpp
@@ -134,10 +134,54 @@
     }
     return true;
 }
+
+/**
+ * Build VP9 Codec Feature Metadata (CodecPrivate) to set CSD for VP9 codec.
+ * For reference:
+ * https://www.webmproject.org/docs/container/#vp9-codec-feature-metadata-codecprivate.
+ *
+ * @param meta          A pointer to AMediaFormat object.
+ * @param profile       The profile value of the VP9 stream.
+ * @param level         The VP9 codec level. If the level is unknown, pass -1 to this parameter.
+ * @param bitDepth      The bit depth of the luma and color components of the VP9 stream.
+ * @param chromaSubsampling  The chroma subsampling of the VP9 stream. If chromaSubsampling is
+ *                           unknown, pass -1 to this parameter.
+ * @return true if CodecPrivate is set as CSD of AMediaFormat object.
+ *
+ */
+static bool MakeVP9CodecPrivate(AMediaFormat* meta, int32_t profile, int32_t level,
+                                int32_t bitDepth, int32_t chromaSubsampling) {
+    if (meta == nullptr) {
+        return false;
+    }
+
+    std::vector<uint8_t> codecPrivate;
+    // Construct CodecPrivate in WebM format (ID | Length | Data).
+    // Helper lambda to add a field to the codec private data
+    auto addField = [&codecPrivate](uint8_t id, uint8_t value) {
+        codecPrivate.push_back(id);
+        codecPrivate.push_back(0x01);  // Length is always 1
+        codecPrivate.push_back(value);
+    };
+
+    // Add fields
+    addField(0x01, static_cast<uint8_t>(profile));
+    if (level >= 0) {
+        addField(0x02, static_cast<uint8_t>(level));
+    }
+    addField(0x03, static_cast<uint8_t>(bitDepth));
+    if (chromaSubsampling >= 0) {
+        addField(0x04, static_cast<uint8_t>(chromaSubsampling));
+    }
+    // Set CSD in the meta format
+    AMediaFormat_setBuffer(meta, AMEDIAFORMAT_KEY_CSD_0, codecPrivate.data(), codecPrivate.size());
+    return true;
+}
+
 // The param data contains the first frame data, starting with the uncompressed frame
 // header. This uncompressed header (refer section 6.2 of the VP9 bitstream spec) is
 // used to parse profile, bitdepth and subsampling.
-bool MakeVP9CodecSpecificData(AMediaFormat* meta, const uint8_t* data, size_t size) {
+bool MakeVP9CodecSpecificDataFromFirstFrame(AMediaFormat* meta, const uint8_t* data, size_t size) {
     if (meta == nullptr || data == nullptr || size == 0) {
         return false;
     }
@@ -227,29 +271,29 @@
     if (chromaSubsampling != -1) {
         csdSize += 3;
     }
+    // As level is not present in first frame build CodecPrivate without it.
+    return MakeVP9CodecPrivate(meta, profile, -1, bitDepth, chromaSubsampling);
+}
 
-    // Create VP9 Codec Feature Metadata (CodecPrivate) that can be parsed
-    // https://www.webmproject.org/docs/container/#vp9-codec-feature-metadata-codecprivate
-    sp<ABuffer> csd = sp<ABuffer>::make(csdSize);
-    uint8_t* csdData = csd->data();
-
-    *csdData++ = 0x01 /* FEATURE PROFILE */;
-    *csdData++ = 0x01 /* length */;
-    *csdData++ = profile;
-
-    *csdData++ = 0x03 /* FEATURE BITDEPTH */;
-    *csdData++ = 0x01 /* length */;
-    *csdData++ = bitDepth;
-
-    // csdSize more than 6 means chroma subsampling data was found.
-    if (csdSize > 6) {
-        *csdData++ = 0x04 /* FEATURE SUBSAMPLING */;
-        *csdData++ = 0x01 /* length */;
-        *csdData++ = chromaSubsampling;
+bool MakeVP9CodecPrivateFromVpcC(AMediaFormat* meta, const uint8_t* csdData, size_t size) {
+    if (meta == nullptr || csdData == nullptr || size < 12) {
+        return false;
     }
 
-    AMediaFormat_setBuffer(meta, AMEDIAFORMAT_KEY_CSD_0, csd->data(), csd->size());
-    return true;
+    // Check the first 4 bytes (VersionAndFlags) if they match the required value.
+    if (csdData[0] != 0x01 || csdData[1] != 0x00 || csdData[2] != 0x00 || csdData[3] != 0x00) {
+        return false;
+    }
+
+    // Create VP9 Codec Feature Metadata (CodecPrivate) that can be parsed.
+    // https://www.webmproject.org/docs/container/#vp9-codec-feature-metadata-codecprivate
+    const uint8_t* vpcCData = csdData + 4;  // Skip the first 4 bytes (VersionAndFlags)
+
+    int32_t profile = vpcCData[0];
+    int32_t level = vpcCData[1];
+    int32_t bitDepth = (vpcCData[2] >> 4) & 0x0F;           // Bit Depth (4 bits).
+    int32_t chromaSubsampling = (vpcCData[2] >> 1) & 0x07;  // Chroma Subsampling (3 bits).
+    return MakeVP9CodecPrivate(meta, profile, level, bitDepth, chromaSubsampling);
 }
 
 bool MakeAACCodecSpecificData(MetaDataBase &meta, const uint8_t *data, size_t size) {
diff --git a/media/module/metadatautils/include/media/stagefright/MetaDataUtils.h b/media/module/metadatautils/include/media/stagefright/MetaDataUtils.h
index 69cf21a..9988544 100644
--- a/media/module/metadatautils/include/media/stagefright/MetaDataUtils.h
+++ b/media/module/metadatautils/include/media/stagefright/MetaDataUtils.h
@@ -38,7 +38,10 @@
 void parseVorbisComment(
         AMediaFormat *fileMeta, const char *comment, size_t commentLength);
 
-bool MakeVP9CodecSpecificData(AMediaFormat* meta, const uint8_t* data, size_t size);
+bool MakeVP9CodecSpecificData(AMediaFormat* meta, int32_t csdSize, int32_t profile, int32_t level,
+                              int32_t bitDepth, int32_t chromaSubsampling);
+bool MakeVP9CodecSpecificDataFromFirstFrame(AMediaFormat* meta, const uint8_t* data, size_t size);
+bool MakeVP9CodecPrivateFromVpcC(AMediaFormat* meta, const uint8_t* data, size_t size);
 
