Merge "Move 'session' field to AudioPortExtSys AIDL"
diff --git a/PREUPLOAD.cfg b/PREUPLOAD.cfg
index 716b550..1f7083b 100644
--- a/PREUPLOAD.cfg
+++ b/PREUPLOAD.cfg
@@ -11,3 +11,4 @@
 clang_format = --commit ${PREUPLOAD_COMMIT} --style file --extensions c,h,cc,cpp
                media/libmediatranscoding/
                services/mediatranscoding/
+               media/libaudioclient/tests/
diff --git a/media/libaudioclient/AudioRecord.cpp b/media/libaudioclient/AudioRecord.cpp
index 15203d6..69d73ad 100644
--- a/media/libaudioclient/AudioRecord.cpp
+++ b/media/libaudioclient/AudioRecord.cpp
@@ -146,39 +146,6 @@
         audio_channel_mask_t channelMask,
         const AttributionSourceState& client,
         size_t frameCount,
-        legacy_callback_t callback,
-        void* user,
-        uint32_t notificationFrames,
-        audio_session_t sessionId,
-        transfer_type transferType,
-        audio_input_flags_t flags,
-        const audio_attributes_t* pAttributes,
-        audio_port_handle_t selectedDeviceId,
-        audio_microphone_direction_t selectedMicDirection,
-        float microphoneFieldDimension)
-    : mActive(false),
-      mStatus(NO_INIT),
-      mClientAttributionSource(client),
-      mSessionId(AUDIO_SESSION_ALLOCATE),
-      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
-      mPreviousSchedulingGroup(SP_DEFAULT),
-      mProxy(NULL)
-{
-    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(mClientAttributionSource.uid));
-    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid));
-    (void)set(inputSource, sampleRate, format, channelMask, frameCount, callback, user,
-            notificationFrames, false /*threadCanCallJava*/, sessionId, transferType, flags,
-            uid, pid, pAttributes, selectedDeviceId, selectedMicDirection,
-            microphoneFieldDimension);
-}
-
-AudioRecord::AudioRecord(
-        audio_source_t inputSource,
-        uint32_t sampleRate,
-        audio_format_t format,
-        audio_channel_mask_t channelMask,
-        const AttributionSourceState& client,
-        size_t frameCount,
         const wp<IAudioRecordCallback>& callback,
         uint32_t notificationFrames,
         audio_session_t sessionId,
@@ -255,37 +222,6 @@
         mDeviceCallback.clear();
     }
 }
-namespace {
-class LegacyCallbackWrapper : public AudioRecord::IAudioRecordCallback {
-    const AudioRecord::legacy_callback_t mCallback;
-    void* const mData;
-
-  public:
-    LegacyCallbackWrapper(AudioRecord::legacy_callback_t callback, void* user)
-        : mCallback(callback), mData(user) {}
-
-    size_t onMoreData(const AudioRecord::Buffer& buffer) override {
-        AudioRecord::Buffer copy = buffer;
-        mCallback(AudioRecord::EVENT_MORE_DATA, mData, &copy);
-        return copy.size();
-    }
-
-    void onOverrun() override { mCallback(AudioRecord::EVENT_OVERRUN, mData, nullptr); }
-
-    void onMarker(uint32_t markerPosition) override {
-        mCallback(AudioRecord::EVENT_MARKER, mData, &markerPosition);
-    }
-
-    void onNewPos(uint32_t newPos) override {
-        mCallback(AudioRecord::EVENT_NEW_POS, mData, &newPos);
-    }
-
-    void onNewIAudioRecord() override {
-        mCallback(AudioRecord::EVENT_NEW_IAUDIORECORD, mData, nullptr);
-    }
-};
-}  // namespace
-
 status_t AudioRecord::set(
         audio_source_t inputSource,
         uint32_t sampleRate,
@@ -479,37 +415,6 @@
     return status;
 }
 
-status_t AudioRecord::set(
-        audio_source_t inputSource,
-        uint32_t sampleRate,
-        audio_format_t format,
-        audio_channel_mask_t channelMask,
-        size_t frameCount,
-        legacy_callback_t callback,
-        void* user,
-        uint32_t notificationFrames,
-        bool threadCanCallJava,
-        audio_session_t sessionId,
-        transfer_type transferType,
-        audio_input_flags_t flags,
-        uid_t uid,
-        pid_t pid,
-        const audio_attributes_t* pAttributes,
-        audio_port_handle_t selectedDeviceId,
-        audio_microphone_direction_t selectedMicDirection,
-        float microphoneFieldDimension,
-        int32_t maxSharedAudioHistoryMs)
-{
-    if (callback != nullptr) {
-        mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
-    } else if (user) {
-        LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
-    }
-    return set(inputSource, sampleRate, format, channelMask, frameCount, mLegacyCallbackWrapper,
-        notificationFrames, threadCanCallJava, sessionId, transferType, flags, uid, pid,
-        pAttributes, selectedDeviceId, selectedMicDirection, microphoneFieldDimension,
-        maxSharedAudioHistoryMs);
-}
 // -------------------------------------------------------------------------
 
 status_t AudioRecord::start(AudioSystem::sync_event_t event, audio_session_t triggerSession)
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 6deef8f..96fc544 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -325,45 +325,6 @@
         }
     };
 }
-
-AudioTrack::AudioTrack(
-        audio_stream_type_t streamType,
-        uint32_t sampleRate,
-        audio_format_t format,
-        audio_channel_mask_t channelMask,
-        size_t frameCount,
-        audio_output_flags_t flags,
-        legacy_callback_t callback,
-        void* user,
-        int32_t notificationFrames,
-        audio_session_t sessionId,
-        transfer_type transferType,
-        const audio_offload_info_t *offloadInfo,
-        const AttributionSourceState& attributionSource,
-        const audio_attributes_t* pAttributes,
-        bool doNotReconnect,
-        float maxRequiredSpeed,
-        audio_port_handle_t selectedDeviceId)
-    : mStatus(NO_INIT),
-      mState(STATE_STOPPED),
-      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
-      mPreviousSchedulingGroup(SP_DEFAULT),
-      mPausedPosition(0),
-      mAudioTrackCallback(new AudioTrackCallback())
-{
-    mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
-    if (callback != nullptr) {
-        mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
-    } else if (user) {
-        LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
-    }
-    mSetParams = std::unique_ptr<SetParams>{new SetParams{
-            streamType, sampleRate, format, channelMask, frameCount, flags, mLegacyCallbackWrapper,
-            notificationFrames, 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId,
-            transferType, offloadInfo, attributionSource, pAttributes, doNotReconnect,
-            maxRequiredSpeed, selectedDeviceId}};
-}
-
 AudioTrack::AudioTrack(
         audio_stream_type_t streamType,
         uint32_t sampleRate,
@@ -397,44 +358,6 @@
                           doNotReconnect, maxRequiredSpeed, AUDIO_PORT_HANDLE_NONE}};
 }
 
-AudioTrack::AudioTrack(
-        audio_stream_type_t streamType,
-        uint32_t sampleRate,
-        audio_format_t format,
-        audio_channel_mask_t channelMask,
-        const sp<IMemory>& sharedBuffer,
-        audio_output_flags_t flags,
-        legacy_callback_t callback,
-        void* user,
-        int32_t notificationFrames,
-        audio_session_t sessionId,
-        transfer_type transferType,
-        const audio_offload_info_t *offloadInfo,
-        const AttributionSourceState& attributionSource,
-        const audio_attributes_t* pAttributes,
-        bool doNotReconnect,
-        float maxRequiredSpeed)
-    : mStatus(NO_INIT),
-      mState(STATE_STOPPED),
-      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
-      mPreviousSchedulingGroup(SP_DEFAULT),
-      mPausedPosition(0),
-      mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
-      mAudioTrackCallback(new AudioTrackCallback())
-{
-    mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
-    if (callback) {
-        mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
-    } else if (user) {
-        LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
-    }
-    mSetParams = std::unique_ptr<SetParams>{new SetParams{
-            streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags,
-            mLegacyCallbackWrapper, notificationFrames, sharedBuffer, false /*threadCanCallJava*/,
-            sessionId, transferType, offloadInfo, attributionSource, pAttributes, doNotReconnect,
-            maxRequiredSpeed, AUDIO_PORT_HANDLE_NONE}};
-}
-
 void AudioTrack::onFirstRef() {
     if (mSetParams) {
         set(*mSetParams);
@@ -496,38 +419,6 @@
         mDeviceCallback.clear();
     }
 }
-
-status_t AudioTrack::set(
-        audio_stream_type_t streamType,
-        uint32_t sampleRate,
-        audio_format_t format,
-        audio_channel_mask_t channelMask,
-        size_t frameCount,
-        audio_output_flags_t flags,
-        legacy_callback_t callback,
-        void * user,
-        int32_t notificationFrames,
-        const sp<IMemory>& sharedBuffer,
-        bool threadCanCallJava,
-        audio_session_t sessionId,
-        transfer_type transferType,
-        const audio_offload_info_t *offloadInfo,
-        const AttributionSourceState& attributionSource,
-        const audio_attributes_t* pAttributes,
-        bool doNotReconnect,
-        float maxRequiredSpeed,
-        audio_port_handle_t selectedDeviceId)
-{
-    if (callback) {
-        mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
-    } else if (user) {
-        LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
-    }
-    return set(streamType, sampleRate,format, channelMask, frameCount, flags,
-               mLegacyCallbackWrapper, notificationFrames, sharedBuffer, threadCanCallJava,
-               sessionId, transferType, offloadInfo, attributionSource, pAttributes,
-               doNotReconnect, maxRequiredSpeed, selectedDeviceId);
-}
 status_t AudioTrack::set(
         audio_stream_type_t streamType,
         uint32_t sampleRate,
diff --git a/media/libaudioclient/TEST_MAPPING b/media/libaudioclient/TEST_MAPPING
index 3751f80..d36cf10 100644
--- a/media/libaudioclient/TEST_MAPPING
+++ b/media/libaudioclient/TEST_MAPPING
@@ -4,6 +4,9 @@
       "name": "audio_aidl_conversion_tests"
     },
     {
+      "name": "audio_aidl_status_tests"
+    },
+    {
       "name": "CtsNativeMediaAAudioTestCases",
       "options" : [
         {
@@ -11,5 +14,22 @@
         }
       ]
     }
+  ],
+  "postsubmit": [
+    {
+      "name": "audieorecord_tests"
+    },
+    {
+      "name": "audioeffect_tests"
+    },
+    {
+      "name": "audiorouting_tests"
+    },
+    {
+      "name": "audioclient_serialization_tests"
+    },
+    {
+      "name": "trackplayerbase_tests"
+    }
   ]
 }
diff --git a/media/libaudioclient/include/media/AudioRecord.h b/media/libaudioclient/include/media/AudioRecord.h
index cb05dd9..5a1ff65 100644
--- a/media/libaudioclient/include/media/AudioRecord.h
+++ b/media/libaudioclient/include/media/AudioRecord.h
@@ -46,27 +46,6 @@
 {
 public:
 
