diff --git a/media/codec2/core/include/C2ParamDef.h b/media/codec2/core/include/C2ParamDef.h
index d578820..86dfe65 100644
--- a/media/codec2/core/include/C2ParamDef.h
+++ b/media/codec2/core/include/C2ParamDef.h
@@ -404,6 +404,7 @@
     /// Specialization for an input port parameter.
     struct input : public T, public S,
             public _C2StructCheck<S, ParamIndex, T::PARAM_KIND | T::Index::DIR_INPUT> {
+        using T::operator!=;
         _C2_CORE_INDEX_OVERRIDE(ParamIndex)
         /// Wrapper around base structure's constructor.
         template<typename ...Args>
@@ -416,6 +417,7 @@
     /// Specialization for an output port parameter.
     struct output : public T, public S,
             public _C2StructCheck<S, ParamIndex, T::PARAM_KIND | T::Index::DIR_OUTPUT> {
+        using T::operator!=;
         _C2_CORE_INDEX_OVERRIDE(ParamIndex)
         /// Wrapper around base structure's constructor.
         template<typename ...Args>
@@ -470,6 +472,7 @@
     /// Specialization for an input port parameter.
     struct input : public T,
             public _C2FlexStructCheck<S, ParamIndex, T::PARAM_KIND | T::Index::DIR_INPUT> {
+        using T::operator!=;
     private:
         /// Wrapper around base structure's constructor while also specifying port/direction.
         template<typename ...Args>
@@ -486,6 +489,7 @@
     /// Specialization for an output port parameter.
     struct output : public T,
             public _C2FlexStructCheck<S, ParamIndex, T::PARAM_KIND | T::Index::DIR_OUTPUT> {
+        using T::operator!=;
     private:
         /// Wrapper around base structure's constructor while also specifying port/direction.
         template<typename ...Args>
@@ -549,6 +553,7 @@
     struct input : public T, public S,
             public _C2StructCheck<S, ParamIndex,
                     T::PARAM_KIND | T::Index::IS_STREAM_FLAG | T::Type::DIR_INPUT> {
+        using T::operator!=;
         _C2_CORE_INDEX_OVERRIDE(ParamIndex)
 
         /// Default constructor. Stream-ID is undefined.
@@ -567,6 +572,7 @@
     struct output : public T, public S,
             public _C2StructCheck<S, ParamIndex,
                     T::PARAM_KIND | T::Index::IS_STREAM_FLAG | T::Type::DIR_OUTPUT> {
+        using T::operator!=;
         _C2_CORE_INDEX_OVERRIDE(ParamIndex)
 
         /// Default constructor. Stream-ID is undefined.
@@ -634,6 +640,7 @@
     struct input : public T,
             public _C2FlexStructCheck<S, ParamIndex,
                     T::PARAM_KIND | T::Index::IS_STREAM_FLAG | T::Type::DIR_INPUT> {
+        using T::operator!=;
     private:
         /// Default constructor. Stream-ID is undefined.
         inline input(size_t flexCount) : T(_Type::CalcSize(flexCount), input::PARAM_TYPE) { }
@@ -656,6 +663,7 @@
     struct output : public T,
             public _C2FlexStructCheck<S, ParamIndex,
                     T::PARAM_KIND | T::Index::IS_STREAM_FLAG | T::Type::DIR_OUTPUT> {
+        using T::operator!=;
     private:
         /// Default constructor. Stream-ID is undefined.
         inline output(size_t flexCount) : T(_Type::CalcSize(flexCount), output::PARAM_TYPE) { }
diff --git a/media/libeffects/downmix/Android.bp b/media/libeffects/downmix/Android.bp
index a5259aa..37633ae 100644
--- a/media/libeffects/downmix/Android.bp
+++ b/media/libeffects/downmix/Android.bp
@@ -56,9 +56,7 @@
         ":effectCommonFile",
     ],
     defaults: [
-        "aidlaudioservice_defaults",
-        "latest_android_hardware_audio_effect_ndk_shared",
-        "latest_android_media_audio_common_types_ndk_shared",
+        "aidlaudioeffectservice_defaults",
     ],
     header_libs: [
         "libaudioeffects",
diff --git a/media/libeffects/downmix/aidl/DownmixContext.cpp b/media/libeffects/downmix/aidl/DownmixContext.cpp
index 0e76d1d..13e0e5a 100644
--- a/media/libeffects/downmix/aidl/DownmixContext.cpp
+++ b/media/libeffects/downmix/aidl/DownmixContext.cpp
@@ -20,12 +20,60 @@
 
 #include "DownmixContext.h"
 
-using aidl::android::hardware::audio::effect::IEffect;
 using aidl::android::hardware::audio::common::getChannelCount;
+using aidl::android::hardware::audio::effect::IEffect;
 using aidl::android::media::audio::common::AudioChannelLayout;
+using aidl::android::media::audio::common::AudioConfig;
 
 namespace aidl::android::hardware::audio::effect {
 
+namespace {
+
+inline bool isChannelMaskValid(const AudioChannelLayout& channelMask) {
+    if (channelMask.getTag() != AudioChannelLayout::layoutMask) return false;
+    int chMask = channelMask.get<AudioChannelLayout::layoutMask>();
+    // check against unsupported channels (up to FCC_26)
+    constexpr uint32_t MAXIMUM_CHANNEL_MASK = AudioChannelLayout::LAYOUT_22POINT2 |
+                                              AudioChannelLayout::CHANNEL_FRONT_WIDE_LEFT |
+                                              AudioChannelLayout::CHANNEL_FRONT_WIDE_RIGHT;
+    if (chMask & ~MAXIMUM_CHANNEL_MASK) {
+        LOG(ERROR) << "Unsupported channels in " << (chMask & ~MAXIMUM_CHANNEL_MASK);
+        return false;
+    }
+    return true;
+}
+
+inline bool isStereoChannelMask(const AudioChannelLayout& channelMask) {
+    if (channelMask.getTag() != AudioChannelLayout::layoutMask) return false;
+
+    return channelMask.get<AudioChannelLayout::layoutMask>() == AudioChannelLayout::LAYOUT_STEREO;
+}
+
+}  // namespace
+
+bool DownmixContext::validateCommonConfig(const Parameter::Common& common) {
+    const AudioConfig& input = common.input;
+    const AudioConfig& output = common.output;
+    if (input.base.sampleRate != output.base.sampleRate) {
+        LOG(ERROR) << __func__ << ": SRC not supported, input: " << input.toString()
+                   << " output: " << output.toString();
+        return false;
+    }
+
+    if (!isStereoChannelMask(output.base.channelMask)) {
+        LOG(ERROR) << __func__ << ": output should be stereo, not "
+                   << output.base.channelMask.toString();
+        return false;
+    }
+
+    if (!isChannelMaskValid(input.base.channelMask)) {
+        LOG(ERROR) << __func__ << ": invalid input channel, " << input.base.channelMask.toString();
+        return false;
+    }
+
+    return true;
+}
+
 DownmixContext::DownmixContext(int statusDepth, const Parameter::Common& common)
     : EffectContext(statusDepth, common) {
     LOG(DEBUG) << __func__;
@@ -62,7 +110,7 @@
     resetBuffer();
 }
 
-IEffect::Status DownmixContext::lvmProcess(float* in, float* out, int samples) {
+IEffect::Status DownmixContext::downmixProcess(float* in, float* out, int samples) {
     LOG(DEBUG) << __func__ << " in " << in << " out " << out << " sample " << samples;
     IEffect::Status status = {EX_ILLEGAL_ARGUMENT, 0, 0};
 
@@ -122,18 +170,4 @@
     }
 }
 
-bool DownmixContext::isChannelMaskValid(AudioChannelLayout channelMask) {
-    if (channelMask.getTag() != AudioChannelLayout::layoutMask) return false;
-    int chMask = channelMask.get<AudioChannelLayout::layoutMask>();
-    // check against unsupported channels (up to FCC_26)
-    constexpr uint32_t MAXIMUM_CHANNEL_MASK = AudioChannelLayout::LAYOUT_22POINT2 |
-                                              AudioChannelLayout::CHANNEL_FRONT_WIDE_LEFT |
-                                              AudioChannelLayout::CHANNEL_FRONT_WIDE_RIGHT;
-    if (chMask & ~MAXIMUM_CHANNEL_MASK) {
-        LOG(ERROR) << "Unsupported channels in " << (chMask & ~MAXIMUM_CHANNEL_MASK);
-        return false;
-    }
-    return true;
-}
-
 }  // namespace aidl::android::hardware::audio::effect
diff --git a/media/libeffects/downmix/aidl/DownmixContext.h b/media/libeffects/downmix/aidl/DownmixContext.h
index 1571c38..a381d7f 100644
--- a/media/libeffects/downmix/aidl/DownmixContext.h
+++ b/media/libeffects/downmix/aidl/DownmixContext.h
@@ -50,7 +50,9 @@
         return RetCode::SUCCESS;
     }
 