 }  // namespace android
 
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..8215247 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -2923,7 +2923,7 @@
                                                         audio_config_base_t *mixerConfig,
                                                         audio_devices_t deviceType,
                                                         const String8& address,
-                                                        audio_output_flags_t flags,
+                                                        audio_output_flags_t *flags,
                                                         const audio_attributes_t attributes)
 {
     AudioHwDevice *outHwDev = findSuitableHwDev_l(module, deviceType);
@@ -2958,7 +2958,7 @@
     mHardwareStatus = AUDIO_HW_IDLE;
 
     if (status == NO_ERROR) {
-        if (flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) {
+        if (*flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) {
             const sp<IAfMmapPlaybackThread> thread = IAfMmapPlaybackThread::create(
                     this, *output, outHwDev, outputStream, mSystemReady);
             mMmapThreads.add(*output, thread);
@@ -2967,22 +2967,22 @@
             return thread;
         } else {
             sp<IAfPlaybackThread> thread;
-            if (flags & AUDIO_OUTPUT_FLAG_BIT_PERFECT) {
+            if (*flags & AUDIO_OUTPUT_FLAG_BIT_PERFECT) {
                 thread = IAfPlaybackThread::createBitPerfectThread(
                         this, outputStream, *output, mSystemReady);
                 ALOGV("%s() created bit-perfect output: ID %d thread %p",
                       __func__, *output, thread.get());
-            } else if (flags & AUDIO_OUTPUT_FLAG_SPATIALIZER) {
+            } else if (*flags & AUDIO_OUTPUT_FLAG_SPATIALIZER) {
                 thread = IAfPlaybackThread::createSpatializerThread(this, outputStream, *output,
                                                     mSystemReady, mixerConfig);
                 ALOGV("openOutput_l() created spatializer output: ID %d thread %p",
                       *output, thread.get());
-            } else if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+            } else if (*flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
                 thread = IAfPlaybackThread::createOffloadThread(this, outputStream, *output,
                         mSystemReady, halConfig->offload_info);
                 ALOGV("openOutput_l() created offload output: ID %d thread %p",
                       *output, thread.get());
-            } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
+            } else if ((*flags & AUDIO_OUTPUT_FLAG_DIRECT)
                     || !IAfThreadBase::isValidPcmSinkFormat(halConfig->format)
                     || !IAfThreadBase::isValidPcmSinkChannelMask(halConfig->channel_mask)) {
                 thread = IAfPlaybackThread::createDirectOutputThread(this, outputStream, *output,
@@ -3046,7 +3046,7 @@
     audio_utils::lock_guard _l(mutex());
 
     const sp<IAfThreadBase> thread = openOutput_l(module, &output, &halConfig,
-            &mixerConfig, deviceType, address, flags, attributes);
+            &mixerConfig, deviceType, address, &flags, attributes);
     if (thread != 0) {
         uint32_t latencyMs = 0;
         if ((flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0) {
@@ -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/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 21c171d..6777075 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -337,7 +337,7 @@
             audio_config_base_t* mixerConfig,
             audio_devices_t deviceType,
             const String8& address,
-            audio_output_flags_t flags,
+            audio_output_flags_t* flags,
             audio_attributes_t attributes) final REQUIRES(mutex());
     const DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*>&
             getAudioHwDevs_l() const final REQUIRES(mutex(), hardwareMutex()) {
diff --git a/services/audioflinger/IAfPatchPanel.h b/services/audioflinger/IAfPatchPanel.h
index 37dce3a..15b6ddf 100644
--- a/services/audioflinger/IAfPatchPanel.h
+++ b/services/audioflinger/IAfPatchPanel.h
@@ -82,7 +82,7 @@
             audio_config_base_t* mixerConfig,
             audio_devices_t deviceType,
             const String8& address,
-            audio_output_flags_t flags,
+            audio_output_flags_t* flags,
             audio_attributes_t attributes) REQUIRES(mutex()) = 0;
     virtual audio_utils::mutex& mutex() const
             RETURN_CAPABILITY(audio_utils::AudioFlinger_Mutex) = 0;
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index 35f17c1..994dd47 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -268,7 +268,7 @@
                                                             &mixerConfig,
                                                             outputDevice,
                                                             outputDeviceAddress,
-                                                            flags,
+                                                            &flags,
                                                             attributes);
                     ALOGV("mAfPatchPanelCallback->openOutput_l() returned %p", thread.get());
                     if (thread == 0) {
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index ecbd0ae..147a5d6 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -3220,9 +3220,9 @@
 
     // Calculate size of normal sink buffer relative to the HAL output buffer size
     double multiplier = 1.0;
-    // Note: mType == SPATIALIZER does not support FastMixer.
-    if (mType == MIXER && (kUseFastMixer == FastMixer_Static ||
-            kUseFastMixer == FastMixer_Dynamic)) {
+    // Note: mType == SPATIALIZER does not support FastMixer and DEEP is by definition not "fast"
+    if ((mType == MIXER && !(mOutput->flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER)) &&
+            (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) {
         size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000;
         size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000;
 
@@ -4017,7 +4017,13 @@
     // FIXME could this be made local to while loop?
     writeFrames = 0;
 
-    cacheParameters_l();
+    {
+        audio_utils::lock_guard l(mutex());
+
+        cacheParameters_l();
+        checkSilentMode_l();
+    }
+
     mSleepTimeUs = mIdleSleepTimeUs;
 
     if (mType == MIXER || mType == SPATIALIZER) {
@@ -4042,8 +4048,6 @@
     // suspended mode (for now) to help schedule the wait time until next iteration.
     nsecs_t timeLoopNextNs = 0;
 
-    checkSilentMode_l();
-
     audio_patch_handle_t lastDownstreamPatchHandle = AUDIO_PATCH_HANDLE_NONE;
 
     sendCheckOutputStageEffectsEvent();
@@ -5135,7 +5139,16 @@
             break;
         case FastMixer_Static:
         case FastMixer_Dynamic:
-            initFastMixer = mFrameCount < mNormalFrameCount;
+            if (mType == MIXER && (output->flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER)) {
+                /* Do not init fast mixer on deep buffer, warn if buffers are confed too small */
+                initFastMixer = false;
+                ALOGW_IF(mFrameCount * 1000 / mSampleRate < kMinNormalSinkBufferSizeMs,
+                         "HAL DEEP BUFFER Buffer (%zu ms) is smaller than set minimal buffer "
+                         "(%u ms), seems like a configuration error",
+                         mFrameCount * 1000 / mSampleRate, kMinNormalSinkBufferSizeMs);
+            } else {
+                initFastMixer = mFrameCount < mNormalFrameCount;
+            }
             break;
         }
         ALOGW_IF(initFastMixer == false && mFrameCount < mNormalFrameCount,
diff --git a/services/audioflinger/datapath/AudioHwDevice.cpp b/services/audioflinger/datapath/AudioHwDevice.cpp
index 5314e9e..c2e538c 100644
--- a/services/audioflinger/datapath/AudioHwDevice.cpp
+++ b/services/audioflinger/datapath/AudioHwDevice.cpp
@@ -41,19 +41,20 @@
         AudioStreamOut **ppStreamOut,
         audio_io_handle_t handle,
         audio_devices_t deviceType,
-        audio_output_flags_t flags,
+        audio_output_flags_t *flags,
         struct audio_config *config,
         const char *address,
         const std::vector<playback_track_metadata_v7_t>& sourceMetadata)
 {
 
     struct audio_config originalConfig = *config;
-    auto outputStream = new AudioStreamOut(this, flags);
+    auto outputStream = new AudioStreamOut(this);
 