-    /* Events used by AudioRecord callback function (legacy_callback_t).
-     * Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*.
-     */
-    enum event_type {
-        EVENT_MORE_DATA = 0,        // Request to read available data from buffer.
-                                    // If this event is delivered but the callback handler
-                                    // does not want to read the available data, the handler must
-                                    // explicitly ignore the event by setting frameCount to zero.
-        EVENT_OVERRUN = 1,          // Buffer overrun occurred.
-        EVENT_MARKER = 2,           // Record head is at the specified marker position
-                                    // (See setMarkerPosition()).
-        EVENT_NEW_POS = 3,          // Record head is at a new position
-                                    // (See setPositionUpdatePeriod()).
-        EVENT_NEW_IAUDIORECORD = 4, // IAudioRecord was re-created, either due to re-routing and
-                                    // voluntary invalidation by mediaserver, or mediaserver crash.
-    };
-
-    /* Client should declare a Buffer and pass address to obtainBuffer()
-     * and releaseBuffer().  See also legacy_callback_t for EVENT_MORE_DATA.
-     */
-
     class Buffer
     {
       friend AudioRecord;
@@ -122,7 +101,6 @@
      *          - EVENT_NEW_IAUDIORECORD: unused.
      */
 
-    typedef void (*legacy_callback_t)(int event, void* user, void *info);
 
     class IAudioRecordCallback : public virtual RefBase {
         friend AudioRecord;
@@ -226,24 +204,6 @@
                                     float selectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT);
 
 
-                        AudioRecord(audio_source_t inputSource,
-                                    uint32_t sampleRate,
-                                    audio_format_t format,
-                                    audio_channel_mask_t channelMask,
-                                    const android::content::AttributionSourceState& client,
-                                    size_t frameCount,
-                                    legacy_callback_t callback,
-                                    void* user,
-                                    uint32_t notificationFrames = 0,
-                                    audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
-                                    transfer_type transferType = TRANSFER_DEFAULT,
-                                    audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
-                                    const audio_attributes_t* pAttributes = nullptr,
-                                    audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE,
-                                    audio_microphone_direction_t
-                                        selectedMicDirection = MIC_DIRECTION_UNSPECIFIED,
-                                    float selectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT);
-
     /* Terminates the AudioRecord and unregisters it from AudioFlinger.
      * Also destroys all resources associated with the AudioRecord.
      */
@@ -286,27 +246,6 @@
                             float selectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT,
                             int32_t maxSharedAudioHistoryMs = 0);
 
-           status_t    set(audio_source_t inputSource,
-                            uint32_t sampleRate,
-                            audio_format_t format,
-                            audio_channel_mask_t channelMask,
-                            size_t frameCount,
-                            legacy_callback_t callback,
-                            void* user,
-                            uint32_t notificationFrames = 0,
-                            bool threadCanCallJava = false,
-                            audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
-                            transfer_type transferType = TRANSFER_DEFAULT,
-                            audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
-                            uid_t uid = AUDIO_UID_INVALID,
-                            pid_t pid = -1,
-                            const audio_attributes_t* pAttributes = nullptr,
-                            audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE,
-                            audio_microphone_direction_t
-                                selectedMicDirection = MIC_DIRECTION_UNSPECIFIED,
-                            float selectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT,
-                            int32_t maxSharedAudioHistoryMs = 0);
-
     /* Result of constructing the AudioRecord. This must be checked for successful initialization
      * before using any AudioRecord API (except for set()), because using
      * an uninitialized AudioRecord produces undefined results.
diff --git a/media/libaudioclient/include/media/AudioTrack.h b/media/libaudioclient/include/media/AudioTrack.h
index 9f540e6..b6ee483 100644
--- a/media/libaudioclient/include/media/AudioTrack.h
+++ b/media/libaudioclient/include/media/AudioTrack.h
@@ -148,7 +148,6 @@
      *          - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp.
      */
 
-    typedef void (*legacy_callback_t)(int event, void* user, void* info);
     class IAudioTrackCallback : public virtual RefBase {
       friend AudioTrack;
       protected:
@@ -343,26 +342,6 @@
                                     float maxRequiredSpeed = 1.0f,
                                     audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE);
 
-
-                        AudioTrack( audio_stream_type_t streamType,
-                                    uint32_t sampleRate,
-                                    audio_format_t format,
-                                    audio_channel_mask_t channelMask,
-                                    size_t frameCount,
-                                    audio_output_flags_t flags,
-                                    legacy_callback_t cbf,
-                                    void* user = nullptr,
-                                    int32_t notificationFrames = 0,
-                                    audio_session_t sessionId  = AUDIO_SESSION_ALLOCATE,
-                                    transfer_type transferType = TRANSFER_DEFAULT,
-                                    const audio_offload_info_t *offloadInfo = nullptr,
-                                    const AttributionSourceState& attributionSource =
-                                        AttributionSourceState(),
-                                    const audio_attributes_t* pAttributes = nullptr,
-                                    bool doNotReconnect = false,
-                                    float maxRequiredSpeed = 1.0f,
-                                    audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE);
-
     /* Creates an audio track and registers it with AudioFlinger.
      * With this constructor, the track is configured for static buffer mode.
      * Data to be rendered is passed in a shared memory buffer
@@ -391,25 +370,6 @@
                                     bool doNotReconnect = false,
                                     float maxRequiredSpeed = 1.0f);
 
-
-                        AudioTrack( audio_stream_type_t streamType,
-                                    uint32_t sampleRate,
-                                    audio_format_t format,
-                                    audio_channel_mask_t channelMask,
-                                    const sp<IMemory>& sharedBuffer,
-                                    audio_output_flags_t flags,
-                                    legacy_callback_t cbf,
-                                    void* user          = nullptr,
-                                    int32_t notificationFrames = 0,
-                                    audio_session_t sessionId   = AUDIO_SESSION_ALLOCATE,
-                                    transfer_type transferType = TRANSFER_DEFAULT,
-                                    const audio_offload_info_t *offloadInfo = nullptr,
-                                    const AttributionSourceState& attributionSource =
-                                        AttributionSourceState(),
-                                    const audio_attributes_t* pAttributes = nullptr,
-                                    bool doNotReconnect = false,
-                                    float maxRequiredSpeed = 1.0f);
-
     /* Terminates the AudioTrack and unregisters it from AudioFlinger.
      * Also destroys all resources associated with the AudioTrack.
      */
@@ -490,28 +450,8 @@
                         }
             void       onFirstRef() override;
         public:
-            status_t    set(audio_stream_type_t streamType,
-                            uint32_t sampleRate,
-                            audio_format_t format,
-                            audio_channel_mask_t channelMask,
-                            size_t frameCount,
-                            audio_output_flags_t flags,
-                            legacy_callback_t callback,
-                            void * user = nullptr,
-                            int32_t notificationFrames = 0,
-                            const sp<IMemory>& sharedBuffer = 0,
-                            bool threadCanCallJava = false,
-                            audio_session_t sessionId  = AUDIO_SESSION_ALLOCATE,
-                            transfer_type transferType = TRANSFER_DEFAULT,
-                            const audio_offload_info_t *offloadInfo = nullptr,
-                            const AttributionSourceState& attributionSource =
-                                AttributionSourceState(),
-                            const audio_attributes_t* pAttributes = nullptr,
-                            bool doNotReconnect = false,
-                            float maxRequiredSpeed = 1.0f,
-                            audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE);
-
-    // FIXME(b/169889714): Vendor code depends on the old method signature at link time
+            typedef void (*legacy_callback_t)(int event, void* user, void* info);
+            // FIXME(b/169889714): Vendor code depends on the old method signature at link time
             status_t    set(audio_stream_type_t streamType,
                             uint32_t sampleRate,
                             audio_format_t format,
diff --git a/media/libaudioclient/tests/Android.bp b/media/libaudioclient/tests/Android.bp
index 891293e..6535b5b 100644
--- a/media/libaudioclient/tests/Android.bp
+++ b/media/libaudioclient/tests/Android.bp
@@ -93,3 +93,107 @@
     ],
     data: ["record_test_input_*.txt"],
 }
+
+cc_defaults {
+    name: "libaudioclient_gtests_defaults",
+    cflags: [
+        "-Wall",
+        "-Werror",
+    ],
+    shared_libs: [
+        "capture_state_listener-aidl-cpp",
+        "framework-permission-aidl-cpp",
+        "libbase",
+        "libbinder",
+        "libcgrouprc",
+        "libcutils",
+        "libdl",
+        "liblog",
+        "libmedia",
+        "libmediametrics",
+        "libmediautils",
+        "libmedia_helper",
+        "libnblog",
+        "libprocessgroup",
+        "libshmemcompat",
+        "libstagefright_foundation",
+        "libutils",
+        "libvibrator",
+        "mediametricsservice-aidl-cpp",
+        "packagemanager_aidl-cpp",
+        "shared-file-region-aidl-cpp",
+    ],
+    static_libs: [
+        "android.hardware.audio.common@7.0-enums",
+        "android.media.audio.common.types-V1-cpp",
+        "audioclient-types-aidl-cpp",
+        "audioflinger-aidl-cpp",
+        "audiopolicy-aidl-cpp",
+        "audiopolicy-types-aidl-cpp",
+        "av-types-aidl-cpp",
+        "effect-aidl-cpp",
+        "libaudioclient",
+        "libaudioclient_aidl_conversion",
+        "libaudiofoundation",
+        "libaudiomanager",
+        "libaudiopolicy",
+        "libaudioutils",
+    ],
+    data: ["bbb*.raw"],
+    test_config_template: "audio_test_template.xml",
+    test_suites: ["device-tests"],
+}
+
+cc_test {
+    name: "audiorecord_tests",
+    defaults: ["libaudioclient_gtests_defaults"],
+    srcs: [
+        "audiorecord_tests.cpp",
+        "audio_test_utils.cpp",
+    ],
+}
+
+cc_test {
+    name: "audiotrack_tests",
+    defaults: ["libaudioclient_gtests_defaults"],
+    srcs: [
+        "audiotrack_tests.cpp",
+        "audio_test_utils.cpp",
+    ],
+}
+
+cc_test {
+    name: "audioeffect_tests",
+    defaults: ["libaudioclient_gtests_defaults"],
+    srcs: [
+        "audioeffect_tests.cpp",
+        "audio_test_utils.cpp",
+    ],
+}
+
+cc_test {
+    name: "audiorouting_tests",
+    defaults: ["libaudioclient_gtests_defaults"],
+    srcs: [
+        "audiorouting_tests.cpp",
+        "audio_test_utils.cpp",
+    ],
+    shared_libs: [
+        "libxml2",
+    ],
+}
+
+cc_test {
+    name: "audioclient_serialization_tests",
+    defaults: ["libaudioclient_gtests_defaults"],
+    srcs: [
+        "audioclient_serialization_tests.cpp",
+        "audio_test_utils.cpp",
+    ],
+}
+
+cc_test {
+    name: "trackplayerbase_tests",
+    defaults: ["libaudioclient_gtests_defaults"],
+    srcs: ["trackplayerbase_tests.cpp"],
+}
diff --git a/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp b/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp
index 997f62a..9e663bc 100644
--- a/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp
+++ b/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp
@@ -16,22 +16,29 @@
 