-    IEffect::Status lvmProcess(float* in, float* out, int samples);
+    IEffect::Status downmixProcess(float* in, float* out, int samples);
+
+    static bool validateCommonConfig(const Parameter::Common& common);
 
   private:
     DownmixState mState;
@@ -60,7 +62,6 @@
 
     // Common Params
     void init_params(const Parameter::Common& common);
-    bool isChannelMaskValid(::aidl::android::media::audio::common::AudioChannelLayout channelMask);
 };
 
 }  // namespace aidl::android::hardware::audio::effect
diff --git a/media/libeffects/downmix/aidl/EffectDownmix.cpp b/media/libeffects/downmix/aidl/EffectDownmix.cpp
index 7068c5c..702a6f0 100644
--- a/media/libeffects/downmix/aidl/EffectDownmix.cpp
+++ b/media/libeffects/downmix/aidl/EffectDownmix.cpp
@@ -193,6 +193,8 @@
         return mContext;
     }
 
+    if (!DownmixContext::validateCommonConfig(common)) return nullptr;
+
     mContext = std::make_shared<DownmixContext>(1 /* statusFmqDepth */, common);
     return mContext;
 }
@@ -210,7 +212,7 @@
         LOG(ERROR) << __func__ << " nullContext";
         return {EX_NULL_POINTER, 0, 0};
     }
-    return mContext->lvmProcess(in, out, sampleToProcess);
+    return mContext->downmixProcess(in, out, sampleToProcess);
 }
 
 }  // namespace aidl::android::hardware::audio::effect
diff --git a/media/libeffects/dynamicsproc/Android.bp b/media/libeffects/dynamicsproc/Android.bp
index 7838117..9e154cf 100644
--- a/media/libeffects/dynamicsproc/Android.bp
+++ b/media/libeffects/dynamicsproc/Android.bp
@@ -86,9 +86,7 @@
     ],
 
     defaults: [
-        "aidlaudioservice_defaults",
-        "latest_android_hardware_audio_effect_ndk_shared",
-        "latest_android_media_audio_common_types_ndk_shared",
+        "aidlaudioeffectservice_defaults",
         "dynamicsprocessingdefaults",
     ],
 
diff --git a/media/libeffects/dynamicsproc/aidl/DynamicsProcessingContext.cpp b/media/libeffects/dynamicsproc/aidl/DynamicsProcessingContext.cpp
index e5e5368..f3a3860 100644
--- a/media/libeffects/dynamicsproc/aidl/DynamicsProcessingContext.cpp
+++ b/media/libeffects/dynamicsproc/aidl/DynamicsProcessingContext.cpp
@@ -313,8 +313,8 @@
 void DynamicsProcessingContext::init() {
     std::lock_guard lg(mMutex);
     mState = DYNAMICS_PROCESSING_STATE_INITIALIZED;
-    mChannelCount = ::aidl::android::hardware::audio::common::getChannelCount(
-            mCommon.input.base.channelMask);
+    mChannelCount = static_cast<int>(::aidl::android::hardware::audio::common::getChannelCount(
+            mCommon.input.base.channelMask));
 }
 
 dp_fx::DPChannel* DynamicsProcessingContext::getChannel_l(int channel) {
diff --git a/media/libeffects/dynamicsproc/aidl/DynamicsProcessingContext.h b/media/libeffects/dynamicsproc/aidl/DynamicsProcessingContext.h
index ced7f19..839c6dd 100644
--- a/media/libeffects/dynamicsproc/aidl/DynamicsProcessingContext.h
+++ b/media/libeffects/dynamicsproc/aidl/DynamicsProcessingContext.h
@@ -74,7 +74,7 @@
     static constexpr float kPreferredProcessingDurationMs = 10.0f;
     static constexpr int kBandCount = 5;
     std::mutex mMutex;
-    size_t mChannelCount GUARDED_BY(mMutex) = 0;
+    int mChannelCount GUARDED_BY(mMutex) = 0;
     DynamicsProcessingState mState GUARDED_BY(mMutex) = DYNAMICS_PROCESSING_STATE_UNINITIALIZED;
     std::unique_ptr<dp_fx::DPFrequency> mDpFreq GUARDED_BY(mMutex) = nullptr;
     bool mEngineInited GUARDED_BY(mMutex) = false;
diff --git a/media/libeffects/hapticgenerator/Android.bp b/media/libeffects/hapticgenerator/Android.bp
index fc80211..cc19a80 100644
--- a/media/libeffects/hapticgenerator/Android.bp
+++ b/media/libeffects/hapticgenerator/Android.bp
@@ -75,9 +75,7 @@
     ],
 
     defaults: [
-        "aidlaudioservice_defaults",
-        "latest_android_hardware_audio_effect_ndk_shared",
-        "latest_android_media_audio_common_types_ndk_shared",
+        "aidlaudioeffectservice_defaults",
         "hapticgeneratordefaults",
     ],
 
diff --git a/media/libeffects/hapticgenerator/aidl/HapticGeneratorContext.cpp b/media/libeffects/hapticgenerator/aidl/HapticGeneratorContext.cpp
index de44e05..354ee00 100644
--- a/media/libeffects/hapticgenerator/aidl/HapticGeneratorContext.cpp
+++ b/media/libeffects/hapticgenerator/aidl/HapticGeneratorContext.cpp
@@ -14,6 +14,7 @@
  * limitations under the License.
  */
 
+#include <cstddef>
 #define LOG_TAG "AHAL_HapticGeneratorContext"
 
 #include <Utils.h>
@@ -162,8 +163,8 @@
     }
 
     // Construct input buffer according to haptic channel source
-    for (size_t i = 0; i < mFrameCount; ++i) {
-        for (size_t j = 0; j < mParams.mHapticChannelCount; ++j) {
+    for (int64_t i = 0; i < mFrameCount; ++i) {
+        for (int j = 0; j < mParams.mHapticChannelCount; ++j) {
             mInputBuffer[i * mParams.mHapticChannelCount + j] =
                     in[i * mParams.mAudioChannelCount + mParams.mHapticChannelSource[j]];
         }
@@ -180,8 +181,7 @@
     // buffer, which contains haptic data at the end of the buffer, directly to sink buffer.
     // In that case, copy haptic data to input buffer instead of output buffer.
     // Note: this may not work with rpc/binder calls
-    int offset = samples;
-    for (int i = 0; i < hapticSampleCount; ++i) {
+    for (size_t i = 0; i < hapticSampleCount; ++i) {
         in[samples + i] = hapticOutBuffer[i];
     }
     return {STATUS_OK, samples, static_cast<int32_t>(samples + hapticSampleCount)};
@@ -199,7 +199,7 @@
     mParams.mHapticChannelCount = ::aidl::android::hardware::audio::common::getChannelCount(
             outputChMask, media::audio::common::AudioChannelLayout::LAYOUT_HAPTIC_AB);
     LOG_ALWAYS_FATAL_IF(mParams.mHapticChannelCount > 2, "haptic channel count is too large");
-    for (size_t i = 0; i < mParams.mHapticChannelCount; ++i) {
+    for (int i = 0; i < mParams.mHapticChannelCount; ++i) {
         // By default, use the first audio channel to generate haptic channels.
         mParams.mHapticChannelSource[i] = 0;
     }
diff --git a/media/libeffects/hapticgenerator/aidl/HapticGeneratorContext.h b/media/libeffects/hapticgenerator/aidl/HapticGeneratorContext.h
index a0a0a4c..26e69e4 100644
--- a/media/libeffects/hapticgenerator/aidl/HapticGeneratorContext.h
+++ b/media/libeffects/hapticgenerator/aidl/HapticGeneratorContext.h
@@ -92,7 +92,7 @@
     HapticGeneratorState mState;
     HapticGeneratorParam mParams GUARDED_BY(mMutex);
     int mSampleRate;
-    int mFrameCount = 0;
+    int64_t mFrameCount = 0;
 