     // Try to open the HAL first using the current format.
     ALOGV("openOutputStream(), try sampleRate %d, format %#x, channelMask %#x", config->sample_rate,
             config->format, config->channel_mask);
-    status_t status = outputStream->open(handle, deviceType, config, address, sourceMetadata);
+    status_t status = outputStream->open(handle, deviceType, config, flags, address,
+                                        sourceMetadata);
 
     if (status != NO_ERROR) {
         delete outputStream;
@@ -67,19 +68,25 @@
 
         // If the data is encoded then try again using wrapped PCM.
         const bool wrapperNeeded = !audio_has_proportional_frames(originalConfig.format)
-                && ((flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0)
-                && ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0);
+                && ((*flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0)
+                && ((*flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0);
 
         if (wrapperNeeded) {
             if (SPDIFEncoder::isFormatSupported(originalConfig.format)) {
-                outputStream = new SpdifStreamOut(this, flags, originalConfig.format);
-                status = outputStream->open(handle, deviceType, &originalConfig, address,
+                outputStream = new SpdifStreamOut(this, originalConfig.format);
+                status = outputStream->open(handle, deviceType, &originalConfig, flags, address,
                                             sourceMetadata);
                 if (status != NO_ERROR) {
                     ALOGE("ERROR - openOutputStream(), SPDIF open returned %d",
                         status);
                     delete outputStream;
                     outputStream = nullptr;
+                } else {
+                    // on success, we need to assign the actual HAL stream config so that clients
+                    // know and can later patch correctly.
+                    config->format = originalConfig.format;
+                    config->channel_mask = originalConfig.channel_mask;
+                    config->sample_rate = originalConfig.sample_rate;
                 }
             } else {
                 ALOGE("ERROR - openOutputStream(), SPDIFEncoder does not support format 0x%08x",
@@ -153,6 +160,12 @@
                         status);
                     delete inputStream;
                     inputStream = nullptr;
+                } else {
+                    // on success, we need to assign the actual HAL stream config so that clients
+                    // know and can later patch correctly.
+                    config->format = originalConfig.format;
+                    config->channel_mask = originalConfig.channel_mask;
+                    config->sample_rate = originalConfig.sample_rate;
                 }
             } else {
                 ALOGE("ERROR - openInputStream(), SPDIFDecoder does not support format 0x%08x",
diff --git a/services/audioflinger/datapath/AudioHwDevice.h b/services/audioflinger/datapath/AudioHwDevice.h
index e1a9018..6a35b91 100644
--- a/services/audioflinger/datapath/AudioHwDevice.h
+++ b/services/audioflinger/datapath/AudioHwDevice.h
@@ -85,7 +85,7 @@
             AudioStreamOut **ppStreamOut,
             audio_io_handle_t handle,
             audio_devices_t deviceType,
-            audio_output_flags_t flags,
+            audio_output_flags_t *flags,
             struct audio_config *config,
             const char *address,
             const std::vector<playback_track_metadata_v7_t>& sourceMetadata);
diff --git a/services/audioflinger/datapath/AudioStreamOut.cpp b/services/audioflinger/datapath/AudioStreamOut.cpp
index c65373e..7aadda3 100644
--- a/services/audioflinger/datapath/AudioStreamOut.cpp
+++ b/services/audioflinger/datapath/AudioStreamOut.cpp
@@ -30,9 +30,8 @@
 namespace android {
 
 // ----------------------------------------------------------------------------
-AudioStreamOut::AudioStreamOut(AudioHwDevice *dev, audio_output_flags_t flags)
+AudioStreamOut::AudioStreamOut(AudioHwDevice *dev)
         : audioHwDev(dev)
-        , flags(flags)
 {
 }
 
@@ -93,14 +92,16 @@
         audio_io_handle_t handle,
         audio_devices_t deviceType,
         struct audio_config *config,
+        audio_output_flags_t *flagsPtr,
         const char *address,
         const std::vector<playback_track_metadata_v7_t>& sourceMetadata)
 {
     sp<StreamOutHalInterface> outStream;
 
-    const audio_output_flags_t customFlags = (config->format == AUDIO_FORMAT_IEC61937)
-                ? (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO)
-                : flags;
+    audio_output_flags_t customFlags = (config->format == AUDIO_FORMAT_IEC61937)
+                ? (audio_output_flags_t)(*flagsPtr | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO)
+                : *flagsPtr;
+    *flagsPtr = flags = customFlags;
 
     int status = hwDev()->openOutputStream(
             handle,
diff --git a/services/audioflinger/datapath/AudioStreamOut.h b/services/audioflinger/datapath/AudioStreamOut.h
index 2bf94a1..1857099 100644
--- a/services/audioflinger/datapath/AudioStreamOut.h
+++ b/services/audioflinger/datapath/AudioStreamOut.h
@@ -37,16 +37,17 @@
 public:
     AudioHwDevice * const audioHwDev;
     sp<StreamOutHalInterface> stream;
-    const audio_output_flags_t flags;
+    audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
 
     [[nodiscard]] sp<DeviceHalInterface> hwDev() const;
 
-    AudioStreamOut(AudioHwDevice *dev, audio_output_flags_t flags);
+    explicit AudioStreamOut(AudioHwDevice *dev);
 
     virtual status_t open(
             audio_io_handle_t handle,
             audio_devices_t deviceType,
             struct audio_config *config,
+            audio_output_flags_t *flagsPtr,
             const char *address,
             const std::vector<playback_track_metadata_v7_t>& sourceMetadata);
 
diff --git a/services/audioflinger/datapath/SpdifStreamIn.cpp b/services/audioflinger/datapath/SpdifStreamIn.cpp
index 98ce712..0090bc5 100644
--- a/services/audioflinger/datapath/SpdifStreamIn.cpp
+++ b/services/audioflinger/datapath/SpdifStreamIn.cpp
@@ -81,6 +81,11 @@
             outputDevice,
             outputDeviceAddress);
 