 #include <gtest/gtest.h>
 
-#include <media/AudioCommonTypes.h>
 #include <media/AidlConversion.h>
+#include <media/AudioCommonTypes.h>
 
 using namespace android;
 using namespace android::aidl_utils;
 
+using android::media::AudioDirectMode;
 using media::audio::common::AudioChannelLayout;
 using media::audio::common::AudioDeviceDescription;
 using media::audio::common::AudioDeviceType;
+using media::audio::common::AudioEncapsulationMetadataType;
+using media::audio::common::AudioEncapsulationType;
 using media::audio::common::AudioFormatDescription;
 using media::audio::common::AudioFormatType;
+using media::audio::common::AudioGainMode;
+using media::audio::common::AudioStandard;
+using media::audio::common::ExtraAudioDescriptor;
 using media::audio::common::PcmType;
 
 namespace {
 
-template<typename T> size_t hash(const T& t) {
+template <typename T>
+size_t hash(const T& t) {
     return std::hash<T>{}(t);
 }
 
@@ -52,10 +59,8 @@
     return AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
             // Use channels that exist both for input and output,
             // but doesn't form a known layout mask.
-            AudioChannelLayout::CHANNEL_FRONT_LEFT |
-            AudioChannelLayout::CHANNEL_FRONT_RIGHT |
-            AudioChannelLayout::CHANNEL_TOP_SIDE_LEFT |
-            AudioChannelLayout::CHANNEL_TOP_SIDE_RIGHT);
+            AudioChannelLayout::CHANNEL_FRONT_LEFT | AudioChannelLayout::CHANNEL_FRONT_RIGHT |
+            AudioChannelLayout::CHANNEL_TOP_SIDE_LEFT | AudioChannelLayout::CHANNEL_TOP_SIDE_RIGHT);
 }
 
 AudioChannelLayout make_ACL_ChannelIndex2() {
@@ -74,7 +79,7 @@
 }
 
 AudioDeviceDescription make_AudioDeviceDescription(AudioDeviceType type,
-        const std::string& connection = "") {
+                                                   const std::string& connection = "") {
     AudioDeviceDescription result;
     result.type = type;
     result.connection = connection;
@@ -95,12 +100,12 @@
 
 AudioDeviceDescription make_ADD_WiredHeadset() {
     return make_AudioDeviceDescription(AudioDeviceType::OUT_HEADSET,
-            AudioDeviceDescription::CONNECTION_ANALOG());
+                                       AudioDeviceDescription::CONNECTION_ANALOG());
 }
 
 AudioDeviceDescription make_ADD_BtScoHeadset() {
     return make_AudioDeviceDescription(AudioDeviceType::OUT_HEADSET,
-            AudioDeviceDescription::CONNECTION_BT_SCO());
+                                       AudioDeviceDescription::CONNECTION_BT_SCO());
 }
 
 AudioFormatDescription make_AudioFormatDescription(AudioFormatType type) {
@@ -121,8 +126,7 @@
     return result;
 }
 
-AudioFormatDescription make_AudioFormatDescription(PcmType transport,
-        const std::string& encoding) {
+AudioFormatDescription make_AudioFormatDescription(PcmType transport, const std::string& encoding) {
     auto result = make_AudioFormatDescription(encoding);
     result.pcm = transport;
     return result;
@@ -154,6 +158,22 @@
     return afd;
 }
 
+android::media::TrackSecondaryOutputInfo make_TrackSecondaryOutputInfo() {
+    android::media::TrackSecondaryOutputInfo result;
+    result.portId = 1;
+    result.secondaryOutputIds = {0, 5, 7};
+    return result;
+}
+
+ExtraAudioDescriptor make_ExtraAudioDescriptor(AudioStandard audioStandard,
+                                               AudioEncapsulationType audioEncapsulationType) {
+    ExtraAudioDescriptor result;
+    result.standard = audioStandard;
+    result.audioDescriptor = {0xb4, 0xaf, 0x98, 0x1a};
+    result.encapsulationType = audioEncapsulationType;
+    return result;
+}
+
 }  // namespace
 
 // Verify that two independently constructed ADDs/AFDs have the same hash.
@@ -163,7 +183,8 @@
 // is identical to the same format description constructed by the framework.
 class HashIdentityTest : public ::testing::Test {
   public:
-    template<typename T> void verifyHashIdentity(const std::vector<std::function<T()>>& valueGens) {
+    template <typename T>
+    void verifyHashIdentity(const std::vector<std::function<T()>>& valueGens) {
         for (size_t i = 0; i < valueGens.size(); ++i) {
             for (size_t j = 0; j < valueGens.size(); ++j) {
                 if (i == j) {
@@ -177,27 +198,25 @@
 };
 
 TEST_F(HashIdentityTest, AudioChannelLayoutHashIdentity) {
-    verifyHashIdentity<AudioChannelLayout>({
-            make_ACL_None, make_ACL_Invalid, make_ACL_Stereo,
-            make_ACL_LayoutArbitrary, make_ACL_ChannelIndex2,
-            make_ACL_ChannelIndexArbitrary, make_ACL_VoiceCall});
+    verifyHashIdentity<AudioChannelLayout>({make_ACL_None, make_ACL_Invalid, make_ACL_Stereo,
+                                            make_ACL_LayoutArbitrary, make_ACL_ChannelIndex2,
+                                            make_ACL_ChannelIndexArbitrary, make_ACL_VoiceCall});
 }
 
 TEST_F(HashIdentityTest, AudioDeviceDescriptionHashIdentity) {
-    verifyHashIdentity<AudioDeviceDescription>({
-            make_ADD_None, make_ADD_DefaultIn, make_ADD_DefaultOut, make_ADD_WiredHeadset,
-            make_ADD_BtScoHeadset});
+    verifyHashIdentity<AudioDeviceDescription>({make_ADD_None, make_ADD_DefaultIn,
+                                                make_ADD_DefaultOut, make_ADD_WiredHeadset,
+                                                make_ADD_BtScoHeadset});
 }
 
 TEST_F(HashIdentityTest, AudioFormatDescriptionHashIdentity) {
-    verifyHashIdentity<AudioFormatDescription>({
-            make_AFD_Default, make_AFD_Invalid, make_AFD_Pcm16Bit, make_AFD_Bitstream,
-            make_AFD_Encap, make_AFD_Encap_with_Enc});
+    verifyHashIdentity<AudioFormatDescription>({make_AFD_Default, make_AFD_Invalid,
+                                                make_AFD_Pcm16Bit, make_AFD_Bitstream,
+                                                make_AFD_Encap, make_AFD_Encap_with_Enc});
 }
 
 using ChannelLayoutParam = std::tuple<AudioChannelLayout, bool /*isInput*/>;
-class AudioChannelLayoutRoundTripTest :
-        public testing::TestWithParam<ChannelLayoutParam> {};
+class AudioChannelLayoutRoundTripTest : public testing::TestWithParam<ChannelLayoutParam> {};
 TEST_P(AudioChannelLayoutRoundTripTest, Aidl2Legacy2Aidl) {
     const auto initial = std::get<0>(GetParam());
     const bool isInput = std::get<1>(GetParam());
@@ -207,21 +226,82 @@
     ASSERT_TRUE(convBack.ok());
     EXPECT_EQ(initial, convBack.value());
 }
-INSTANTIATE_TEST_SUITE_P(AudioChannelLayoutRoundTrip,
-        AudioChannelLayoutRoundTripTest,
+
+INSTANTIATE_TEST_SUITE_P(
+        AudioChannelLayoutRoundTrip, AudioChannelLayoutRoundTripTest,
         testing::Combine(
                 testing::Values(AudioChannelLayout{}, make_ACL_Invalid(), make_ACL_Stereo(),
-                        make_ACL_LayoutArbitrary(), make_ACL_ChannelIndex2(),
-                        make_ACL_ChannelIndexArbitrary()),
+                                make_ACL_LayoutArbitrary(), make_ACL_ChannelIndex2(),
+                                make_ACL_ChannelIndexArbitrary(),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_FRONT_LEFT),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_FRONT_RIGHT),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_BACK_CENTER),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_BACK_LEFT),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_BACK_RIGHT),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_FRONT_CENTER),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_LOW_FREQUENCY),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_TOP_SIDE_LEFT),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_TOP_SIDE_RIGHT)),
                 testing::Values(false, true)));
-INSTANTIATE_TEST_SUITE_P(AudioChannelVoiceRoundTrip,
-        AudioChannelLayoutRoundTripTest,
-        // In legacy constants the voice call is only defined for input.
-        testing::Combine(testing::Values(make_ACL_VoiceCall()), testing::Values(true)));
+INSTANTIATE_TEST_SUITE_P(AudioChannelVoiceRoundTrip, AudioChannelLayoutRoundTripTest,
+                         // In legacy constants the voice call is only defined for input.
+                         testing::Combine(testing::Values(make_ACL_VoiceCall()),
+                                          testing::Values(true)));
+
+INSTANTIATE_TEST_SUITE_P(
+        OutAudioChannelLayoutLayoutRoundTrip, AudioChannelLayoutRoundTripTest,
+        testing::Combine(
+                testing::Values(AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_FRONT_LEFT_OF_CENTER),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_FRONT_RIGHT_OF_CENTER),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_SIDE_LEFT),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_SIDE_RIGHT),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_TOP_CENTER),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_TOP_FRONT_LEFT),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_TOP_FRONT_CENTER),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_TOP_FRONT_RIGHT),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_TOP_BACK_LEFT),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_TOP_BACK_CENTER),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_TOP_BACK_RIGHT),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_BOTTOM_FRONT_LEFT),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_BOTTOM_FRONT_CENTER),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_BOTTOM_FRONT_RIGHT),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_LOW_FREQUENCY_2),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_FRONT_WIDE_LEFT),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_FRONT_WIDE_RIGHT),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_HAPTIC_A),
+                                AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+                                        AudioChannelLayout::CHANNEL_HAPTIC_B)),
+                testing::Values(false)));
 