     // A cache for all shared pointers of the HapticGenerator
     struct HapticGeneratorProcessorsRecord mProcessorsRecord;
diff --git a/media/libeffects/loudness/Android.bp b/media/libeffects/loudness/Android.bp
index 7acba11..05bbec3 100644
--- a/media/libeffects/loudness/Android.bp
+++ b/media/libeffects/loudness/Android.bp
@@ -54,9 +54,7 @@
         ":effectCommonFile",
     ],
     defaults: [
-        "aidlaudioservice_defaults",
-        "latest_android_hardware_audio_effect_ndk_shared",
-        "latest_android_media_audio_common_types_ndk_shared",
+        "aidlaudioeffectservice_defaults",
     ],
     header_libs: [
         "libaudioeffects",
diff --git a/media/libeffects/lvm/wrapper/Aidl/BundleContext.cpp b/media/libeffects/lvm/wrapper/Aidl/BundleContext.cpp
index a163f4b..71bb2ef 100644
--- a/media/libeffects/lvm/wrapper/Aidl/BundleContext.cpp
+++ b/media/libeffects/lvm/wrapper/Aidl/BundleContext.cpp
@@ -227,7 +227,7 @@
         bool viEnabled = params.VirtualizerOperatingMode == LVM_MODE_ON;
 
         if (eqEnabled) {
-            for (int i = 0; i < lvm::MAX_NUM_BANDS; i++) {
+            for (size_t i = 0; i < lvm::MAX_NUM_BANDS; i++) {
                 float bandFactor = mBandGainMdB[i] / 1500.0;
                 float bandCoefficient = lvm::kBandEnergyCoefficient[i];
                 float bandEnergy = bandFactor * bandCoefficient * bandCoefficient;
@@ -236,7 +236,7 @@
 
             // cross EQ coefficients
             float bandFactorSum = 0;
-            for (int i = 0; i < lvm::MAX_NUM_BANDS - 1; i++) {
+            for (size_t i = 0; i < lvm::MAX_NUM_BANDS - 1; i++) {
                 float bandFactor1 = mBandGainMdB[i] / 1500.0;
                 float bandFactor2 = mBandGainMdB[i + 1] / 1500.0;
 
@@ -259,7 +259,7 @@
             energyContribution += boostFactor * boostCoefficient * boostCoefficient;
 
             if (eqEnabled) {
-                for (int i = 0; i < lvm::MAX_NUM_BANDS; i++) {
+                for (size_t i = 0; i < lvm::MAX_NUM_BANDS; i++) {
                     float bandFactor = mBandGainMdB[i] / 1500.0;
                     float bandCrossCoefficient = lvm::kBassBoostEnergyCrossCoefficient[i];
                     float bandEnergy = boostFactor * bandFactor * bandCrossCoefficient;
@@ -421,7 +421,6 @@
 
 RetCode BundleContext::setVolumeStereo(const Parameter::VolumeStereo& volume) {
     LVM_ControlParams_t params;
-    LVM_ReturnStatus_en status = LVM_SUCCESS;
 
     // Convert volume to dB
     float leftdB = VolToDb(volume.left);
@@ -512,7 +511,7 @@
     const auto [min, max] =
             std::minmax_element(bandLevels.begin(), bandLevels.end(),
                                 [](const auto& a, const auto& b) { return a.index < b.index; });
-    return min->index >= 0 && max->index < lvm::MAX_NUM_BANDS;
+    return min->index >= 0 && static_cast<size_t>(max->index) < lvm::MAX_NUM_BANDS;
 }
 
 RetCode BundleContext::updateControlParameter(const std::vector<Equalizer::BandLevel>& bandLevels) {
diff --git a/media/libeffects/lvm/wrapper/Android.bp b/media/libeffects/lvm/wrapper/Android.bp
index fa300d2..5fb3966 100644
--- a/media/libeffects/lvm/wrapper/Android.bp
+++ b/media/libeffects/lvm/wrapper/Android.bp
@@ -110,9 +110,7 @@
     ],
     static_libs: ["libmusicbundle"],
     defaults: [
-        "aidlaudioservice_defaults",
-        "latest_android_hardware_audio_effect_ndk_shared",
-        "latest_android_media_audio_common_types_ndk_shared",
+        "aidlaudioeffectservice_defaults",
     ],
     local_include_dirs: ["Aidl"],
     header_libs: [
@@ -140,9 +138,7 @@
     ],
     static_libs: ["libreverb"],
     defaults: [
-        "aidlaudioservice_defaults",
-        "latest_android_hardware_audio_effect_ndk_shared",
-        "latest_android_media_audio_common_types_ndk_shared",
+        "aidlaudioeffectservice_defaults",
     ],
     local_include_dirs: ["Reverb/aidl"],
     header_libs: [
diff --git a/media/libeffects/lvm/wrapper/Reverb/aidl/ReverbContext.cpp b/media/libeffects/lvm/wrapper/Reverb/aidl/ReverbContext.cpp
index 79e67f2..468b268 100644
--- a/media/libeffects/lvm/wrapper/Reverb/aidl/ReverbContext.cpp
+++ b/media/libeffects/lvm/wrapper/Reverb/aidl/ReverbContext.cpp
@@ -329,7 +329,7 @@
  */
 
 int ReverbContext::convertLevel(int level) {
-    for (int i = 0; i < kLevelMapping.size(); i++) {
+    for (std::size_t i = 0; i < kLevelMapping.size(); i++) {
         if (level <= kLevelMapping[i]) {
             return i;
         }
diff --git a/media/libeffects/preprocessing/Android.bp b/media/libeffects/preprocessing/Android.bp
index d018c47..564eb36 100644
--- a/media/libeffects/preprocessing/Android.bp
+++ b/media/libeffects/preprocessing/Android.bp
@@ -67,9 +67,7 @@
         ":effectCommonFile",
     ],
     defaults: [
-        "aidlaudioservice_defaults",
-        "latest_android_hardware_audio_effect_ndk_shared",
-        "latest_android_media_audio_common_types_ndk_shared",
+        "aidlaudioeffectservice_defaults",
     ],
     local_include_dirs: ["aidl"],
     shared_libs: [
diff --git a/media/libeffects/visualizer/Android.bp b/media/libeffects/visualizer/Android.bp
index cf782f7..a8b665b 100644
--- a/media/libeffects/visualizer/Android.bp
+++ b/media/libeffects/visualizer/Android.bp
@@ -60,8 +60,6 @@
     ],
     defaults: [
         "aidlaudioeffectservice_defaults",
-        "latest_android_hardware_audio_effect_ndk_shared",
-        "latest_android_media_audio_common_types_ndk_shared",
         "visualizer_defaults",
     ],
     cflags: [
diff --git a/media/utils/MethodStatistics.cpp b/media/utils/MethodStatistics.cpp
index 086757b..80f0fc4 100644
--- a/media/utils/MethodStatistics.cpp
+++ b/media/utils/MethodStatistics.cpp
@@ -20,6 +20,8 @@
 
 // Repository for MethodStatistics Objects
 
+// It's important to have the HAL class name defined with suffix "Hidl/Aidl" because
+// TimerThread::isRequestFromHal use this string to match binder call to/from hal.
 std::shared_ptr<std::vector<std::string>>
 getStatisticsClassesForModule(std::string_view moduleName) {
     static const std::map<std::string, std::shared_ptr<std::vector<std::string>>,
@@ -34,6 +36,15 @@
                 "StreamOutHalHidl",
               })
         },
+        {
+            METHOD_STATISTICS_MODULE_NAME_AUDIO_AIDL,
+            std::shared_ptr<std::vector<std::string>>(
+                new std::vector<std::string>{
+                "DeviceHalAidl",
+                "EffectHalAidl",
+                "StreamHalAidl",
+              })
+        },
     };
     auto it = m.find(moduleName);
     if (it == m.end()) return {};
@@ -61,6 +72,9 @@
             addClassesToMap(
                     getStatisticsClassesForModule(METHOD_STATISTICS_MODULE_NAME_AUDIO_HIDL),
                     m);
+            addClassesToMap(
+                    getStatisticsClassesForModule(METHOD_STATISTICS_MODULE_NAME_AUDIO_AIDL),
+                    m);
             return m;
         }();
 
diff --git a/media/utils/TimerThread.cpp b/media/utils/TimerThread.cpp
index 3966103..25852e4 100644
--- a/media/utils/TimerThread.cpp
+++ b/media/utils/TimerThread.cpp
@@ -104,11 +104,16 @@
 //
 /* static */
 bool TimerThread::isRequestFromHal(const std::shared_ptr<const Request>& request) {
-    const size_t hidlPos = request->tag.asStringView().find("Hidl");
-    if (hidlPos == std::string::npos) return false;
-    // should be a separator afterwards Hidl which indicates the string was in the class.
-    const size_t separatorPos = request->tag.asStringView().find("::", hidlPos);
-    return separatorPos != std::string::npos;
+    for (const auto& s : {"Hidl", "Aidl"}) {
+        const auto& tagSV = request->tag.asStringView();
+        const size_t halStrPos = tagSV.find(s);
+        // should be a separator afterwards Hidl/Aidl which indicates the string was in the class.
+        if (halStrPos != std::string::npos && tagSV.find("::", halStrPos) != std::string::npos) {
+            return true;
+        }
+    }
+
+    return false;
 }
 
 struct TimerThread::SnapshotAnalysis TimerThread::getSnapshotAnalysis(size_t retiredCount) const {
diff --git a/media/utils/include/mediautils/MethodStatistics.h b/media/utils/include/mediautils/MethodStatistics.h
index c8b36d8..2543dfa 100644
--- a/media/utils/include/mediautils/MethodStatistics.h
+++ b/media/utils/include/mediautils/MethodStatistics.h
@@ -124,6 +124,7 @@
 // Managed Statistics support.
 // Supported Modules
 #define METHOD_STATISTICS_MODULE_NAME_AUDIO_HIDL "AudioHidl"
+#define METHOD_STATISTICS_MODULE_NAME_AUDIO_AIDL "AudioAidl"
 