+    // reset config back to whatever is returned by HAL
+    config->sample_rate = customConfig.sample_rate;
+    config->format = customConfig.format;
+    config->channel_mask = customConfig.channel_mask;
+
     ALOGI("SpdifStreamIn::open() status = %d", status);
 
 #ifdef TEE_SINK
diff --git a/services/audioflinger/datapath/SpdifStreamOut.cpp b/services/audioflinger/datapath/SpdifStreamOut.cpp
index d3983b0..a565955 100644
--- a/services/audioflinger/datapath/SpdifStreamOut.cpp
+++ b/services/audioflinger/datapath/SpdifStreamOut.cpp
@@ -33,10 +33,8 @@
  * PCM then we need to wrap the data in an SPDIF wrapper.
  */
 SpdifStreamOut::SpdifStreamOut(AudioHwDevice *dev,
-            audio_output_flags_t flags,
             audio_format_t format)
-        // Tell the HAL that the data will be compressed audio wrapped in a data burst.
-        : AudioStreamOut(dev, (audio_output_flags_t) (flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO))
+        : AudioStreamOut(dev)
         , mSpdifEncoder(this, format)
 {
 }
@@ -45,6 +43,7 @@
         audio_io_handle_t handle,
         audio_devices_t devices,
         struct audio_config *config,
+        audio_output_flags_t *flags,
         const char *address,
         const std::vector<playback_track_metadata_v7_t>& sourceMetadata)
 {
@@ -63,6 +62,8 @@
 
     customConfig.format = AUDIO_FORMAT_PCM_16_BIT;
     customConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+    // Tell the HAL that the data will be compressed audio wrapped in a data burst.
+    *flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO);
 
     // Always print this because otherwise it could be very confusing if the
     // HAL and AudioFlinger are using different formats.
@@ -76,9 +77,15 @@
             handle,
             devices,
             &customConfig,
+            flags,
             address,
             sourceMetadata);
 
+    // reset config back to whatever is returned by HAL
+    config->sample_rate = customConfig.sample_rate;
+    config->format = customConfig.format;
+    config->channel_mask = customConfig.channel_mask;
+
     ALOGI("SpdifStreamOut::open() status = %d", status);
 
 #ifdef TEE_SINK
diff --git a/services/audioflinger/datapath/SpdifStreamOut.h b/services/audioflinger/datapath/SpdifStreamOut.h
index 1cd8f65..3241d6f 100644
--- a/services/audioflinger/datapath/SpdifStreamOut.h
+++ b/services/audioflinger/datapath/SpdifStreamOut.h
@@ -36,13 +36,13 @@
 class SpdifStreamOut : public AudioStreamOut {
 public:
 
-    SpdifStreamOut(AudioHwDevice *dev, audio_output_flags_t flags,
-            audio_format_t format);
+    SpdifStreamOut(AudioHwDevice *dev, audio_format_t format);
 
     status_t open(
             audio_io_handle_t handle,
             audio_devices_t devices,
             struct audio_config *config,
+            audio_output_flags_t *flags,
             const char *address,
             const std::vector<playback_track_metadata_v7_t>& sourceMetadata) override;
 
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 1bac259..35973c1 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -477,7 +477,7 @@
                                 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 audioAttributes) = 0;
     // creates a special output that is duplicated to the two outputs passed as arguments.
     // The duplication is performed by a special mixer thread in the AudioFlinger.
diff --git a/services/audiopolicy/common/include/policy.h b/services/audiopolicy/common/include/policy.h
index 3b7cae3..d499222 100644
--- a/services/audiopolicy/common/include/policy.h
+++ b/services/audiopolicy/common/include/policy.h
@@ -29,19 +29,21 @@
 /**
  * Legacy audio policy product strategies IDs. These strategies are supported by the default
  * policy engine.
+ * IMPORTANT NOTE: the order of this enum is important as it determines the priority
+ * between active strategies for routing decisions: lower enum value => higher prioriy
  */
 enum legacy_strategy {
     STRATEGY_NONE = -1,
-    STRATEGY_MEDIA,
     STRATEGY_PHONE,
     STRATEGY_SONIFICATION,
-    STRATEGY_SONIFICATION_RESPECTFUL,
-    STRATEGY_DTMF,
     STRATEGY_ENFORCED_AUDIBLE,
-    STRATEGY_TRANSMITTED_THROUGH_SPEAKER,
     STRATEGY_ACCESSIBILITY,
-    STRATEGY_REROUTING,
+    STRATEGY_SONIFICATION_RESPECTFUL,
+    STRATEGY_MEDIA,
+    STRATEGY_DTMF,
     STRATEGY_CALL_ASSISTANT,
+    STRATEGY_TRANSMITTED_THROUGH_SPEAKER,
+    STRATEGY_REROUTING,
     STRATEGY_PATCH,
 };
 
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
index bfb28a5..a18cf1f 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -412,7 +412,7 @@
                       const audio_config_base_t *mixerConfig,
                       const DeviceVector &devices,
                       audio_stream_type_t stream,
-                      audio_output_flags_t flags,
+                      audio_output_flags_t *flags,
                       audio_io_handle_t *output,
                       audio_attributes_t attributes);
 
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/include/HwModule.h b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
index d206637..26bb94f 100644
--- a/services/audiopolicy/common/managerdefinitions/include/HwModule.h
+++ b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
@@ -137,6 +137,7 @@
 class HwModuleCollection : public Vector<sp<HwModule> >
 {
 public:
+    sp<HwModule> getModuleFromHandle(audio_module_handle_t handle) const;
     sp<HwModule> getModuleFromName(const char *name) const;
 
     /**
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index a8663fa..3c2f46a 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -587,7 +587,7 @@
                                        const audio_config_base_t *mixerConfig,
                                        const DeviceVector &devices,
                                        audio_stream_type_t stream,
-                                       audio_output_flags_t flags,
+                                       audio_output_flags_t *flags,
                                        audio_io_handle_t *output,
                                        audio_attributes_t attributes)
 {
@@ -617,7 +617,7 @@
     // create a default one
     if ((mProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
             lHalConfig.offload_info.format == AUDIO_FORMAT_DEFAULT) {
-        flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+        *flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
         lHalConfig.offload_info = AUDIO_INFO_INITIALIZER;
         lHalConfig.offload_info.sample_rate = lHalConfig.sample_rate;
         lHalConfig.offload_info.channel_mask = lHalConfig.channel_mask;
@@ -635,7 +635,7 @@
         lMixerConfig = *mixerConfig;
     }
 
-    mFlags = (audio_output_flags_t)(mFlags | flags);
+    mFlags = (audio_output_flags_t)(mFlags | *flags);
 