 using ChannelLayoutEdgeCaseParam = std::tuple<int /*legacy*/, bool /*isInput*/, bool /*isValid*/>;
-class AudioChannelLayoutEdgeCaseTest :
-        public testing::TestWithParam<ChannelLayoutEdgeCaseParam> {};
+class AudioChannelLayoutEdgeCaseTest : public testing::TestWithParam<ChannelLayoutEdgeCaseParam> {};
 TEST_P(AudioChannelLayoutEdgeCaseTest, Legacy2Aidl) {
     const audio_channel_mask_t legacy = static_cast<audio_channel_mask_t>(std::get<0>(GetParam()));
     const bool isInput = std::get<1>(GetParam());
@@ -229,8 +309,8 @@
     auto conv = legacy2aidl_audio_channel_mask_t_AudioChannelLayout(legacy, isInput);
     EXPECT_EQ(isValid, conv.ok());
 }
-INSTANTIATE_TEST_SUITE_P(AudioChannelLayoutEdgeCase,
-        AudioChannelLayoutEdgeCaseTest,
+INSTANTIATE_TEST_SUITE_P(
+        AudioChannelLayoutEdgeCase, AudioChannelLayoutEdgeCaseTest,
         testing::Values(
                 // Valid legacy input masks.
                 std::make_tuple(AUDIO_CHANNEL_IN_VOICE_UPLINK_MONO, true, true),
@@ -240,25 +320,26 @@
                 std::make_tuple(
                         // This has the same numerical representation as Mask 'A' below
                         AUDIO_CHANNEL_OUT_FRONT_CENTER | AUDIO_CHANNEL_OUT_LOW_FREQUENCY |
-                        AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT, false, true),
+                                AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT,
+                        false, true),
                 std::make_tuple(
                         // This has the same numerical representation as Mask 'B' below
                         AUDIO_CHANNEL_OUT_FRONT_CENTER | AUDIO_CHANNEL_OUT_LOW_FREQUENCY |
-                        AUDIO_CHANNEL_OUT_TOP_BACK_LEFT, false, true),
+                                AUDIO_CHANNEL_OUT_TOP_BACK_LEFT,
+                        false, true),
                 // Invalid legacy input masks.
                 std::make_tuple(AUDIO_CHANNEL_IN_6, true, false),
-                std::make_tuple(
-                        AUDIO_CHANNEL_IN_6 | AUDIO_CHANNEL_IN_FRONT_PROCESSED, true, false),
-                std::make_tuple(
-                        AUDIO_CHANNEL_IN_PRESSURE | AUDIO_CHANNEL_IN_X_AXIS |
-                        AUDIO_CHANNEL_IN_Y_AXIS | AUDIO_CHANNEL_IN_Z_AXIS, true, false),
+                std::make_tuple(AUDIO_CHANNEL_IN_6 | AUDIO_CHANNEL_IN_FRONT_PROCESSED, true, false),
+                std::make_tuple(AUDIO_CHANNEL_IN_PRESSURE | AUDIO_CHANNEL_IN_X_AXIS |
+                                        AUDIO_CHANNEL_IN_Y_AXIS | AUDIO_CHANNEL_IN_Z_AXIS,
+                                true, false),
                 std::make_tuple(  // Mask 'A'
                         AUDIO_CHANNEL_IN_STEREO | AUDIO_CHANNEL_IN_VOICE_UPLINK, true, false),
                 std::make_tuple(  // Mask 'B'
                         AUDIO_CHANNEL_IN_STEREO | AUDIO_CHANNEL_IN_VOICE_DNLINK, true, false)));
 
-class AudioDeviceDescriptionRoundTripTest :
-        public testing::TestWithParam<AudioDeviceDescription> {};
+class AudioDeviceDescriptionRoundTripTest : public testing::TestWithParam<AudioDeviceDescription> {
+};
 TEST_P(AudioDeviceDescriptionRoundTripTest, Aidl2Legacy2Aidl) {
     const auto initial = GetParam();
     auto conv = aidl2legacy_AudioDeviceDescription_audio_devices_t(initial);
@@ -267,13 +348,13 @@
     ASSERT_TRUE(convBack.ok());
     EXPECT_EQ(initial, convBack.value());
 }
-INSTANTIATE_TEST_SUITE_P(AudioDeviceDescriptionRoundTrip,
-        AudioDeviceDescriptionRoundTripTest,
-        testing::Values(AudioDeviceDescription{}, make_ADD_DefaultIn(),
-                make_ADD_DefaultOut(), make_ADD_WiredHeadset(), make_ADD_BtScoHeadset()));
+INSTANTIATE_TEST_SUITE_P(AudioDeviceDescriptionRoundTrip, AudioDeviceDescriptionRoundTripTest,
+                         testing::Values(AudioDeviceDescription{}, make_ADD_DefaultIn(),
+                                         make_ADD_DefaultOut(), make_ADD_WiredHeadset(),
+                                         make_ADD_BtScoHeadset()));
 
-class AudioFormatDescriptionRoundTripTest :
-        public testing::TestWithParam<AudioFormatDescription> {};
+class AudioFormatDescriptionRoundTripTest : public testing::TestWithParam<AudioFormatDescription> {
+};
 TEST_P(AudioFormatDescriptionRoundTripTest, Aidl2Legacy2Aidl) {
     const auto initial = GetParam();
     auto conv = aidl2legacy_AudioFormatDescription_audio_format_t(initial);
@@ -282,6 +363,140 @@
     ASSERT_TRUE(convBack.ok());
     EXPECT_EQ(initial, convBack.value());
 }
-INSTANTIATE_TEST_SUITE_P(AudioFormatDescriptionRoundTrip,
-        AudioFormatDescriptionRoundTripTest,
-        testing::Values(make_AFD_Invalid(), AudioFormatDescription{}, make_AFD_Pcm16Bit()));
+INSTANTIATE_TEST_SUITE_P(AudioFormatDescriptionRoundTrip, AudioFormatDescriptionRoundTripTest,
+                         testing::Values(make_AFD_Invalid(), AudioFormatDescription{},
+                                         make_AFD_Pcm16Bit()));
+
+class AudioDirectModeRoundTripTest : public testing::TestWithParam<AudioDirectMode> {};
+TEST_P(AudioDirectModeRoundTripTest, Aidl2Legacy2Aidl) {
+    const auto initial = GetParam();
+    auto conv = aidl2legacy_AudioDirectMode_audio_direct_mode_t(initial);
+    ASSERT_TRUE(conv.ok());
+    auto convBack = legacy2aidl_audio_direct_mode_t_AudioDirectMode(conv.value());
+    ASSERT_TRUE(convBack.ok());
+    EXPECT_EQ(initial, convBack.value());
+}
+INSTANTIATE_TEST_SUITE_P(AudioDirectMode, AudioDirectModeRoundTripTest,
+                         testing::Values(AudioDirectMode::NONE, AudioDirectMode::OFFLOAD,
+                                         AudioDirectMode::OFFLOAD_GAPLESS,
+                                         AudioDirectMode::BITSTREAM));
+
+class AudioStandardRoundTripTest : public testing::TestWithParam<AudioStandard> {};
+TEST_P(AudioStandardRoundTripTest, Aidl2Legacy2Aidl) {
+    const auto initial = GetParam();
+    auto conv = aidl2legacy_AudioStandard_audio_standard_t(initial);
+    ASSERT_TRUE(conv.ok());
+    auto convBack = legacy2aidl_audio_standard_t_AudioStandard(conv.value());
+    ASSERT_TRUE(convBack.ok());
+    EXPECT_EQ(initial, convBack.value());
+}
+INSTANTIATE_TEST_SUITE_P(AudioStandard, AudioStandardRoundTripTest,
+                         testing::Values(AudioStandard::NONE, AudioStandard::EDID));
+
+class AudioEncapsulationMetadataTypeRoundTripTest
+    : public testing::TestWithParam<AudioEncapsulationMetadataType> {};
+TEST_P(AudioEncapsulationMetadataTypeRoundTripTest, Aidl2Legacy2Aidl) {
+    const auto initial = GetParam();
+    auto conv =
+            aidl2legacy_AudioEncapsulationMetadataType_audio_encapsulation_metadata_type_t(initial);
+    ASSERT_TRUE(conv.ok());
+    auto convBack = legacy2aidl_audio_encapsulation_metadata_type_t_AudioEncapsulationMetadataType(
+            conv.value());
+    ASSERT_TRUE(convBack.ok());
+    EXPECT_EQ(initial, convBack.value());
+}
+INSTANTIATE_TEST_SUITE_P(AudioEncapsulationMetadataType,
+                         AudioEncapsulationMetadataTypeRoundTripTest,
+                         testing::Values(AudioEncapsulationMetadataType::NONE,
+                                         AudioEncapsulationMetadataType::FRAMEWORK_TUNER,
+                                         AudioEncapsulationMetadataType::DVB_AD_DESCRIPTOR));
+
+class AudioGainModeRoundTripTest : public testing::TestWithParam<AudioGainMode> {};
+TEST_P(AudioGainModeRoundTripTest, Aidl2Legacy2Aidl) {
+    const auto initial = GetParam();
+    auto conv = aidl2legacy_AudioGainMode_audio_gain_mode_t(initial);
+    ASSERT_TRUE(conv.ok());
+    auto convBack = legacy2aidl_audio_gain_mode_t_AudioGainMode(conv.value());
+    ASSERT_TRUE(convBack.ok());
+    EXPECT_EQ(initial, convBack.value());
+}
+INSTANTIATE_TEST_SUITE_P(AudioGainMode, AudioGainModeRoundTripTest,
+                         testing::Values(AudioGainMode::JOINT, AudioGainMode::CHANNELS,
+                                         AudioGainMode::RAMP));
+
+TEST(AudioTrackSecondaryOutputInfoRoundTripTest, Aidl2Legacy2Aidl) {
+    const auto initial = make_TrackSecondaryOutputInfo();
+    auto conv = aidl2legacy_TrackSecondaryOutputInfo_TrackSecondaryOutputInfoPair(initial);
+    ASSERT_TRUE(conv.ok());
+    auto convBack = legacy2aidl_TrackSecondaryOutputInfoPair_TrackSecondaryOutputInfo(conv.value());
+    ASSERT_TRUE(convBack.ok());
+    EXPECT_EQ(initial, convBack.value());
+}
+
+using ExtraAudioDescriptorParam = std::tuple<AudioStandard, AudioEncapsulationType>;
+class ExtraAudioDescriptorRoundTripTest : public testing::TestWithParam<ExtraAudioDescriptorParam> {
+};
+TEST_P(ExtraAudioDescriptorRoundTripTest, Aidl2Legacy2Aidl) {
+    ExtraAudioDescriptor initial =
+            make_ExtraAudioDescriptor(std::get<0>(GetParam()), std::get<1>(GetParam()));
+    auto conv = aidl2legacy_ExtraAudioDescriptor_audio_extra_audio_descriptor(initial);
+    ASSERT_TRUE(conv.ok());
+    auto convBack = legacy2aidl_audio_extra_audio_descriptor_ExtraAudioDescriptor(conv.value());
+    ASSERT_TRUE(convBack.ok());
+    EXPECT_EQ(initial, convBack.value());
+}
+
+INSTANTIATE_TEST_SUITE_P(
+        ExtraAudioDescriptor, ExtraAudioDescriptorRoundTripTest,
+        testing::Values(std::make_tuple(AudioStandard::NONE, AudioEncapsulationType::NONE),
+                        std::make_tuple(AudioStandard::EDID, AudioEncapsulationType::NONE),
+                        std::make_tuple(AudioStandard::EDID, AudioEncapsulationType::IEC61937)));
+
+TEST(AudioPortSessionExtRoundTripTest, Aidl2Legacy2Aidl) {
+    const int32_t initial = 7;
+    auto conv = aidl2legacy_int32_t_audio_port_session_ext(initial);
+    ASSERT_TRUE(conv.ok());
+    auto convBack = legacy2aidl_audio_port_session_ext_int32_t(conv.value());
+    ASSERT_TRUE(convBack.ok());
+    EXPECT_EQ(initial, convBack.value());
+}
+
+class AudioGainTest : public testing::TestWithParam<bool> {};
+TEST_P(AudioGainTest, Legacy2Aidl2Legacy) {
+    audio_port_v7 port;
+    port.num_gains = 2;
+    port.gains[0] = {.mode = AUDIO_GAIN_MODE_JOINT,
+                     .channel_mask = AUDIO_CHANNEL_IN_STEREO,
+                     .min_value = -3200,
+                     .max_value = 600,
+                     .default_value = 0,
+                     .step_value = 100,
+                     .min_ramp_ms = 10,
+                     .max_ramp_ms = 20};
+    port.gains[1] = {.mode = AUDIO_GAIN_MODE_JOINT,
+                     .channel_mask = AUDIO_CHANNEL_IN_MONO,
+                     .min_value = -8800,
+                     .max_value = 4000,
+                     .default_value = 0,
+                     .step_value = 100,
+                     .min_ramp_ms = 192,
+                     .max_ramp_ms = 224};
+
+    const auto isInput = GetParam();
+    for (int i = 0; i < port.num_gains; i++) {
+        auto initial = port.gains[i];
+        auto conv = legacy2aidl_audio_gain_AudioGain(initial, isInput);
+        ASSERT_TRUE(conv.ok());
+        auto convBack = aidl2legacy_AudioGain_audio_gain(conv.value(), isInput);
+        ASSERT_TRUE(convBack.ok());
+        EXPECT_EQ(initial.mode, convBack.value().mode);
+        EXPECT_EQ(initial.channel_mask, convBack.value().channel_mask);
+        EXPECT_EQ(initial.min_value, convBack.value().min_value);
+        EXPECT_EQ(initial.max_value, convBack.value().max_value);
+        EXPECT_EQ(initial.default_value, convBack.value().default_value);
+        EXPECT_EQ(initial.step_value, convBack.value().step_value);
+        EXPECT_EQ(initial.min_ramp_ms, convBack.value().min_ramp_ms);
+        EXPECT_EQ(initial.max_ramp_ms, convBack.value().max_ramp_ms);
+    }
+}
+INSTANTIATE_TEST_SUITE_P(AudioGain, AudioGainTest, testing::Values(true, false));
diff --git a/media/libaudioclient/tests/audio_aidl_status_tests.cpp b/media/libaudioclient/tests/audio_aidl_status_tests.cpp
index 5517091..8a7e6c1 100644
--- a/media/libaudioclient/tests/audio_aidl_status_tests.cpp
+++ b/media/libaudioclient/tests/audio_aidl_status_tests.cpp
@@ -37,25 +37,10 @@
 