 // Returns a vector of class names for the module, or a nullptr if module not found.
 std::shared_ptr<std::vector<std::string>>
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 98b6c27..0ebb8ed 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -859,12 +859,15 @@
             dprintf(fd, "\nIEffect binder call profile:\n");
             write(fd, timeCheckStats.c_str(), timeCheckStats.size());
 
-            // Automatically fetch HIDL statistics.
-            std::shared_ptr<std::vector<std::string>> hidlClassNames =
-                    mediautils::getStatisticsClassesForModule(
-                            METHOD_STATISTICS_MODULE_NAME_AUDIO_HIDL);
-            if (hidlClassNames) {
-                for (const auto& className : *hidlClassNames) {
+            // Automatically fetch HIDL or AIDL statistics.
+            const std::string_view halType = (mDevicesFactoryHal->getHalVersion().getType() ==
+                                      AudioHalVersionInfo::Type::HIDL)
+                                             ? METHOD_STATISTICS_MODULE_NAME_AUDIO_HIDL
+                                             : METHOD_STATISTICS_MODULE_NAME_AUDIO_AIDL;
+            const std::shared_ptr<std::vector<std::string>> halClassNames =
+                    mediautils::getStatisticsClassesForModule(halType);
+            if (halClassNames) {
+                for (const auto& className : *halClassNames) {
                     auto stats = mediautils::getStatisticsForClass(className);
                     if (stats) {
                         timeCheckStats = stats->dump();
@@ -3196,41 +3199,19 @@
         return 0;
     }
 
-    audio_config_t halconfig = *config;
-    sp<DeviceHalInterface> inHwHal = inHwDev->hwDevice();
-    sp<StreamInHalInterface> inStream;
-    status_t status = inHwHal->openInputStream(
-            *input, devices, &halconfig, flags, address, source,
-            outputDevice, outputDeviceAddress, &inStream);
-    ALOGV("openInput_l() openInputStream returned input %p, devices %#x, SamplingRate %d"
-           ", Format %#x, Channels %#x, flags %#x, status %d addr %s",
-            inStream.get(),
+    AudioStreamIn *inputStream = nullptr;
+    status_t status = inHwDev->openInputStream(
+            &inputStream,
+            *input,
             devices,
-            halconfig.sample_rate,
-            halconfig.format,
-            halconfig.channel_mask,
             flags,
-            status, address);
+            config,
+            address,
+            source,
+            outputDevice,
+            outputDeviceAddress.c_str());
 
-    // If the input could not be opened with the requested parameters and we can handle the
-    // conversion internally, try to open again with the proposed parameters.
-    if (status == BAD_VALUE &&
-        audio_is_linear_pcm(config->format) &&
-        audio_is_linear_pcm(halconfig.format) &&
-        (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) &&
-        (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_LIMIT) &&
-        (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_LIMIT)) {
-        // FIXME describe the change proposed by HAL (save old values so we can log them here)
-        ALOGV("openInput_l() reopening with proposed sampling rate and channel mask");
-        inStream.clear();
-        status = inHwHal->openInputStream(
-                *input, devices, &halconfig, flags, address, source,
-                outputDevice, outputDeviceAddress, &inStream);
-        // FIXME log this new status; HAL should not propose any further changes
-    }
-
-    if (status == NO_ERROR && inStream != 0) {
-        AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream, flags);
+    if (status == NO_ERROR) {
         if ((flags & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0) {
             const sp<IAfMmapCaptureThread> thread =
                     IAfMmapCaptureThread::create(this, *input, inHwDev, inputStream, mSystemReady);
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index d8d727c..4c6b02c 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -9544,10 +9544,24 @@
 
 void RecordThread::readInputParameters_l()
 {
-    status_t result = mInput->stream->getAudioProperties(&mSampleRate, &mChannelMask, &mHALFormat);
-    LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving audio properties from HAL: %d", result);
-    mFormat = mHALFormat;
+    const audio_config_base_t audioConfig = mInput->getAudioProperties();
+    mSampleRate = audioConfig.sample_rate;
+    mChannelMask = audioConfig.channel_mask;
+    if (!audio_is_input_channel(mChannelMask)) {
+        LOG_ALWAYS_FATAL("Channel mask %#x not valid for input", mChannelMask);
+    }
+
     mChannelCount = audio_channel_count_from_in_mask(mChannelMask);
+
+    // Get actual HAL format.
+    status_t result = mInput->stream->getAudioProperties(nullptr, nullptr, &mHALFormat);
+    LOG_ALWAYS_FATAL_IF(result != OK, "Error when retrieving input stream format: %d", result);
+    // Get format from the shim, which will be different than the HAL format
+    // if recording compressed audio from IEC61937 wrapped sources.
+    mFormat = audioConfig.format;
+    if (!audio_is_valid_format(mFormat)) {
+        LOG_ALWAYS_FATAL("Format %#x not valid for input", mFormat);
+    }
     if (audio_is_linear_pcm(mFormat)) {
         LOG_ALWAYS_FATAL_IF(mChannelCount > FCC_LIMIT, "HAL channel count %d > %d",
                 mChannelCount, FCC_LIMIT);
@@ -9555,8 +9569,7 @@
         // Can have more that FCC_LIMIT channels in encoded streams.
         ALOGI("HAL format %#x is not linear pcm", mFormat);
     }
-    result = mInput->stream->getFrameSize(&mFrameSize);
-    LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving frame size from HAL: %d", result);
+    mFrameSize = mInput->getFrameSize();
     LOG_ALWAYS_FATAL_IF(mFrameSize <= 0, "Error frame size was %zu but must be greater than zero",
             mFrameSize);
     result = mInput->stream->getBufferSize(&mBufferSize);
diff --git a/services/audioflinger/datapath/Android.bp b/services/audioflinger/datapath/Android.bp
index ee98aef..4235f14 100644
--- a/services/audioflinger/datapath/Android.bp
+++ b/services/audioflinger/datapath/Android.bp
@@ -43,11 +43,14 @@
 
     srcs: [
         "AudioHwDevice.cpp",
+        "AudioStreamIn.cpp",
         "AudioStreamOut.cpp",
+        "SpdifStreamIn.cpp",
         "SpdifStreamOut.cpp",
     ],
 
     header_libs: [
+        "libaudioclient_headers",
         "libaudiohal_headers",
         "liberror_headers",
     ],
diff --git a/services/audioflinger/datapath/AudioHwDevice.cpp b/services/audioflinger/datapath/AudioHwDevice.cpp
index 67e9991..95e9ecc 100644
--- a/services/audioflinger/datapath/AudioHwDevice.cpp
+++ b/services/audioflinger/datapath/AudioHwDevice.cpp
@@ -1,19 +1,19 @@
 /*
-**
-** Copyright 2007, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-**     http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
+ *
+ * Copyright 2007, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *     http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
 
 #define LOG_TAG "AudioHwDevice"
 //#define LOG_NDEBUG 0
@@ -21,10 +21,13 @@
 #include <system/audio.h>
 #include <utils/Log.h>
 
+#include <audio_utils/spdif/SPDIFDecoder.h>
 #include <audio_utils/spdif/SPDIFEncoder.h>
+#include <media/AudioResamplerPublic.h>
 