     // If no mixer config is specified for a spatializer output, default to 5.1 for proper
     // configuration of the final downmixer or spatializer
@@ -653,8 +653,9 @@
                                                    &lMixerConfig,
                                                    device,
                                                    &mLatency,
-                                                   mFlags,
+                                                   &mFlags,
                                                    attributes);
+    *flags = mFlags;
 
     if (status == NO_ERROR) {
         LOG_ALWAYS_FATAL_IF(*output == AUDIO_IO_HANDLE_NONE,
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyConfig.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyConfig.cpp
index abeaaf8..723887d 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>
@@ -341,7 +342,13 @@
                 AUDIO_FORMAT_AAC_XHE}},
         {AUDIO_FORMAT_DOLBY_TRUEHD, {}},
         {AUDIO_FORMAT_E_AC3_JOC, {}},
-        {AUDIO_FORMAT_AC4, {}}};
+        {AUDIO_FORMAT_AC4, {}},     // L0-3
+        {AUDIO_FORMAT_AC4_L4, {}}};
+}
+
+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/common/managerdefinitions/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
index 6696b45..2d8231a 100644
--- a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
@@ -283,6 +283,16 @@
     dumpAudioRouteVector(mRoutes, dst, spaces);
 }
 
+sp<HwModule> HwModuleCollection::getModuleFromHandle(audio_module_handle_t handle) const
+{
+    for (const auto& module : *this) {
+        if (module->getHandle() == handle) {
+            return module;
+        }
+    }
+    return nullptr;
+}
+
 sp <HwModule> HwModuleCollection::getModuleFromName(const char *name) const
 {
     for (const auto& module : *this) {
diff --git a/services/audiopolicy/config/surround_sound_configuration_5_0.xml b/services/audiopolicy/config/surround_sound_configuration_5_0.xml
index 590a181..6a268d8 100644
--- a/services/audiopolicy/config/surround_sound_configuration_5_0.xml
+++ b/services/audiopolicy/config/surround_sound_configuration_5_0.xml
@@ -27,5 +27,6 @@
     <format name="AUDIO_FORMAT_DTS_HD" />
     <format name="AUDIO_FORMAT_AAC_LC" subformats="AUDIO_FORMAT_AAC_HE_V1 AUDIO_FORMAT_AAC_HE_V2 AUDIO_FORMAT_AAC_ELD AUDIO_FORMAT_AAC_XHE" />
     <format name="AUDIO_FORMAT_AC4" />
+    <format name="AUDIO_FORMAT_AC4_L4" />
   </formats>
 </surroundSound>
diff --git a/services/audiopolicy/config/surround_sound_configuration_aidl.xml b/services/audiopolicy/config/surround_sound_configuration_aidl.xml
index cf15711..51ccaa9 100644
--- a/services/audiopolicy/config/surround_sound_configuration_aidl.xml
+++ b/services/audiopolicy/config/surround_sound_configuration_aidl.xml
@@ -30,5 +30,6 @@
     <format name="AUDIO_FORMAT_DTS_UHD_P2" />
     <format name="AUDIO_FORMAT_AAC_LC" subformats="AUDIO_FORMAT_AAC_HE_V1 AUDIO_FORMAT_AAC_HE_V2 AUDIO_FORMAT_AAC_ELD AUDIO_FORMAT_AAC_XHE" />
     <format name="AUDIO_FORMAT_AC4" />
+    <format name="AUDIO_FORMAT_AC4_L4" />
   </formats>
 </surroundSound>
diff --git a/services/audiopolicy/engine/common/include/ProductStrategy.h b/services/audiopolicy/engine/common/include/ProductStrategy.h
index 8162720..9b1125d 100644
--- a/services/audiopolicy/engine/common/include/ProductStrategy.h
+++ b/services/audiopolicy/engine/common/include/ProductStrategy.h
@@ -92,6 +92,10 @@
 
     bool isDefault() const;
 
+    bool isPatchStrategy() const {
+        return getVolumeGroupForStreamType(AUDIO_STREAM_PATCH) != VOLUME_GROUP_NONE;
+    }
+
     void dump(String8 *dst, int spaces = 0) const;
 
 private:
diff --git a/services/audiopolicy/engine/common/src/EngineBase.cpp b/services/audiopolicy/engine/common/src/EngineBase.cpp
index 976791f..fb8379e 100644
--- a/services/audiopolicy/engine/common/src/EngineBase.cpp
+++ b/services/audiopolicy/engine/common/src/EngineBase.cpp
@@ -311,6 +311,9 @@
     }
     StrategyVector orderedStrategies;
     for (const auto &iter : strategies) {
+        if (iter.second->isPatchStrategy()) {
+            continue;
+        }
         orderedStrategies.push_back(iter.second->getId());
     }
     return orderedStrategies;
@@ -742,6 +745,9 @@
     auto defaultDevices = DeviceVector(getApmObserver()->getDefaultOutputDevice());
     for (const auto &iter : getProductStrategies()) {
         const auto &strategy = iter.second;
+        if (strategy->isPatchStrategy()) {
+            continue;
+        }
         mDevicesForStrategies[strategy->getId()] = defaultDevices;
         setStrategyDevices(strategy, defaultDevices);
     }
@@ -750,6 +756,9 @@
 void EngineBase::updateDeviceSelectionCache() {
     for (const auto &iter : getProductStrategies()) {
         const auto& strategy = iter.second;
+        if (strategy->isPatchStrategy()) {
+            continue;
+        }
         auto devices = getDevicesForProductStrategy(strategy->getId());
         mDevicesForStrategies[strategy->getId()] = devices;
         setStrategyDevices(strategy, devices);
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index f36d8d5..354c59c 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -122,17 +122,16 @@
     }
 }
 
-void AudioPolicyManager::broadcastDeviceConnectionState(const sp<DeviceDescriptor> &device,
+status_t AudioPolicyManager::broadcastDeviceConnectionState(const sp<DeviceDescriptor> &device,
                                                         media::DeviceConnectedState state)
 {
     audio_port_v7 devicePort;
     device->toAudioPort(&devicePort);
-    if (status_t status = mpClientInterface->setDeviceConnectedState(&devicePort, state);
-            status != OK) {
-        ALOGE("Error %d while setting connected state %d for device %s",
-                status, static_cast<int>(state),
-                device->getDeviceTypeAddr().toString(false).c_str());
-    }
+    status_t status = mpClientInterface->setDeviceConnectedState(&devicePort, state);
+    ALOGE_IF(status != OK, "Error %d while setting connected state %d for device %s", status,
+             static_cast<int>(state), device->getDeviceTypeAddr().toString(false).c_str());
+
+    return status;
 }
 
 status_t AudioPolicyManager::setDeviceConnectionStateInt(
@@ -213,7 +212,14 @@
 
             // Before checking outputs, broadcast connect event to allow HAL to retrieve dynamic
             // parameters on newly connected devices (instead of opening the outputs...)
-            broadcastDeviceConnectionState(device, media::DeviceConnectedState::CONNECTED);
+            if (broadcastDeviceConnectionState(
+                        device, media::DeviceConnectedState::CONNECTED) != NO_ERROR) {
+                mAvailableOutputDevices.remove(device);
+                mHwModules.cleanUpForDevice(device);
+                ALOGE("%s() device %s format %x connection failed", __func__,
+                      device->toString().c_str(), device->getEncodedFormat());
+                return INVALID_OPERATION;
+            }
 
             if (checkOutputsForDevice(device, state, outputs) != NO_ERROR) {
                 mAvailableOutputDevices.remove(device);
@@ -398,7 +404,14 @@
 