 // Special status values are preserved on round trip.
 TEST(audio_aidl_status_tests, statusRoundTripSpecialValues) {
-    for (status_t status : {
-            OK,
-            UNKNOWN_ERROR,
-            NO_MEMORY,
-            INVALID_OPERATION,
-            BAD_VALUE,
-            BAD_TYPE,
-            NAME_NOT_FOUND,
-            PERMISSION_DENIED,
-            NO_INIT,
-            ALREADY_EXISTS,
-            DEAD_OBJECT,
-            FAILED_TRANSACTION,
-            BAD_INDEX,
-            NOT_ENOUGH_DATA,
-            WOULD_BLOCK,
-            TIMED_OUT,
-            UNKNOWN_TRANSACTION,
-            FDS_NOT_ALLOWED}) {
+    for (status_t status :
+         {OK, UNKNOWN_ERROR, NO_MEMORY, INVALID_OPERATION, BAD_VALUE, BAD_TYPE, NAME_NOT_FOUND,
+          PERMISSION_DENIED, NO_INIT, ALREADY_EXISTS, DEAD_OBJECT, FAILED_TRANSACTION, BAD_INDEX,
+          NOT_ENOUGH_DATA, WOULD_BLOCK, TIMED_OUT, UNKNOWN_TRANSACTION, FDS_NOT_ALLOWED}) {
         ASSERT_EQ(status, statusTFromBinderStatus(binderStatusFromStatusT(status)));
     }
 }
@@ -63,47 +48,29 @@
 // Binder exceptions show as an error (not fixed at this time); these come fromExceptionCode().
 TEST(audio_aidl_status_tests, binderStatusExceptions) {
     for (int exceptionCode : {
-            //Status::EX_NONE,
-            Status::EX_SECURITY,
-            Status::EX_BAD_PARCELABLE,
-            Status::EX_ILLEGAL_ARGUMENT,
-            Status::EX_NULL_POINTER,
-            Status::EX_ILLEGAL_STATE,
-            Status::EX_NETWORK_MAIN_THREAD,
-            Status::EX_UNSUPPORTED_OPERATION,
-            //Status::EX_SERVICE_SPECIFIC, -- tested fromServiceSpecificError()
-            Status::EX_PARCELABLE,
-            // This is special and Java specific; see Parcel.java.
-            Status::EX_HAS_REPLY_HEADER,
-            // This is special, and indicates to C++ binder proxies that the
-            // transaction has failed at a low level.
-            //Status::EX_TRANSACTION_FAILED, -- tested fromStatusT().
-            }) {
+                 // Status::EX_NONE,
+                 Status::EX_SECURITY, Status::EX_BAD_PARCELABLE, Status::EX_ILLEGAL_ARGUMENT,
+                 Status::EX_NULL_POINTER, Status::EX_ILLEGAL_STATE, Status::EX_NETWORK_MAIN_THREAD,
+                 Status::EX_UNSUPPORTED_OPERATION,
+                 // Status::EX_SERVICE_SPECIFIC, -- tested fromServiceSpecificError()
+                 Status::EX_PARCELABLE,
+                 // This is special and Java specific; see Parcel.java.
+                 Status::EX_HAS_REPLY_HEADER,
+                 // This is special, and indicates to C++ binder proxies that the
+                 // transaction has failed at a low level.
+                 // Status::EX_TRANSACTION_FAILED, -- tested fromStatusT().
+         }) {
         ASSERT_NE(OK, statusTFromBinderStatus(Status::fromExceptionCode(exceptionCode)));
     }
 }
 