 #include "AudioHwDevice.h"
 #include "AudioStreamOut.h"
+#include "SpdifStreamIn.h"
 #include "SpdifStreamOut.h"
 
 namespace android {
@@ -47,12 +50,8 @@
     auto outputStream = new AudioStreamOut(this, flags);
 
     // 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);
+    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);
 
     if (status != NO_ERROR) {
@@ -62,13 +61,8 @@
         // FIXME Look at any modification to the config.
         // The HAL might modify the config to suggest a wrapped format.
         // Log this so we can see what the HALs are doing.
-        ALOGI("openOutputStream(), HAL returned"
-            " sampleRate %d, Format %#x, "
-            "channelMask %#x, status %d",
-            config->sample_rate,
-            config->format,
-            config->channel_mask,
-            status);
+        ALOGI("openOutputStream(), HAL returned sampleRate %d, format %#x, channelMask %#x,"
+                " status %d", config->sample_rate, config->format, config->channel_mask, status);
 
         // If the data is encoded then try again using wrapped PCM.
         const bool wrapperNeeded = !audio_has_proportional_frames(originalConfig.format)
@@ -96,6 +90,79 @@
     return status;
 }
 
+status_t AudioHwDevice::openInputStream(
+        AudioStreamIn **ppStreamIn,
+        audio_io_handle_t handle,
+        audio_devices_t deviceType,
+        audio_input_flags_t flags,
+        struct audio_config *config,
+        const char *address,
+        audio_source_t source,
+        audio_devices_t outputDevice,
+        const char *outputDeviceAddress) {
+
+    struct audio_config originalConfig = *config;
+    auto inputStream = new AudioStreamIn(this, flags);
+
+    // Try to open the HAL first using the current format.
+    ALOGV("openInputStream(), try sampleRate %d, format %#x, channelMask %#x", config->sample_rate,
+            config->format, config->channel_mask);
+    status_t status = inputStream->open(handle, deviceType, config, address, source, outputDevice,
+                                        outputDeviceAddress);
+
+    // If the input could not be opened with the requested parameters and we can handle the
+    // conversion internally, try to open again with the proposed parameters.
+    if (status == BAD_VALUE &&
+        audio_is_linear_pcm(originalConfig.format) &&
+        audio_is_linear_pcm(config->format) &&
+        (config->sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) &&
+        (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_LIMIT) &&
+        (audio_channel_count_from_in_mask(originalConfig.channel_mask) <= FCC_LIMIT)) {
+        // FIXME describe the change proposed by HAL (save old values so we can log them here)
+        ALOGV("openInputStream() reopening with proposed sampling rate and channel mask");
+        status = inputStream->open(handle, deviceType, config, address, source,
+                outputDevice, outputDeviceAddress);
+        // FIXME log this new status; HAL should not propose any further changes
+        if (status != NO_ERROR) {
+            delete inputStream;
+            inputStream = nullptr;
+        }
+    } else if (status != NO_ERROR) {
+        delete inputStream;
+        inputStream = nullptr;
+
+        // FIXME Look at any modification to the config.
+        // The HAL might modify the config to suggest a wrapped format.
+        // Log this so we can see what the HALs are doing.
+        ALOGI("openInputStream(), HAL returned sampleRate %d, format %#x, channelMask %#x,"
+                " status %d", config->sample_rate, config->format, config->channel_mask, status);
+
+        // If the data is encoded then try again using wrapped PCM.
+        const bool unwrapperNeeded = !audio_has_proportional_frames(originalConfig.format)
+                && ((flags & AUDIO_INPUT_FLAG_DIRECT) != 0);
+
+        if (unwrapperNeeded) {
+            if (SPDIFDecoder::isFormatSupported(originalConfig.format)) {
+                inputStream = new SpdifStreamIn(this, flags, originalConfig.format);
+                status = inputStream->open(handle, deviceType, &originalConfig, address, source,
+                        outputDevice, outputDeviceAddress);
+                if (status != NO_ERROR) {
+                    ALOGE("ERROR - openInputStream(), SPDIF open returned %d",
+                        status);
+                    delete inputStream;
+                    inputStream = nullptr;
+                }
+            } else {
+                ALOGE("ERROR - openInputStream(), SPDIFDecoder does not support format 0x%08x",
+                    originalConfig.format);
+            }
+        }
+    }
+
+    *ppStreamIn = inputStream;
+    return status;
+}
+
 bool AudioHwDevice::supportsAudioPatches() const {
     bool result;
     return mHwDevice->supportsAudioPatches(&result) == OK ? result : false;
diff --git a/services/audioflinger/datapath/AudioHwDevice.h b/services/audioflinger/datapath/AudioHwDevice.h
index cfb6fbd..80c1473 100644
--- a/services/audioflinger/datapath/AudioHwDevice.h
+++ b/services/audioflinger/datapath/AudioHwDevice.h
@@ -1,22 +1,21 @@
 /*
-**
-** Copyright 2007, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-**     http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
+ *
+ * Copyright 2007, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *     http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
 
-#ifndef ANDROID_AUDIO_HW_DEVICE_H
-#define ANDROID_AUDIO_HW_DEVICE_H
+#pragma once
 
 #include <stdint.h>
 #include <stdlib.h>
@@ -30,6 +29,7 @@
 
 namespace android {
 
+class AudioStreamIn;
 class AudioStreamOut;
 
 class AudioHwDevice {
@@ -89,6 +89,17 @@
             struct audio_config *config,
             const char *address);
 
+    status_t openInputStream(
+            AudioStreamIn **ppStreamIn,
+            audio_io_handle_t handle,
+            audio_devices_t deviceType,
+            audio_input_flags_t flags,
+            struct audio_config *config,
+            const char *address,
+            audio_source_t source,
+            audio_devices_t outputDevice,
+            const char *outputDeviceAddress);
+
     [[nodiscard]] bool supportsAudioPatches() const;
 
     [[nodiscard]] status_t getAudioPort(struct audio_port_v7 *port) const;
@@ -112,5 +123,3 @@
 };
 
 } // namespace android
-
-#endif // ANDROID_AUDIO_HW_DEVICE_H
diff --git a/services/audioflinger/datapath/AudioStreamIn.cpp b/services/audioflinger/datapath/AudioStreamIn.cpp
new file mode 100644
index 0000000..24f3bb9
--- /dev/null
+++ b/services/audioflinger/datapath/AudioStreamIn.cpp
@@ -0,0 +1,137 @@
+/*
+ *
+ * Copyright 2023, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *     http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioFlinger"
+//#define LOG_NDEBUG 0
+#include "AudioStreamIn.h"
+
+#include <media/audiohal/DeviceHalInterface.h>
+#include <media/audiohal/StreamHalInterface.h>
+#include <system/audio.h>
+#include <utils/Log.h>
+
+#include "AudioHwDevice.h"
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+AudioStreamIn::AudioStreamIn(AudioHwDevice *dev, audio_input_flags_t flags)
+        : audioHwDev(dev)
+        , flags(flags)
+{
+}
+
+// This must be defined here together with the HAL includes above and
+// not solely in the header.
+AudioStreamIn::~AudioStreamIn() = default;
+
+sp<DeviceHalInterface> AudioStreamIn::hwDev() const
+{
+    return audioHwDev->hwDevice();
+}
+
+status_t AudioStreamIn::getCapturePosition(int64_t* frames, int64_t* time)
+{
+    if (stream == nullptr) {
+        return NO_INIT;
+    }
+
+    int64_t halPosition = 0;
+    const status_t status = stream->getCapturePosition(&halPosition, time);
+    if (status != NO_ERROR) {
+        return status;
+    }
+
+    // Adjust for standby using HAL rate frames.
+    // Only apply this correction if the HAL is getting PCM frames.
+    if (mHalFormatHasProportionalFrames) {
+        const uint64_t adjustedPosition = (halPosition <= mFramesReadAtStandby) ?
+                0 : (halPosition - mFramesReadAtStandby);
+        // Scale from HAL sample rate to application rate.
+        *frames = adjustedPosition / mRateMultiplier;
+    } else {
+        // For compressed formats.
+        *frames = halPosition;
+    }
+
+    return status;
+}
+
+status_t AudioStreamIn::open(
+        audio_io_handle_t handle,
+        audio_devices_t deviceType,
+        struct audio_config *config,
+        const char *address,
+        audio_source_t source,
+        audio_devices_t outputDevice,
+        const char *outputDeviceAddress)
+{
+    sp<StreamInHalInterface> inStream;
+
+    int status = hwDev()->openInputStream(
+            handle,
+            deviceType,
+            config,
+            flags,
+            address,
+            source,
+            outputDevice,
+            outputDeviceAddress,
+            &inStream);
+    ALOGV("AudioStreamIn::open(), HAL returned stream %p, sampleRate %d, format %#x,"
+            " channelMask %#x, status %d", inStream.get(), config->sample_rate, config->format,
+            config->channel_mask, status);
+
+    if (status == NO_ERROR) {
+        stream = inStream;
+        mHalFormatHasProportionalFrames = audio_has_proportional_frames(config->format);
+        status = stream->getFrameSize(&mHalFrameSize);
+        LOG_ALWAYS_FATAL_IF(status != OK, "Error retrieving frame size from HAL: %d", status);
+        LOG_ALWAYS_FATAL_IF(mHalFrameSize == 0, "Error frame size was %zu but must be greater than"
+                " zero", mHalFrameSize);
+    }
+
+    return status;
+}
+
+audio_config_base_t AudioStreamIn::getAudioProperties() const
+{
+    audio_config_base_t result = AUDIO_CONFIG_BASE_INITIALIZER;
+    if (stream->getAudioProperties(&result) != OK) {
+        result.sample_rate = 0;
+        result.channel_mask = AUDIO_CHANNEL_INVALID;
+        result.format = AUDIO_FORMAT_INVALID;
+    }
+    return result;
+}
+
+status_t AudioStreamIn::standby()
+{
+    mFramesReadAtStandby = mFramesRead;
+    return stream->standby();
+}
+
+status_t AudioStreamIn::read(void* buffer, size_t bytes, size_t* read)
+{
+    const status_t result = stream->read(buffer, bytes, read);
+    if (result == OK && *read > 0 && mHalFrameSize > 0) {
+        mFramesRead += *read / mHalFrameSize;
+    }
+    return result;
+}
+
+} // namespace android
diff --git a/services/audioflinger/datapath/AudioStreamIn.h b/services/audioflinger/datapath/AudioStreamIn.h
index 604a4e4..6d1c6a7 100644
--- a/services/audioflinger/datapath/AudioStreamIn.h
+++ b/services/audioflinger/datapath/AudioStreamIn.h
@@ -31,30 +31,57 @@
     virtual status_t standby() = 0;
 };
 