             // Before checking intputs, broadcast connect event to allow HAL to retrieve dynamic
             // parameters on newly connected devices (instead of opening the inputs...)
-            broadcastDeviceConnectionState(device, media::DeviceConnectedState::CONNECTED);
+            if (broadcastDeviceConnectionState(
+                        device, media::DeviceConnectedState::CONNECTED) != NO_ERROR) {
+                mAvailableInputDevices.remove(device);
+                mHwModules.cleanUpForDevice(device);
+                ALOGE("%s() device %s format %x connection failed", __func__,
+                      device->toString().c_str(), device->getEncodedFormat());
+                return INVALID_OPERATION;
+            }
             // Propagate device availability to Engine
             setEngineDeviceConnectionState(device, state);
 
@@ -1153,8 +1166,7 @@
 
     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 +1263,8 @@
 
     // 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 +1540,13 @@
         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.
@@ -1603,14 +1623,19 @@
     releaseMsdOutputPatches(devices);
 
     status_t status =
-            outputDesc->open(config, nullptr /* mixerConfig */, devices, stream, flags, output,
+            outputDesc->open(config, nullptr /* mixerConfig */, devices, stream, &flags, output,
                              attributes);
 
-    // only accept an output with the requested parameters
+    // only accept an output with the requested parameters, unless the format can be IEC61937
+    // encapsulated and opened by AudioFlinger as wrapped IEC61937.
+    const bool ignoreRequestedParametersCheck = audio_is_iec61937_compatible(config->format)
+            && (flags & AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO)
+            && audio_has_proportional_frames(outputDesc->getFormat());
     if (status != NO_ERROR ||
-        (config->sample_rate != 0 && config->sample_rate != outputDesc->getSamplingRate()) ||
-        (config->format != AUDIO_FORMAT_DEFAULT && config->format != outputDesc->getFormat()) ||
-        (config->channel_mask != 0 && config->channel_mask != outputDesc->getChannelMask())) {
+        (!ignoreRequestedParametersCheck &&
+        ((config->sample_rate != 0 && config->sample_rate != outputDesc->getSamplingRate()) ||
+         (config->format != AUDIO_FORMAT_DEFAULT && config->format != outputDesc->getFormat()) ||
+         (config->channel_mask != 0 && config->channel_mask != outputDesc->getChannelMask())))) {
         ALOGV("%s failed opening direct output: output %d sample rate %d %d,"
                 "format %d %d, channel mask %04x %04x", __func__, *output, config->sample_rate,
                 outputDesc->getSamplingRate(), config->format, outputDesc->getFormat(),
@@ -1630,11 +1655,11 @@
     outputDesc->mDirectClientSession = session;
 
     addOutput(*output, outputDesc);
-    setOutputDevices(__func__, outputDesc,
-                     devices,
-                     true,
-                     0,
-                     NULL);
+    // The version check is essentially to avoid making this call in the case of the HIDL HAL.
+    if (auto hwModule = mHwModules.getModuleFromHandle(mPrimaryModuleHandle); hwModule &&
+            hwModule->getHalVersionMajor() >= 3) {
+        setOutputDevices(__func__, outputDesc, devices, true, 0, NULL);
+    }
     mPreviousOutputs = mOutputs;
     ALOGV("%s returns new direct output %d", __func__, *output);
     mpClientInterface->onAudioPortListUpdate();
@@ -1675,8 +1700,7 @@
     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;
     }
@@ -3472,6 +3496,11 @@
     }
     ALOGV("%s: group %d matching with %s index %d",
             __FUNCTION__, group, toString(attributes).c_str(), index);
+    if (mEngine->getStreamTypeForAttributes(attributes) == AUDIO_STREAM_PATCH) {
+        ALOGV("%s: cannot change volume for PATCH stream, attrs: %s",
+                __FUNCTION__, toString(attributes).c_str());
+        return NO_ERROR;
+    }
     status_t status = NO_ERROR;
     IVolumeCurves &curves = getVolumeCurves(attributes);
     VolumeSource vs = toVolumeSource(group);
@@ -5979,7 +6008,8 @@
         audio_devices_t deviceType = device->type();
         // Enabling/disabling formats are applied to only HDMI devices. So, this function
         // returns formats reported by HDMI devices.
-        if (deviceType != AUDIO_DEVICE_OUT_HDMI) {
+        if (deviceType != AUDIO_DEVICE_OUT_HDMI &&
+            deviceType != AUDIO_DEVICE_OUT_HDMI_ARC && deviceType != AUDIO_DEVICE_OUT_HDMI_EARC) {
             continue;
         }
         // Formats reported by sink devices
@@ -6048,13 +6078,13 @@
 