 // Binder transaction errors show exactly in status_t; these come fromStatusT().
 TEST(audio_aidl_status_tests, binderStatusTransactionError) {
-    for (status_t status : {
-            OK, // Note: fromStatusT does check if this is 0, so this is no error.
-            UNKNOWN_ERROR,
-            NO_MEMORY,
-            INVALID_OPERATION,
-            BAD_VALUE,
-            BAD_TYPE,
-            NAME_NOT_FOUND,
-            PERMISSION_DENIED,
-            NO_INIT,
-            ALREADY_EXISTS,
-            DEAD_OBJECT,
-            FAILED_TRANSACTION,
-            BAD_INDEX,
-            NOT_ENOUGH_DATA,
-            WOULD_BLOCK,
-            TIMED_OUT,
-            UNKNOWN_TRANSACTION,
-            FDS_NOT_ALLOWED}) {
+    for (status_t status :
+         {OK,  // Note: fromStatusT does check if this is 0, so this is no error.
+          UNKNOWN_ERROR, NO_MEMORY, INVALID_OPERATION, BAD_VALUE, BAD_TYPE, NAME_NOT_FOUND,
+          PERMISSION_DENIED, NO_INIT, ALREADY_EXISTS, DEAD_OBJECT, FAILED_TRANSACTION, BAD_INDEX,
+          NOT_ENOUGH_DATA, WOULD_BLOCK, TIMED_OUT, UNKNOWN_TRANSACTION, FDS_NOT_ALLOWED}) {
         ASSERT_EQ(status, statusTFromBinderStatus(Status::fromStatusT(status)));
     }
 }
diff --git a/media/libaudioclient/tests/audio_test_template.xml b/media/libaudioclient/tests/audio_test_template.xml
new file mode 100644
index 0000000..ed0cb21
--- /dev/null
+++ b/media/libaudioclient/tests/audio_test_template.xml
@@ -0,0 +1,32 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2022 The Android Open Source Project
+
+     Licensed under the Apache License, Version 2.0 (the"License");
+     you may not use this file except in compliance with the License.
+     You may obtain a copy of the License at
+
+          http://www.apache.org/licenses/LICENSE-2.0
+
+     Unless required by applicable law or agreed to in writing, software
+     distributed under the License is distributed on an"AS IS" BASIS,
+     WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+     See the License for the specific language governing permissions and
+     limitations under the License.
+-->
+<configuration description="Unit test configuration for {MODULE}">
+    <target_preparer class="com.android.tradefed.targetprep.RootTargetPreparer" />
+
+    <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
+        <option name="cleanup" value="true" />
+        <option name="push-file" key="{MODULE}" value="/data/local/tmp/{MODULE}" />
+
+        <!-- Files used for audio testing -->
+        <option name="push-file" key="bbb_1ch_8kHz_s16le.raw" value="/data/local/tmp/bbb_1ch_8kHz_s16le.raw" />
+        <option name="push-file" key="bbb_2ch_24kHz_s16le.raw" value="/data/local/tmp/bbb_2ch_24kHz_s16le.raw" />
+    </target_preparer>
+
+    <test class="com.android.tradefed.testtype.GTest" >
+        <option name="native-test-device-path" value="/data/local/tmp" />
+        <option name="module-name" value="{MODULE}" />
+    </test>
+</configuration>
diff --git a/media/libaudioclient/tests/audio_test_utils.cpp b/media/libaudioclient/tests/audio_test_utils.cpp
new file mode 100644
index 0000000..018d920
--- /dev/null
+++ b/media/libaudioclient/tests/audio_test_utils.cpp
@@ -0,0 +1,795 @@
+/*
+ * Copyright (C) 2021 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 "AudioTestUtils"
+
+#include <utils/Log.h>
+
+#include "audio_test_utils.h"
+
+// Generates a random string.
+void CreateRandomFile(int& fd) {
+    std::string filename = "/data/local/tmp/record-XXXXXX";
+    fd = mkstemp(filename.data());
+}
+
+void OnAudioDeviceUpdateNotifier::onAudioDeviceUpdate(audio_io_handle_t audioIo,
+                                                      audio_port_handle_t deviceId) {
+    std::unique_lock<std::mutex> lock{mMutex};
+    ALOGD("%s  audioIo=%d deviceId=%d", __func__, audioIo, deviceId);
+    mAudioIo = audioIo;
+    mDeviceId = deviceId;
+    mCondition.notify_all();
+}
+
+status_t OnAudioDeviceUpdateNotifier::waitForAudioDeviceCb() {
+    std::unique_lock<std::mutex> lock{mMutex};
+    if (mAudioIo == AUDIO_IO_HANDLE_NONE) {
+        mCondition.wait_for(lock, std::chrono::milliseconds(500));
+        if (mAudioIo == AUDIO_IO_HANDLE_NONE) return TIMED_OUT;
+    }
+    return OK;
+}
+
+AudioPlayback::AudioPlayback(uint32_t sampleRate, audio_format_t format,
+                             audio_channel_mask_t channelMask, audio_output_flags_t flags,
+                             audio_session_t sessionId, AudioTrack::transfer_type transferType,
+                             audio_attributes_t* attributes, audio_offload_info_t* info)
+    : mSampleRate(sampleRate),
+      mFormat(format),
+      mChannelMask(channelMask),
+      mFlags(flags),
+      mSessionId(sessionId),
+      mTransferType(transferType),
+      mAttributes(attributes),
+      mOffloadInfo(info) {
+    mStopPlaying = false;
+    mBytesUsedSoFar = 0;
+    mState = PLAY_NO_INIT;
+    mMemCapacity = 0;
+    mMemoryDealer = nullptr;
+    mMemory = nullptr;
+}
+
+AudioPlayback::~AudioPlayback() {
+    stop();
+}
+
+status_t AudioPlayback::create() {
+    if (mState != PLAY_NO_INIT) return INVALID_OPERATION;
+    std::string packageName{"AudioPlayback"};
+    AttributionSourceState attributionSource;
+    attributionSource.packageName = packageName;
+    attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
+    attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
+    attributionSource.token = sp<BBinder>::make();
+    if (mTransferType == AudioTrack::TRANSFER_OBTAIN) {
+        mTrack = new AudioTrack(attributionSource);
+        mTrack->set(AUDIO_STREAM_MUSIC, mSampleRate, mFormat, mChannelMask, 0 /* frameCount */,
+                    mFlags, nullptr /* callback */, 0 /* notificationFrames */,
+                    nullptr /* sharedBuffer */, false /*canCallJava */, mSessionId, mTransferType,
+                    mOffloadInfo, attributionSource, mAttributes);
+    } else if (mTransferType == AudioTrack::TRANSFER_SHARED) {
+        mTrack = new AudioTrack(AUDIO_STREAM_MUSIC, mSampleRate, mFormat, mChannelMask, mMemory,
+                                mFlags, wp<AudioTrack::IAudioTrackCallback>::fromExisting(this), 0,
+                                mSessionId, mTransferType, nullptr, attributionSource, mAttributes);
+    } else {
+        ALOGE("Test application is not handling transfer type %s",
+              AudioTrack::convertTransferToText(mTransferType));
+        return INVALID_OPERATION;
+    }
+    mTrack->setCallerName(packageName);
+    status_t status = mTrack->initCheck();
+    if (NO_ERROR == status) mState = PLAY_READY;
+    return status;
+}
+
+status_t AudioPlayback::loadResource(const char* name) {
+    status_t status = OK;
+    FILE* fp = fopen(name, "rbe");
+    struct stat buf {};
+    if (fp && !fstat(fileno(fp), &buf)) {
+        mMemCapacity = buf.st_size;
+        mMemoryDealer = new MemoryDealer(mMemCapacity, "AudioPlayback");
+        if (nullptr == mMemoryDealer.get()) {
+            ALOGE("couldn't get MemoryDealer!");
+            fclose(fp);
+            return NO_MEMORY;
+        }
+        mMemory = mMemoryDealer->allocate(mMemCapacity);
+        if (nullptr == mMemory.get()) {
+            ALOGE("couldn't get IMemory!");
+            fclose(fp);
+            return NO_MEMORY;
+        }
+        uint8_t* ipBuffer = static_cast<uint8_t*>(static_cast<void*>(mMemory->unsecurePointer()));
+        fread(ipBuffer, sizeof(uint8_t), mMemCapacity, fp);
+    } else {
+        ALOGE("unable to open input file %s", name);
+        status = NAME_NOT_FOUND;
+    }
+    if (fp) fclose(fp);
+    return status;
+}
+
+sp<AudioTrack> AudioPlayback::getAudioTrackHandle() {
+    return (PLAY_NO_INIT != mState) ? mTrack : nullptr;
+}
+
+status_t AudioPlayback::start() {
+    status_t status;
+    if (PLAY_READY != mState) {
+        return INVALID_OPERATION;
+    } else {
+        status = mTrack->start();
+        if (OK == status) {
+            mState = PLAY_STARTED;
+            LOG_FATAL_IF(false != mTrack->stopped());
+        }
+    }
+    return status;
+}
+
+void AudioPlayback::onBufferEnd() {
+    std::unique_lock<std::mutex> lock{mMutex};
+    mStopPlaying = true;
+    mCondition.notify_all();
+}
+
+status_t AudioPlayback::fillBuffer() {
+    if (PLAY_STARTED != mState && PLAY_STOPPED != mState) return INVALID_OPERATION;
+    int retry = 25;
+    uint8_t* ipBuffer = static_cast<uint8_t*>(static_cast<void*>(mMemory->unsecurePointer()));
+    size_t nonContig = 0;
+    size_t bytesAvailable = mMemCapacity - mBytesUsedSoFar;
+    while (bytesAvailable > 0) {
+        AudioTrack::Buffer trackBuffer;
+        trackBuffer.frameCount = mTrack->frameCount() * 2;
+        status_t status = mTrack->obtainBuffer(&trackBuffer, retry, &nonContig);
+        if (OK == status) {
+            size_t bytesToCopy = std::min(bytesAvailable, trackBuffer.size());
+            if (bytesToCopy > 0) {
+                memcpy(trackBuffer.data(), ipBuffer + mBytesUsedSoFar, bytesToCopy);
+            }
+            mTrack->releaseBuffer(&trackBuffer);
+            mBytesUsedSoFar += bytesToCopy;
+            bytesAvailable = mMemCapacity - mBytesUsedSoFar;
+            if (bytesAvailable == 0) {
+                stop();
+            }
+        } else if (WOULD_BLOCK == status) {
+            if (mStopPlaying)
+                return OK;
+            else
+                return TIMED_OUT;
+        }
+    }
+    return OK;
+}
+
+status_t AudioPlayback::waitForConsumption(bool testSeek) {
+    if (PLAY_STARTED != mState) return INVALID_OPERATION;
+    // in static buffer mode, lets not play clips with duration > 30 sec
+    int retry = 30;
+    // Total number of frames in the input file.
+    size_t totalFrameCount = mMemCapacity / mTrack->frameSize();
+    while (!mStopPlaying && retry > 0) {
+        // Get the total numbers of frames played.
+        uint32_t currPosition;
+        mTrack->getPosition(&currPosition);
+        if (testSeek && (currPosition > totalFrameCount * 0.6)) {
+            testSeek = false;
+            if (!mTrack->hasStarted()) return BAD_VALUE;
+            mTrack->pauseAndWait(std::chrono::seconds(2));
+            if (mTrack->hasStarted()) return BAD_VALUE;
+            mTrack->reload();
+            mTrack->getPosition(&currPosition);