-// AudioStreamIn is immutable, so its fields are const.
-// The methods must not be const to match StreamHalInterface signature.
-
-struct AudioStreamIn : public Source {
+/**
+ * Managed access to a HAL input stream.
+ */
+class AudioStreamIn : public Source {
+public:
     const AudioHwDevice* const audioHwDev;
-    const sp<StreamInHalInterface> stream;
+    sp<StreamInHalInterface> stream;
     const audio_input_flags_t flags;
 
-    AudioStreamIn(
-            const AudioHwDevice* dev, const sp<StreamInHalInterface>& in,
-            audio_input_flags_t flags)
-        : audioHwDev(dev), stream(in), flags(flags) {}
+    [[nodiscard]] sp<DeviceHalInterface> hwDev() const;
 
-    status_t read(void* buffer, size_t bytes, size_t* read) final {
-        return stream->read(buffer, bytes, read);
-    }
+    AudioStreamIn(AudioHwDevice *dev, audio_input_flags_t flags);
 
-    status_t getCapturePosition(int64_t* frames, int64_t* time) final {
-        return stream->getCapturePosition(frames, time);
-    }
+    virtual status_t open(
+            audio_io_handle_t handle,
+            audio_devices_t deviceType,
+            struct audio_config *config,
+            const char *address,
+            audio_source_t source,
+            audio_devices_t outputDevice,
+            const char *outputDeviceAddress);
 
-    status_t standby() final { return stream->standby(); }
+    ~AudioStreamIn() override;
 
-    sp<DeviceHalInterface> hwDev() const { return audioHwDev->hwDevice(); }
+    status_t getCapturePosition(int64_t* frames, int64_t* time) override;
+
+    status_t read(void* buffer, size_t bytes, size_t* read) override;
+
+    /**
+     * @return frame size from the perspective of the application and the AudioFlinger.
+     */
+    [[nodiscard]] virtual size_t getFrameSize() const { return mHalFrameSize; }
+
+    /**
+     * @return audio stream configuration: channel mask, format, sample rate:
+     *   - channel mask from the perspective of the application and the AudioFlinger,
+     *     The HAL is in stereo mode when playing multi-channel compressed audio over HDMI;
+     *   - format from the perspective of the application and the AudioFlinger;
+     *   - sample rate from the perspective of the application and the AudioFlinger,
+     *     The HAL may be running at a higher sample rate if, for example, playing wrapped EAC3.
+     */
+    [[nodiscard]] virtual audio_config_base_t getAudioProperties() const;
+
+    status_t standby() override;
+
+protected:
+    uint64_t mFramesRead = 0;
+    int64_t mFramesReadAtStandby = 0;
+    int mRateMultiplier = 1;
+    bool mHalFormatHasProportionalFrames = false;
+    size_t mHalFrameSize = 0;
 };
 
 }  // namespace android
diff --git a/services/audioflinger/datapath/AudioStreamOut.cpp b/services/audioflinger/datapath/AudioStreamOut.cpp
index 6fa82e5..1830d15 100644
--- a/services/audioflinger/datapath/AudioStreamOut.cpp
+++ b/services/audioflinger/datapath/AudioStreamOut.cpp
@@ -1,30 +1,31 @@
 /*
-**
-** Copyright 2015, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-**     http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
+ *
+ * Copyright 2015, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *     http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
 
 #define LOG_TAG "AudioFlinger"
 //#define LOG_NDEBUG 0
 
+#include "AudioStreamOut.h"
+
 #include <media/audiohal/DeviceHalInterface.h>
 #include <media/audiohal/StreamHalInterface.h>
 #include <system/audio.h>
 #include <utils/Log.h>
 
 #include "AudioHwDevice.h"
-#include "AudioStreamOut.h"
 
 namespace android {
 
@@ -132,14 +133,9 @@
             config,
             address,
             &outStream);
-    ALOGV("AudioStreamOut::open(), HAL returned "
-            " stream %p, sampleRate %d, Format %#x, "
-            "channelMask %#x, status %d",
-            outStream.get(),
-            config->sample_rate,
-            config->format,
-            config->channel_mask,
-            status);
+    ALOGV("AudioStreamOut::open(), HAL returned stream %p, sampleRate %d, format %#x,"
+            " channelMask %#x, status %d", outStream.get(), config->sample_rate, config->format,
+            config->channel_mask, status);
 
     // Some HALs may not recognize AUDIO_FORMAT_IEC61937. But if we declare
     // it as PCM then it will probably work.
@@ -162,7 +158,7 @@
         mHalFormatHasProportionalFrames = audio_has_proportional_frames(config->format);
         status = stream->getFrameSize(&mHalFrameSize);
         LOG_ALWAYS_FATAL_IF(status != OK, "Error retrieving frame size from HAL: %d", status);
-        LOG_ALWAYS_FATAL_IF(mHalFrameSize <= 0, "Error frame size was %zu but must be greater than"
+        LOG_ALWAYS_FATAL_IF(mHalFrameSize == 0, "Error frame size was %zu but must be greater than"
                 " zero", mHalFrameSize);
 
     }
diff --git a/services/audioflinger/datapath/AudioStreamOut.h b/services/audioflinger/datapath/AudioStreamOut.h
index ce00f8c..ea41bba 100644
--- a/services/audioflinger/datapath/AudioStreamOut.h
+++ b/services/audioflinger/datapath/AudioStreamOut.h
@@ -1,27 +1,28 @@
 /*
-**
-** Copyright 2015, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-**     http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
+ *
+ * Copyright 2015, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *     http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
 
-#ifndef ANDROID_AUDIO_STREAM_OUT_H
-#define ANDROID_AUDIO_STREAM_OUT_H
+#pragma once
 
 #include <stdint.h>
 #include <sys/types.h>
 
 #include <system/audio.h>
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
 
 namespace android {
 
@@ -34,9 +35,6 @@
  */
 class AudioStreamOut {
 public:
-// AudioStreamOut is immutable, so its fields are const.
-// For emphasis, we could also make all pointers to them be "const *",
-// but that would clutter the code unnecessarily.
     AudioHwDevice * const audioHwDev;
     sp<StreamOutHalInterface> stream;
     const audio_output_flags_t flags;
@@ -101,15 +99,13 @@
     virtual void presentationComplete() { mExpectRetrograde = true; }
 
 protected:
-    uint64_t             mFramesWritten = 0; // reset by flush
-    uint64_t             mFramesWrittenAtStandby = 0;
-    uint64_t             mRenderPosition = 0; // reset by flush, standby, or presentation complete
-    int                  mRateMultiplier = 1;
-    bool                 mHalFormatHasProportionalFrames = false;
-    size_t               mHalFrameSize = 0;
-    bool                 mExpectRetrograde = false; // see presentationComplete
+    uint64_t mFramesWritten = 0; // reset by flush
+    uint64_t mFramesWrittenAtStandby = 0;
+    uint64_t mRenderPosition = 0; // reset by flush, standby, or presentation complete
+    int mRateMultiplier = 1;
+    bool mHalFormatHasProportionalFrames = false;
+    size_t mHalFrameSize = 0;
+    bool mExpectRetrograde = false; // see presentationComplete
 };
 