     sp<SwAudioOutputDescriptor> outputDesc;
     bool profileUpdated = false;
-    DeviceVector hdmiOutputDevices = mAvailableOutputDevices.getDevicesFromType(
-        AUDIO_DEVICE_OUT_HDMI);
+    DeviceVector hdmiOutputDevices = mAvailableOutputDevices.getDevicesFromTypes(
+        {AUDIO_DEVICE_OUT_HDMI, AUDIO_DEVICE_OUT_HDMI_ARC, AUDIO_DEVICE_OUT_HDMI_EARC});
     for (size_t i = 0; i < hdmiOutputDevices.size(); i++) {
         // Simulate reconnection to update enabled surround sound formats.
         String8 address = String8(hdmiOutputDevices[i]->address().c_str());
         std::string name = hdmiOutputDevices[i]->getName();
-        status_t status = setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_HDMI,
+        status_t status = setDeviceConnectionStateInt(hdmiOutputDevices[i]->type(),
                                                       AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
                                                       address.c_str(),
                                                       name.c_str(),
@@ -6062,7 +6092,7 @@
         if (status != NO_ERROR) {
             continue;
         }
-        status = setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_HDMI,
+        status = setDeviceConnectionStateInt(hdmiOutputDevices[i]->type(),
                                              AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
                                              address.c_str(),
                                              name.c_str(),
@@ -6602,11 +6632,12 @@
             sp<SwAudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(outProfile,
                                                                                  mpClientInterface);
             audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
+            audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
             audio_attributes_t attributes = AUDIO_ATTRIBUTES_INITIALIZER;
             status_t status = outputDesc->open(nullptr /* halConfig */, nullptr /* mixerConfig */,
                                                DeviceVector(supportedDevice),
                                                AUDIO_STREAM_DEFAULT,
-                                               AUDIO_OUTPUT_FLAG_NONE, &output, attributes);
+                                               &flags, &output, attributes);
             if (status != NO_ERROR) {
                 ALOGW("Cannot open output stream for devices %s on hw module %s",
                       supportedDevice->toString().c_str(), hwModule->getName());
@@ -7415,7 +7446,8 @@
                     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());
@@ -7975,9 +8007,21 @@
                         if (result.source == AUDIO_SOURCE_HOTWORD && !inputDesc->isSoundTrigger()) {
                             result.source = AUDIO_SOURCE_VOICE_RECOGNITION;
                         }
-                        return result; }).
+                        return result; });
             //only one input device for now
-                    addSource(device);
+            if (audio_is_remote_submix_device(device->type())) {
+                // remote submix HAL does not support audio conversion, need source device
+                // audio config to match the sink input descriptor audio config, otherwise AIDL
+                // HAL patching will fail
+                audio_port_config srcDevicePortConfig = {};
+                device->toAudioPortConfig(&srcDevicePortConfig, nullptr);
+                srcDevicePortConfig.sample_rate = inputDesc->getSamplingRate();
+                srcDevicePortConfig.channel_mask = inputDesc->getChannelMask();
+                srcDevicePortConfig.format = inputDesc->getFormat();
+                patchBuilder.addSource(srcDevicePortConfig);
+            } else {
+                patchBuilder.addSource(device);
+            }
             status = installPatch(__func__, patchHandle, inputDesc.get(), patchBuilder.patch(), 0);
         }
     }
@@ -8686,6 +8730,8 @@
     mReportedFormatsMap[devDesc] = formats;
 
     if (devDesc->type() == AUDIO_DEVICE_OUT_HDMI ||
+        devDesc->type() == AUDIO_DEVICE_OUT_HDMI_ARC ||
+        devDesc->type() == AUDIO_DEVICE_OUT_HDMI_EARC ||
         isDeviceOfModule(devDesc,AUDIO_HARDWARE_MODULE_ID_MSD)) {
         modifySurroundFormats(devDesc, &formats);
         size_t modifiedNumProfiles = 0;
@@ -8820,7 +8866,7 @@
     audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
     audio_attributes_t attributes = AUDIO_ATTRIBUTES_INITIALIZER;
     status_t status = desc->open(halConfig, mixerConfig, devices,
-            AUDIO_STREAM_DEFAULT, flags, &output, attributes);
+            AUDIO_STREAM_DEFAULT, &flags, &output, attributes);
     if (status != NO_ERROR) {
         ALOGE("%s failed to open output %d", __func__, status);
         return nullptr;
@@ -8858,7 +8904,7 @@
         config.offload_info.channel_mask = config.channel_mask;
         config.offload_info.format = config.format;
 
-        status = desc->open(&config, mixerConfig, devices, AUDIO_STREAM_DEFAULT, flags, &output,
+        status = desc->open(&config, mixerConfig, devices, AUDIO_STREAM_DEFAULT, &flags, &output,
                             attributes);
         if (status != NO_ERROR) {
             return nullptr;
@@ -8866,11 +8912,11 @@
     }
 
     addOutput(output, desc);
-    setOutputDevices(__func__, desc,
-                     devices,
-                     true,
-                     0,
-                     NULL);
+    // The version check is essentially to avoid making this call in the case of the HIDL HAL.
+    if (auto hwModule = mHwModules.getModuleFromHandle(mPrimaryModuleHandle); hwModule &&
+            hwModule->getHalVersionMajor() >= 3) {
+        setOutputDevices(__func__, desc, devices, true, 0, NULL);
+    }
     sp<DeviceDescriptor> speaker = mAvailableOutputDevices.getDevice(
             AUDIO_DEVICE_OUT_SPEAKER, String8(""), AUDIO_FORMAT_DEFAULT);
 
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 953fd2a..9d2166a 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -1105,8 +1105,8 @@
         // It can give a chance to HAL implementer to retrieve dynamic capabilities associated
         // to this device for example.
         // TODO avoid opening stream to retrieve capabilities of a profile.
-        void broadcastDeviceConnectionState(const sp<DeviceDescriptor> &device,
-                                            media::DeviceConnectedState state);
+        status_t broadcastDeviceConnectionState(const sp<DeviceDescriptor> &device,
+                                                media::DeviceConnectedState state);
 
         // updates device caching and output for streams that can influence the
         //    routing of notifications
diff --git a/services/audiopolicy/service/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
index 22fc151..6d2c772 100644
--- a/services/audiopolicy/service/AudioPolicyClientImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
@@ -56,7 +56,7 @@
                                                            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)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
@@ -74,7 +74,7 @@
     request.mixerConfig = VALUE_OR_RETURN_STATUS(
             legacy2aidl_audio_config_base_t_AudioConfigBase(*mixerConfig, false /*isInput*/));
     request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_DeviceDescriptorBase(device));
-    request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
+    request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_t_int32_t_mask(*flags));
     request.attributes = VALUE_OR_RETURN_STATUS(
             legacy2aidl_audio_attributes_t_AudioAttributes(attributes));
 
@@ -89,7 +89,9 @@
             .channel_mask = halConfig->channel_mask,
             .format = halConfig->format,
         };
-        mAudioPolicyService->registerOutput(*output, config, flags);
+        *flags = VALUE_OR_RETURN_STATUS(
+                aidl2legacy_int32_t_audio_output_flags_t_mask(response.flags));
+        mAudioPolicyService->registerOutput(*output, config, *flags);
     }
     return status;
 }
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/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index 5d9813f..eccefa7 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -790,7 +790,7 @@
                                     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);
         // creates a special output that is duplicated to the two outputs passed as arguments. The duplication is performed by
         // a special mixer thread in the AudioFlinger.
diff --git a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
index 5a25a77..483f827 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/AudioPolicyTestClient.h b/services/audiopolicy/tests/AudioPolicyTestClient.h
index 0299160..6116eab 100644
--- a/services/audiopolicy/tests/AudioPolicyTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyTestClient.h
@@ -37,7 +37,7 @@
                         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 { return NO_INIT; }
     audio_io_handle_t openDuplicateOutput(audio_io_handle_t /*output1*/,
                                           audio_io_handle_t /*output2*/) override {
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index 9445af1..5278b73 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);
@@ -222,8 +228,6 @@
     std::unique_ptr<AudioPolicyManagerTestClient> mClient;
     std::unique_ptr<AudioPolicyTestManager> mManager;
 