+            if (currPosition != 0) return BAD_VALUE;
+            mTrack->start();
+            while (currPosition < totalFrameCount * 0.3) {
+                mTrack->getPosition(&currPosition);
+            }
+            mTrack->pauseAndWait(std::chrono::seconds(2));
+            uint32_t setPosition = totalFrameCount * 0.9;
+            mTrack->setPosition(setPosition);
+            uint32_t bufferPosition;
+            mTrack->getBufferPosition(&bufferPosition);
+            if (bufferPosition != setPosition) return BAD_VALUE;
+            mTrack->start();
+        }
+        std::this_thread::sleep_for(std::chrono::milliseconds(300));
+        retry--;
+    }
+    if (!mStopPlaying) return TIMED_OUT;
+    return OK;
+}
+
+status_t AudioPlayback::onProcess(bool testSeek) {
+    if (mTransferType == AudioTrack::TRANSFER_SHARED)
+        return waitForConsumption(testSeek);
+    else if (mTransferType == AudioTrack::TRANSFER_OBTAIN)
+        return fillBuffer();
+    else
+        return INVALID_OPERATION;
+}
+
+void AudioPlayback::stop() {
+    std::unique_lock<std::mutex> lock{mMutex};
+    mStopPlaying = true;
+    if (mState != PLAY_STOPPED) {
+        int32_t msec = 0;
+        (void)mTrack->pendingDuration(&msec);
+        mTrack->stopAndJoinCallbacks();
+        LOG_FATAL_IF(true != mTrack->stopped());
+        mState = PLAY_STOPPED;
+        if (msec > 0) {
+            ALOGD("deleting recycled track, waiting for data drain (%d msec)", msec);
+            usleep(msec * 1000LL);
+        }
+    }
+}
+
+// hold pcm data sent by AudioRecord
+RawBuffer::RawBuffer(int64_t ptsPipeline, int64_t ptsManual, int32_t capacity)
+    : mData(capacity > 0 ? new uint8_t[capacity] : nullptr),
+      mPtsPipeline(ptsPipeline),
+      mPtsManual(ptsManual),
+      mCapacity(capacity) {}
+
+// Simple AudioCapture
+size_t AudioCapture::onMoreData(const AudioRecord::Buffer& buffer) {
+    if (mState != REC_STARTED) {
+        ALOGE("Unexpected Callback from audiorecord, not reading data");
+        return 0;
+    }
+
+    // no more frames to read
+    if (mNumFramesReceived > mNumFramesToRecord || mStopRecording) {
+        mStopRecording = true;
+        return 0;
+    }
+
+    int64_t timeUs = 0, position = 0, timeNs = 0;
+    ExtendedTimestamp ts;
+    ExtendedTimestamp::Location location;
+    const int32_t usPerSec = 1000000;
+
+    if (mRecord->getTimestamp(&ts) == OK &&
+        ts.getBestTimestamp(&position, &timeNs, ExtendedTimestamp::TIMEBASE_MONOTONIC, &location) ==
+                OK) {
+        // Use audio timestamp.
+        timeUs = timeNs / 1000 -
+                 (position - mNumFramesReceived + mNumFramesLost) * usPerSec / mSampleRate;
+    } else {
+        // This should not happen in normal case.
+        ALOGW("Failed to get audio timestamp, fallback to use systemclock");
+        timeUs = systemTime() / 1000LL;
+        // Estimate the real sampling time of the 1st sample in this buffer
+        // from AudioRecord's latency. (Apply this adjustment first so that
+        // the start time logic is not affected.)
+        timeUs -= mRecord->latency() * 1000LL;
+    }
+
+    ALOGV("dataCallbackTimestamp: %" PRId64 " us", timeUs);
+
+    const size_t frameSize = mRecord->frameSize();
+    uint64_t numLostBytes = (uint64_t)mRecord->getInputFramesLost() * frameSize;
+    if (numLostBytes > 0) {
+        ALOGW("Lost audio record data: %" PRIu64 " bytes", numLostBytes);
+    }
+    std::deque<RawBuffer> tmpQueue;
+    while (numLostBytes > 0) {
+        uint64_t bufferSize = numLostBytes;
+        if (numLostBytes > mMaxBytesPerCallback) {
+            numLostBytes -= mMaxBytesPerCallback;
+            bufferSize = mMaxBytesPerCallback;
+        } else {
+            numLostBytes = 0;
+        }
+        const int64_t timestampUs =
+                ((1000000LL * mNumFramesReceived) + (mRecord->getSampleRate() >> 1)) /
+                mRecord->getSampleRate();
+        RawBuffer emptyBuffer{timeUs, timestampUs, static_cast<int32_t>(bufferSize)};
+        memset(emptyBuffer.mData.get(), 0, bufferSize);
+        mNumFramesLost += bufferSize / frameSize;
+        mNumFramesReceived += bufferSize / frameSize;
+        tmpQueue.push_back(std::move(emptyBuffer));
+    }
+
+    if (buffer.size() == 0) {
+        ALOGW("Nothing is available from AudioRecord callback buffer");
+    } else {
+        const size_t bufferSize = buffer.size();
+        const int64_t timestampUs =
+                ((1000000LL * mNumFramesReceived) + (mRecord->getSampleRate() >> 1)) /
+                mRecord->getSampleRate();
+        RawBuffer audioBuffer{timeUs, timestampUs, static_cast<int32_t>(bufferSize)};
+        memcpy(audioBuffer.mData.get(), buffer.data(), bufferSize);
+        mNumFramesReceived += bufferSize / frameSize;
+        tmpQueue.push_back(std::move(audioBuffer));
+    }
+
+    if (tmpQueue.size() > 0) {
+        std::unique_lock<std::mutex> lock{mMutex};
+        for (auto it = tmpQueue.begin(); it != tmpQueue.end(); it++)
+            mBuffersReceived.push_back(std::move(*it));
+        mCondition.notify_all();
+    }
+    return buffer.size();
+}
+
+void AudioCapture::onOverrun() {
+    ALOGV("received event overrun");
+    mBufferOverrun = true;
+}
+
+void AudioCapture::onMarker(uint32_t markerPosition) {
+    ALOGV("received Callback at position %d", markerPosition);
+    mReceivedCbMarkerAtPosition = markerPosition;
+}
+
+void AudioCapture::onNewPos(uint32_t markerPosition) {
+    ALOGV("received Callback at position %d", markerPosition);
+    mReceivedCbMarkerCount++;
+}
+
+void AudioCapture::onNewIAudioRecord() {
+    ALOGV("IAudioRecord is re-created");
+}
+
+AudioCapture::AudioCapture(audio_source_t inputSource, uint32_t sampleRate, audio_format_t format,
+                           audio_channel_mask_t channelMask, audio_input_flags_t flags,
+                           audio_session_t sessionId, AudioRecord::transfer_type transferType)
+    : mInputSource(inputSource),
+      mSampleRate(sampleRate),
+      mFormat(format),
+      mChannelMask(channelMask),
+      mFlags(flags),
+      mSessionId(sessionId),
+      mTransferType(transferType) {
+    mFrameCount = 0;
+    mNotificationFrames = 0;
+    mNumFramesToRecord = 0;
+    mNumFramesReceived = 0;
+    mNumFramesLost = 0;
+    mBufferOverrun = false;
+    mMarkerPosition = 0;
+    mMarkerPeriod = 0;
+    mReceivedCbMarkerAtPosition = -1;
+    mReceivedCbMarkerCount = 0;
+    mState = REC_NO_INIT;
+    mStopRecording = false;
+#if RECORD_TO_FILE
+    CreateRandomFile(mOutFileFd);
+#endif
+}
+
+AudioCapture::~AudioCapture() {
+    if (mOutFileFd > 0) close(mOutFileFd);
+    stop();
+}
+
+status_t AudioCapture::create() {
+    if (mState != REC_NO_INIT) return INVALID_OPERATION;
+    // get Min Frame Count
+    size_t minFrameCount;
+    status_t status =
+            AudioRecord::getMinFrameCount(&minFrameCount, mSampleRate, mFormat, mChannelMask);
+    if (NO_ERROR != status) return status;
+    // Limit notificationFrames basing on client bufferSize
+    const int samplesPerFrame = audio_channel_count_from_in_mask(mChannelMask);
+    const int bytesPerSample = audio_bytes_per_sample(mFormat);
+    mNotificationFrames = mMaxBytesPerCallback / (samplesPerFrame * bytesPerSample);
+    // select frameCount to be at least minFrameCount
+    mFrameCount = 2 * mNotificationFrames;
+    while (mFrameCount < minFrameCount) {
+        mFrameCount += mNotificationFrames;
+    }
+    if (mFlags & AUDIO_INPUT_FLAG_FAST) {
+        ALOGW("Overriding all previous computations");
+        mFrameCount = 0;
+        mNotificationFrames = 0;
+    }
+    mNumFramesToRecord = (mSampleRate * 0.25);  // record .25 sec
+    std::string packageName{"AudioCapture"};
+    AttributionSourceState attributionSource;
+    attributionSource.packageName = packageName;
+    attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
+    attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
+    attributionSource.token = sp<BBinder>::make();
+    if (mTransferType == AudioRecord::TRANSFER_OBTAIN) {
+        if (mSampleRate == 48000) {  // test all available constructors
+            mRecord = new AudioRecord(mInputSource, mSampleRate, mFormat, mChannelMask,
+                                      attributionSource, mFrameCount, nullptr /* callback */,
+                                      mNotificationFrames, mSessionId, mTransferType, mFlags);
+        } else {
+            mRecord = new AudioRecord(attributionSource);
+            status = mRecord->set(mInputSource, mSampleRate, mFormat, mChannelMask, mFrameCount,
+                                  nullptr /* callback */, 0 /* notificationFrames */,
+                                  false /* canCallJava */, mSessionId, mTransferType, mFlags,
+                                  attributionSource.uid, attributionSource.pid);
+        }
+        if (NO_ERROR != status) return status;
+    } else if (mTransferType == AudioRecord::TRANSFER_CALLBACK) {
+        mRecord = new AudioRecord(mInputSource, mSampleRate, mFormat, mChannelMask,
+                                  attributionSource, mFrameCount, this, mNotificationFrames,
+                                  mSessionId, mTransferType, mFlags);
+    } else {
+        ALOGE("Test application is not handling transfer type %s",
+              AudioRecord::convertTransferToText(mTransferType));
+        return NO_INIT;
+    }
+    mRecord->setCallerName(packageName);
+    status = mRecord->initCheck();
+    if (NO_ERROR == status) mState = REC_READY;
+    if (mFlags & AUDIO_INPUT_FLAG_FAST) {
+        mFrameCount = mRecord->frameCount();
+        mNotificationFrames = mRecord->getNotificationPeriodInFrames();
+        mMaxBytesPerCallback = mNotificationFrames * samplesPerFrame * bytesPerSample;
+    }
+    return status;
+}
+
+sp<AudioRecord> AudioCapture::getAudioRecordHandle() {
+    return (REC_NO_INIT == mState) ? nullptr : mRecord;
+}
+
+status_t AudioCapture::start(AudioSystem::sync_event_t event, audio_session_t triggerSession) {
+    status_t status;
+    if (REC_READY != mState) {
+        return INVALID_OPERATION;
+    } else {
+        status = mRecord->start(event, triggerSession);
+        if (OK == status) {
+            mState = REC_STARTED;
+            LOG_FATAL_IF(false != mRecord->stopped());
+        }
+    }
+    return status;