 } // namespace android
-
-#endif // ANDROID_AUDIO_STREAM_OUT_H
diff --git a/services/audioflinger/datapath/SpdifStreamIn.cpp b/services/audioflinger/datapath/SpdifStreamIn.cpp
new file mode 100644
index 0000000..98ce712
--- /dev/null
+++ b/services/audioflinger/datapath/SpdifStreamIn.cpp
@@ -0,0 +1,128 @@
+/*
+ *
+ * Copyright 2023, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *     http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioFlinger"
+//#define LOG_NDEBUG 0
+#include "Configuration.h"
+#include <system/audio.h>
+#include <utils/Log.h>
+
+#include <audio_utils/spdif/SPDIFDecoder.h>
+
+#include "AudioHwDevice.h"
+#include "SpdifStreamIn.h"
+
+namespace android {
+
+/**
+ * If the HAL is generating IEC61937 data and AudioFlinger expects elementary stream then we need to
+ * extract the data using an SPDIF decoder.
+ */
+SpdifStreamIn::SpdifStreamIn(AudioHwDevice *dev,
+            audio_input_flags_t flags,
+            audio_format_t format)
+        : AudioStreamIn(dev, flags)
+        , mSpdifDecoder(this, format)
+{
+}
+
+status_t SpdifStreamIn::open(
+        audio_io_handle_t handle,
+        audio_devices_t devices,
+        struct audio_config *config,
+        const char *address,
+        audio_source_t source,
+        audio_devices_t outputDevice,
+        const char* outputDeviceAddress)
+{
+    struct audio_config customConfig = *config;
+
+    mApplicationConfig.format = config->format;
+    mApplicationConfig.sample_rate = config->sample_rate;
+    mApplicationConfig.channel_mask = config->channel_mask;
+
+    mRateMultiplier = spdif_rate_multiplier(config->format);
+    if (mRateMultiplier <= 0) {
+        ALOGE("ERROR SpdifStreamIn::open() unrecognized format 0x%08X\n", config->format);
+        return BAD_VALUE;
+    }
+    customConfig.sample_rate = config->sample_rate * mRateMultiplier;
+    customConfig.format = AUDIO_FORMAT_PCM_16_BIT;
+    customConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO;
+
+    // Always print this because otherwise it could be very confusing if the
+    // HAL and AudioFlinger are using different formats.
+    // Print before open() because HAL may modify customConfig.
+    ALOGI("SpdifStreamIn::open() AudioFlinger requested sampleRate %d, format %#x, channelMask %#x",
+            config->sample_rate, config->format, config->channel_mask);
+    ALOGI("SpdifStreamIn::open() HAL configured for sampleRate %d, format %#x, channelMask %#x",
+            customConfig.sample_rate, customConfig.format, customConfig.channel_mask);
+
+    const status_t status = AudioStreamIn::open(
+            handle,
+            devices,
+            &customConfig,
+            address,
+            source,
+            outputDevice,
+            outputDeviceAddress);
+
+    ALOGI("SpdifStreamIn::open() status = %d", status);
+
+#ifdef TEE_SINK
+    if (status == OK) {
+        // Don't use PCM 16-bit format to avoid WAV encoding IEC61937 data.
+        mTee.set(customConfig.sample_rate,
+                audio_channel_count_from_in_mask(customConfig.channel_mask),
+                AUDIO_FORMAT_IEC61937, NBAIO_Tee::TEE_FLAG_INPUT_THREAD);
+        mTee.setId(std::string("_") + std::to_string(handle) + "_C");
+    }
+#endif
+
+    return status;
+}
+
+int SpdifStreamIn::standby()
+{
+    mSpdifDecoder.reset();
+    return AudioStreamIn::standby();
+}
+
+status_t SpdifStreamIn::readDataBurst(void* buffer, size_t bytes, size_t* read)
+{
+    status_t status = AudioStreamIn::read(buffer, bytes, read);
+
+#ifdef TEE_SINK
+    if (*read > 0) {
+        mTee.write(reinterpret_cast<const char *>(buffer), *read / AudioStreamIn::getFrameSize());
+    }
+#endif
+    return status;
+}
+
+status_t SpdifStreamIn::read(void* buffer, size_t numBytes, size_t* read)
+{
+    // Read from SPDIF extractor. It will call back to readDataBurst().
+    const auto bytesRead = mSpdifDecoder.read(buffer, numBytes);
+    if (bytesRead >= 0) {
+        *read = bytesRead;
+        return OK;
+    }
+    return NOT_ENOUGH_DATA;
+}
+
+} // namespace android
diff --git a/services/audioflinger/datapath/SpdifStreamIn.h b/services/audioflinger/datapath/SpdifStreamIn.h
new file mode 100644
index 0000000..78832ee
--- /dev/null
+++ b/services/audioflinger/datapath/SpdifStreamIn.h
@@ -0,0 +1,134 @@
+/*
+ *
+ * Copyright 2023, 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 <stdint.h>
+#include <sys/types.h>
+
+#include <system/audio.h>
+
+#include "AudioStreamIn.h"
+
+#include <audio_utils/spdif/SPDIFDecoder.h>
+#include <afutils/NBAIO_Tee.h>
+
+namespace android {
+
+/**
+ * Stream that is a PCM data burst in the HAL but looks like an encoded stream
+ * to the AudioFlinger. Wraps encoded data in an SPDIF wrapper per IEC61973-3.
+ */
+class SpdifStreamIn : public AudioStreamIn {
+public:
+
+    SpdifStreamIn(AudioHwDevice *dev, audio_input_flags_t flags,
+            audio_format_t format);
+
+    status_t open(
+            audio_io_handle_t handle,
+            audio_devices_t devices,
+            struct audio_config *config,
+            const char *address,
+            audio_source_t source,
+            audio_devices_t outputDevice,
+            const char* outputDeviceAddress) override;
+
+    /**
+    * Read audio buffer from driver. If at least one frame was read successfully prior to the error,
+    * it is suggested that the driver return that successful (short) byte count
+    * and then return an error in the subsequent call.
+    *
+    * If set_callback() has previously been called to enable non-blocking mode
+    * the write() is not allowed to block. It must write only the number of
+    * bytes that currently fit in the driver/hardware buffer and then return
+    * this byte count. If this is less than the requested write size the
+    * callback function must be called when more space is available in the
+    * driver/hardware buffer.
+    */
+    status_t read(void* buffer, size_t bytes, size_t* read) override;
+
+    /**
+     * @return frame size from the perspective of the application and the AudioFlinger.
+     */
+    [[nodiscard]] size_t getFrameSize() const override { return sizeof(int8_t); }
+
+    /**
+     * @return audio_config_base_t from the perspective of the application and the AudioFlinger.
+     */
+    [[nodiscard]] audio_config_base_t getAudioProperties() const override {
+        return mApplicationConfig;
+    }
+
+    /**
+     * @return format from the perspective of the application and the AudioFlinger.
+     */
+    [[nodiscard]] virtual audio_format_t getFormat() const { return mApplicationConfig.format; }
+
+    /**
+     * The HAL may be running at a higher sample rate if, for example, reading wrapped EAC3.
+     * @return sample rate from the perspective of the application and the AudioFlinger.
+     */
+    [[nodiscard]] virtual uint32_t getSampleRate() const { return mApplicationConfig.sample_rate; }
+
+    /**
+     * The HAL is in stereo mode when reading multi-channel compressed audio.
+     * @return channel mask from the perspective of the application and the AudioFlinger.
+     */
+    [[nodiscard]] virtual audio_channel_mask_t getChannelMask() const {
+        return mApplicationConfig.channel_mask;
+    }
+
+    status_t standby() override;
+
+private:
+
+    class MySPDIFDecoder : public SPDIFDecoder
+    {
+    public:
+        MySPDIFDecoder(SpdifStreamIn *spdifStreamIn, audio_format_t format)
+          :  SPDIFDecoder(format)
+          , mSpdifStreamIn(spdifStreamIn)
+        {
+        }
+
+        ssize_t readInput(void* buffer, size_t bytes) override
+        {
+            size_t bytesRead = 0;
+            const auto result = mSpdifStreamIn->readDataBurst(buffer, bytes, &bytesRead);
+            if (result < 0) {
+                return result;
+            }
+            return bytesRead;
+        }
+
+    protected:
+        SpdifStreamIn * const mSpdifStreamIn;
+    };
+
+    MySPDIFDecoder mSpdifDecoder;
+    audio_config_base_t mApplicationConfig = AUDIO_CONFIG_BASE_INITIALIZER;
+
+    status_t readDataBurst(void* data, size_t bytes, size_t* read);
+
+#ifdef TEE_SINK
+    NBAIO_Tee mTee;
+#endif
+
+};
+
+} // namespace android
diff --git a/services/audioflinger/datapath/SpdifStreamOut.cpp b/services/audioflinger/datapath/SpdifStreamOut.cpp
index 0c6a5a1..65a4eec 100644
--- a/services/audioflinger/datapath/SpdifStreamOut.cpp
+++ b/services/audioflinger/datapath/SpdifStreamOut.cpp
@@ -1,19 +1,19 @@
 /*
-**
-** Copyright 2015, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-**     http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
+ *
+ * Copyright 2015, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *     http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
 