-    constexpr static const uint32_t k48000SamplingRate = 48000;
-
     static const std::string sTestEngineConfig;
 };
 
@@ -494,6 +498,9 @@
 void AudioPolicyManagerTestMsd::SetUpManagerConfig() {
     // TODO: Consider using Serializer to load part of the config from a string.
     ASSERT_NO_FATAL_FAILURE(AudioPolicyManagerTest::SetUpManagerConfig());
+    mConfig->getHwModules().getModuleFromName(
+            AUDIO_HARDWARE_MODULE_ID_PRIMARY)->setHalVersion(3, 0);
+
     mMsdOutputDevice = new DeviceDescriptor(AUDIO_DEVICE_OUT_BUS);
     sp<AudioProfile> pcmOutputProfile = new AudioProfile(
             AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, k48000SamplingRate);
@@ -525,7 +532,7 @@
                 addOutputProfile(spdifOutputProfile);
     }
 
-    sp<HwModule> msdModule = new HwModule(AUDIO_HARDWARE_MODULE_ID_MSD, 2 /*halVersionMajor*/);
+    sp<HwModule> msdModule = new HwModule(AUDIO_HARDWARE_MODULE_ID_MSD, 3 /*halVersionMajor*/);
     HwModuleCollection modules = mConfig->getHwModules();
     modules.add(msdModule);
     mConfig->setHwModules(modules);
@@ -787,27 +794,27 @@
 
     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;
@@ -854,27 +861,27 @@
 
     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;
@@ -1130,12 +1137,12 @@
     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);
     }
@@ -1176,9 +1183,9 @@
     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;
@@ -2513,7 +2520,7 @@
                         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 {
         return mSimulateFailure ? BAD_VALUE :
                 AudioPolicyManagerTestClient::openOutput(
@@ -2535,8 +2542,29 @@
 
     void setSimulateFailure(bool simulateFailure) { mSimulateFailure = simulateFailure; }
 
+    void setSimulateBroadcastDeviceStatus(audio_devices_t device, status_t status) {
+        if (status != NO_ERROR) {
+            // simulate device connect status
+            mSimulateBroadcastDeviceStatus[device] = status;
+        } else {
+            // remove device connection fixed status
+            mSimulateBroadcastDeviceStatus.erase(device);
+        }
+    }
+
+    status_t setDeviceConnectedState(const struct audio_port_v7* port,
+                                     media::DeviceConnectedState state) override {
+        if (mSimulateBroadcastDeviceStatus.find(port->ext.device.type) !=
+            mSimulateBroadcastDeviceStatus.end()) {
+            // If a simulated status exists, return a status value
+            return mSimulateBroadcastDeviceStatus[port->ext.device.type];
+        }
+        return AudioPolicyManagerTestClient::setDeviceConnectedState(port, state);
+    }
+
   private:
     bool mSimulateFailure = false;
+    std::map<audio_devices_t, status_t> mSimulateBroadcastDeviceStatus;
 };
 
 }  // namespace
@@ -2557,6 +2585,9 @@
     void setSimulateOpenFailure(bool simulateFailure) {
         mFullClient->setSimulateFailure(simulateFailure); }
 
+    void setSimulateBroadcastDeviceStatus(audio_devices_t device, status_t status) {
+        mFullClient->setSimulateBroadcastDeviceStatus(device, status); }
+
     static const std::string sBluetoothConfig;
 
   private:
@@ -2600,6 +2631,30 @@
     }
 }
 
+TEST_P(AudioPolicyManagerTestDeviceConnectionFailed, BroadcastDeviceFailure) {
+    const audio_devices_t type = std::get<0>(GetParam());
+    const std::string name = std::get<1>(GetParam());
+    const std::string address = std::get<2>(GetParam());
+    const audio_format_t format = std::get<3>(GetParam());
+
+    // simulate broadcastDeviceConnectionState return failure
+    setSimulateBroadcastDeviceStatus(type, INVALID_OPERATION);
+    ASSERT_EQ(INVALID_OPERATION, mManager->setDeviceConnectionState(
+            type, AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+            address.c_str(), name.c_str(), format));
+
+    // if broadcast is fail, device should not be added to available devices list
+    if (audio_is_output_device(type)) {
+        auto availableDevices = mManager->getAvailableOutputDevices();
+        EXPECT_FALSE(availableDevices.containsDeviceWithType(type));
+    } else if (audio_is_input_device(type)) {
+        auto availableDevices = mManager->getAvailableInputDevices();
+        EXPECT_FALSE(availableDevices.containsDeviceWithType(type));
+    }
+
+    setSimulateBroadcastDeviceStatus(type, NO_ERROR);
+}
+
 INSTANTIATE_TEST_CASE_P(
         DeviceConnectionFailure,
         AudioPolicyManagerTestDeviceConnectionFailed,
@@ -3184,6 +3239,259 @@
             "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;
@@ -3336,7 +3644,7 @@
     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);
 
@@ -3357,7 +3665,7 @@
     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
@@ -3368,7 +3676,7 @@
     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(
@@ -3394,7 +3702,7 @@
     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);
 
@@ -3407,7 +3715,7 @@
     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
@@ -3418,7 +3726,7 @@
     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(
@@ -3492,7 +3800,7 @@
     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);
@@ -3521,7 +3829,7 @@
     // 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 15e51b0..8e7a697 100644
--- a/services/audiopolicy/tests/resources/Android.bp
+++ b/services/audiopolicy/tests/resources/Android.bp
@@ -20,6 +20,7 @@
         "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/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/camera/virtualcamera/VirtualCameraRenderThread.h b/services/camera/virtualcamera/VirtualCameraRenderThread.h
index 5a5966b..c6b58fb 100644
--- a/services/camera/virtualcamera/VirtualCameraRenderThread.h
+++ b/services/camera/virtualcamera/VirtualCameraRenderThread.h
@@ -216,8 +216,8 @@
   std::mutex mLock;
   std::deque<std::unique_ptr<ProcessCaptureRequestTask>> mQueue GUARDED_BY(mLock);
   std::condition_variable mCondVar;
-  volatile bool mTextureUpdateRequested GUARDED_BY(mLock);
-  volatile bool mPendingExit GUARDED_BY(mLock);
+  volatile bool GUARDED_BY(mLock) mTextureUpdateRequested = false;
+  volatile bool GUARDED_BY(mLock) mPendingExit = false;
 
   // Acquisition timestamp of last frame.
   std::atomic<uint64_t> mLastAcquisitionTimestampNanoseconds;
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;