 #define LOG_TAG "AudioFlinger"
 //#define LOG_NDEBUG 0
@@ -42,10 +42,10 @@
 }
 
 status_t SpdifStreamOut::open(
-                              audio_io_handle_t handle,
-                              audio_devices_t devices,
-                              struct audio_config *config,
-                              const char *address)
+        audio_io_handle_t handle,
+        audio_devices_t devices,
+        struct audio_config *config,
+        const char *address)
 {
     struct audio_config customConfig = *config;
 
@@ -53,22 +53,10 @@
     mApplicationConfig.sample_rate = config->sample_rate;
     mApplicationConfig.channel_mask = config->channel_mask;
 
-    // Some data bursts run at a higher sample rate.
-    // TODO Move this into the audio_utils as a static method.
-    switch(config->format) {
-        case AUDIO_FORMAT_E_AC3:
-        case AUDIO_FORMAT_E_AC3_JOC:
-            mRateMultiplier = 4;
-            break;
-        case AUDIO_FORMAT_AC3:
-        case AUDIO_FORMAT_DTS:
-        case AUDIO_FORMAT_DTS_HD:
-            mRateMultiplier = 1;
-            break;
-        default:
-            ALOGE("ERROR SpdifStreamOut::open() unrecognized format 0x%08X\n",
-                config->format);
-            return BAD_VALUE;
+    mRateMultiplier = spdif_rate_multiplier(config->format);
+    if (mRateMultiplier <= 0) {
+        ALOGE("ERROR SpdifStreamOut::open() unrecognized format 0x%08X\n", config->format);
+        return BAD_VALUE;
     }
     customConfig.sample_rate = config->sample_rate * mRateMultiplier;
 
@@ -78,16 +66,10 @@
     // Always print this because otherwise it could be very confusing if the
     // HAL and AudioFlinger are using different formats.
     // Print before open() because HAL may modify customConfig.
-    ALOGI("SpdifStreamOut::open() AudioFlinger requested"
-            " sampleRate %d, format %#x, channelMask %#x",
-            config->sample_rate,
-            config->format,
-            config->channel_mask);
-    ALOGI("SpdifStreamOut::open() HAL configured for"
-            " sampleRate %d, format %#x, channelMask %#x",
-            customConfig.sample_rate,
-            customConfig.format,
-            customConfig.channel_mask);
+    ALOGI("SpdifStreamOut::open() AudioFlinger requested sampleRate %d, format %#x,"
+            " channelMask %#x", config->sample_rate, config->format, config->channel_mask);
+    ALOGI("SpdifStreamOut::open() HAL configured for sampleRate %d, format %#x, channelMask %#x",
+            customConfig.sample_rate, customConfig.format, customConfig.channel_mask);
 
     const status_t status = AudioStreamOut::open(
             handle,
diff --git a/services/audioflinger/datapath/SpdifStreamOut.h b/services/audioflinger/datapath/SpdifStreamOut.h
index 56d57f6..c6d27ba 100644
--- a/services/audioflinger/datapath/SpdifStreamOut.h
+++ b/services/audioflinger/datapath/SpdifStreamOut.h
@@ -1,22 +1,21 @@
 /*
-**
-** Copyright 2015, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-**     http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
+ *
+ * Copyright 2015, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *     http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
 
-#ifndef ANDROID_SPDIF_STREAM_OUT_H
-#define ANDROID_SPDIF_STREAM_OUT_H
+#pragma once
 
 #include <stdint.h>
 #include <sys/types.h>
@@ -40,8 +39,6 @@
     SpdifStreamOut(AudioHwDevice *dev, audio_output_flags_t flags,
             audio_format_t format);
 
-    ~SpdifStreamOut() override = default;
-
     status_t open(
             audio_io_handle_t handle,
             audio_devices_t devices,
@@ -116,10 +113,10 @@
         SpdifStreamOut * const mSpdifStreamOut;
     };
 
-    MySPDIFEncoder       mSpdifEncoder;
-    audio_config_base_t  mApplicationConfig = AUDIO_CONFIG_BASE_INITIALIZER;
+    MySPDIFEncoder mSpdifEncoder;
+    audio_config_base_t mApplicationConfig = AUDIO_CONFIG_BASE_INITIALIZER;
 
-    ssize_t  writeDataBurst(const void* data, size_t bytes);
+    ssize_t writeDataBurst(const void* data, size_t bytes);
 
 #ifdef TEE_SINK
     NBAIO_Tee mTee;
@@ -128,5 +125,3 @@
 };
 
 } // namespace android
-
-#endif // ANDROID_SPDIF_STREAM_OUT_H
diff --git a/services/audioflinger/datapath/ThreadMetrics.h b/services/audioflinger/datapath/ThreadMetrics.h
index c643a57..4eb8aa0 100644
--- a/services/audioflinger/datapath/ThreadMetrics.h
+++ b/services/audioflinger/datapath/ThreadMetrics.h
@@ -14,8 +14,7 @@
  * limitations under the License.
  */
 
-#ifndef ANDROID_AUDIO_THREADMETRICS_H
-#define ANDROID_AUDIO_THREADMETRICS_H
+#pragma once
 
 #include <media/MediaMetricsItem.h>
 
@@ -210,5 +209,3 @@
 };
 
 } // namespace android
-
-#endif // ANDROID_AUDIO_THREADMETRICS_H
diff --git a/services/audioflinger/datapath/TrackMetrics.h b/services/audioflinger/datapath/TrackMetrics.h
index 2b44acb..ad5d3db 100644
--- a/services/audioflinger/datapath/TrackMetrics.h
+++ b/services/audioflinger/datapath/TrackMetrics.h
@@ -14,8 +14,7 @@
  * limitations under the License.
  */
 
-#ifndef ANDROID_AUDIO_TRACKMETRICS_H
-#define ANDROID_AUDIO_TRACKMETRICS_H
+#pragma once
 
 #include <binder/IActivityManager.h>
 #include <binder/IPCThreadState.h>
@@ -306,5 +305,3 @@
 };
 
 } // namespace android
-
-#endif // ANDROID_AUDIO_TRACKMETRICS_H
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index ab3d323..58e5311 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -4401,8 +4401,8 @@
             if (!mAvailableOutputDevices.containsAtLeastOne(curProfile->getSupportedDevices())) {
                 continue;
             }
-            if ((curProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
-                        != AUDIO_OUTPUT_FLAG_NONE) {
+            if (offloadPossible && ((curProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
+                        != AUDIO_OUTPUT_FLAG_NONE)) {
                 if ((directMode & AUDIO_DIRECT_OFFLOAD_GAPLESS_SUPPORTED)
                         != AUDIO_DIRECT_NOT_SUPPORTED) {
                     // Already reports offload gapless supported. No need to report offload support.
diff --git a/services/audiopolicy/service/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
index 7584632..6de71a3 100644
--- a/services/audiopolicy/service/AudioPolicyClientImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
@@ -164,6 +164,8 @@
     status_t status = af->openInput(request, &response);
     if (status == OK) {
         *input = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_module_handle_t(response.input));
+        *config = VALUE_OR_RETURN_STATUS(
+                aidl2legacy_AudioConfig_audio_config_t(response.config, true /*isInput*/));
     }
     return status;
 }