+}
+
+status_t AudioCapture::stop() {
+    status_t status = OK;
+    mStopRecording = true;
+    if (mState != REC_STOPPED) {
+        mRecord->stopAndJoinCallbacks();
+        mState = REC_STOPPED;
+        LOG_FATAL_IF(true != mRecord->stopped());
+    }
+    return status;
+}
+
+status_t AudioCapture::obtainBuffer(RawBuffer& buffer) {
+    if (REC_STARTED != mState && REC_STOPPED != mState) return INVALID_OPERATION;
+    int retry = 25;
+    AudioRecord::Buffer recordBuffer;
+    recordBuffer.frameCount = mNotificationFrames;
+    size_t nonContig = 0;
+    status_t status = mRecord->obtainBuffer(&recordBuffer, retry, &nonContig);
+    if (OK == status) {
+        const int64_t timestampUs =
+                ((1000000LL * mNumFramesReceived) + (mRecord->getSampleRate() >> 1)) /
+                mRecord->getSampleRate();
+        RawBuffer buff{-1, timestampUs, static_cast<int32_t>(recordBuffer.size())};
+        memcpy(buff.mData.get(), recordBuffer.data(), recordBuffer.size());
+        buffer = std::move(buff);
+        mNumFramesReceived += recordBuffer.size() / mRecord->frameSize();
+        mRecord->releaseBuffer(&recordBuffer);
+        if (mNumFramesReceived > mNumFramesToRecord) {
+            stop();
+        }
+    } else if (status == WOULD_BLOCK) {
+        if (mStopRecording)
+            return WOULD_BLOCK;
+        else
+            return TIMED_OUT;
+    }
+    return OK;
+}
+
+status_t AudioCapture::obtainBufferCb(RawBuffer& buffer) {
+    if (REC_STARTED != mState) return INVALID_OPERATION;
+    int retry = 10;
+    std::unique_lock<std::mutex> lock{mMutex};
+    while (mBuffersReceived.empty() && !mStopRecording && retry > 0) {
+        mCondition.wait_for(lock, std::chrono::milliseconds(100));
+        retry--;
+    }
+    if (!mBuffersReceived.empty()) {
+        auto it = mBuffersReceived.begin();
+        buffer = std::move(*it);
+        mBuffersReceived.erase(it);
+    } else {
+        if (retry == 0) return TIMED_OUT;
+        if (mStopRecording)
+            return WOULD_BLOCK;
+        else
+            return UNKNOWN_ERROR;
+    }
+    return OK;
+}
+
+status_t AudioCapture::audioProcess() {
+    RawBuffer buffer;
+    while (true) {
+        status_t status;
+        if (mTransferType == AudioRecord::TRANSFER_CALLBACK)
+            status = obtainBufferCb(buffer);
+        else
+            status = obtainBuffer(buffer);
+        switch (status) {
+            case OK:
+                if (mOutFileFd > 0) {
+                    const char* ptr =
+                            static_cast<const char*>(static_cast<void*>(buffer.mData.get()));
+                    write(mOutFileFd, ptr, buffer.mCapacity);
+                }
+                break;
+            case WOULD_BLOCK:
+                return OK;
+            case TIMED_OUT:          // "recorder application timed out from receiving buffers"
+            case NO_INIT:            // "recorder not initialized"
+            case INVALID_OPERATION:  // "recorder not started"
+            case UNKNOWN_ERROR:      // "Unknown error"
+            default:
+                return status;
+        }
+    }
+}
+
+status_t listAudioPorts(std::vector<audio_port_v7>& portsVec) {
+    int attempts = 5;
+    status_t status;
+    unsigned int generation1, generation;
+    unsigned int numPorts = 0;
+    do {
+        if (attempts-- < 0) {
+            status = TIMED_OUT;
+            break;
+        }
+        status = AudioSystem::listAudioPorts(AUDIO_PORT_ROLE_NONE, AUDIO_PORT_TYPE_NONE, &numPorts,
+                                             nullptr, &generation1);
+        if (status != NO_ERROR) {
+            ALOGE("AudioSystem::listAudioPorts returned error %d", status);
+            break;
+        }
+        portsVec.resize(numPorts);
+        status = AudioSystem::listAudioPorts(AUDIO_PORT_ROLE_NONE, AUDIO_PORT_TYPE_NONE, &numPorts,
+                                             portsVec.data(), &generation);
+    } while (generation1 != generation && status == NO_ERROR);
+    if (status != NO_ERROR) {
+        numPorts = 0;
+        portsVec.clear();
+    }
+    return status;
+}
+
+status_t getPortById(const audio_port_handle_t portId, audio_port_v7& port) {
+    std::vector<struct audio_port_v7> ports;
+    status_t status = listAudioPorts(ports);
+    if (status != OK) return status;
+    for (auto i = 0; i < ports.size(); i++) {
+        if (ports[i].id == portId) {
+            port = ports[i];
+            return OK;
+        }
+    }
+    return BAD_VALUE;
+}
+
+status_t getPortByAttributes(audio_port_role_t role, audio_port_type_t type,
+                             audio_devices_t deviceType, audio_port_v7& port) {
+    std::vector<struct audio_port_v7> ports;
+    status_t status = listAudioPorts(ports);
+    if (status != OK) return status;
+    for (auto i = 0; i < ports.size(); i++) {
+        if (ports[i].role == role && ports[i].type == type &&
+            ports[i].ext.device.type == deviceType) {
+            port = ports[i];
+            return OK;
+        }
+    }
+    return BAD_VALUE;
+}
+
+status_t listAudioPatches(std::vector<struct audio_patch>& patchesVec) {
+    int attempts = 5;
+    status_t status;
+    unsigned int generation1, generation;
+    unsigned int numPatches = 0;
+    do {
+        if (attempts-- < 0) {
+            status = TIMED_OUT;
+            break;
+        }
+        status = AudioSystem::listAudioPatches(&numPatches, nullptr, &generation1);
+        if (status != NO_ERROR) {
+            ALOGE("AudioSystem::listAudioPatches returned error %d", status);
+            break;
+        }
+        patchesVec.resize(numPatches);
+        status = AudioSystem::listAudioPatches(&numPatches, patchesVec.data(), &generation);
+    } while (generation1 != generation && status == NO_ERROR);
+    if (status != NO_ERROR) {
+        numPatches = 0;
+        patchesVec.clear();
+    }
+    return status;
+}
+
+status_t getPatchForOutputMix(audio_io_handle_t audioIo, audio_patch& patch) {
+    std::vector<struct audio_patch> patches;
+    status_t status = listAudioPatches(patches);
+    if (status != OK) return status;
+
+    for (auto i = 0; i < patches.size(); i++) {
+        for (auto j = 0; j < patches[i].num_sources; j++) {
+            if (patches[i].sources[j].type == AUDIO_PORT_TYPE_MIX &&
+                patches[i].sources[j].ext.mix.handle == audioIo) {
+                patch = patches[i];
+                return OK;
+            }
+        }
+    }
+    return BAD_VALUE;
+}
+
+status_t getPatchForInputMix(audio_io_handle_t audioIo, audio_patch& patch) {
+    std::vector<struct audio_patch> patches;
+    status_t status = listAudioPatches(patches);
+    if (status != OK) return status;
+
+    for (auto i = 0; i < patches.size(); i++) {
+        for (auto j = 0; j < patches[i].num_sinks; j++) {
+            if (patches[i].sinks[j].type == AUDIO_PORT_TYPE_MIX &&
+                patches[i].sinks[j].ext.mix.handle == audioIo) {
+                patch = patches[i];
+                return OK;
+            }
+        }
+    }
+    return BAD_VALUE;
+}
+
+bool patchContainsOutputDevice(audio_port_handle_t deviceId, audio_patch patch) {
+    for (auto j = 0; j < patch.num_sinks; j++) {
+        if (patch.sinks[j].type == AUDIO_PORT_TYPE_DEVICE && patch.sinks[j].id == deviceId) {
+            return true;
+        }
+    }
+    return false;
+}
+
+bool patchContainsInputDevice(audio_port_handle_t deviceId, audio_patch patch) {
+    for (auto j = 0; j < patch.num_sources; j++) {
+        if (patch.sources[j].type == AUDIO_PORT_TYPE_DEVICE && patch.sources[j].id == deviceId) {
+            return true;
+        }
+    }
+    return false;
+}
+
+bool checkPatchPlayback(audio_io_handle_t audioIo, audio_port_handle_t deviceId) {
+    struct audio_patch patch;
+    if (getPatchForOutputMix(audioIo, patch) == OK) {
+        return patchContainsOutputDevice(deviceId, patch);
+    }
+    return false;
+}
+
+bool checkPatchCapture(audio_io_handle_t audioIo, audio_port_handle_t deviceId) {
+    struct audio_patch patch;
+    if (getPatchForInputMix(audioIo, patch) == OK) {
+        return patchContainsInputDevice(deviceId, patch);
+    }
+    return false;
+}
+
+std::string dumpPortConfig(const audio_port_config& port) {
+    std::ostringstream result;
+    std::string deviceInfo;
+    if (port.type == AUDIO_PORT_TYPE_DEVICE) {
+        if (port.ext.device.type & AUDIO_DEVICE_BIT_IN) {
+            InputDeviceConverter::maskToString(port.ext.device.type, deviceInfo);
+        } else {
+            OutputDeviceConverter::maskToString(port.ext.device.type, deviceInfo);
+        }
+        deviceInfo += std::string(", address = ") + port.ext.device.address;
+    }
+    result << "audio_port_handle_t = " << port.id << ", "
+           << "Role = " << (port.role == AUDIO_PORT_ROLE_SOURCE ? "source" : "sink") << ", "
+           << "Type = " << (port.type == AUDIO_PORT_TYPE_DEVICE ? "device" : "mix") << ", "
+           << "deviceInfo = " << (port.type == AUDIO_PORT_TYPE_DEVICE ? deviceInfo : "") << ", "
+           << "config_mask = 0x" << std::hex << port.config_mask << std::dec << ", ";
+    if (port.config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+        result << "sample rate = " << port.sample_rate << ", ";
+    }
+    if (port.config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+        result << "channel mask = " << port.channel_mask << ", ";
+    }
+    if (port.config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+        result << "format = " << port.format << ", ";
+    }
+    result << "input flags = " << port.flags.input << ", ";
+    result << "output flags = " << port.flags.output << ", ";
+    result << "mix io handle = " << (port.type == AUDIO_PORT_TYPE_DEVICE ? 0 : port.ext.mix.handle)
+           << "\n";
+    return result.str();
+}
+
+std::string dumpPatch(const audio_patch& patch) {
+    std::ostringstream result;
+    result << "----------------- Dumping Patch ------------ \n";
+    result << "Patch Handle: " << patch.id << ", sources: " << patch.num_sources
+           << ", sink: " << patch.num_sinks << "\n";
+    audio_port_v7 port;
+    for (uint32_t i = 0; i < patch.num_sources; i++) {
+        result << "----------------- Dumping Source Port Config @ index " << i
+               << " ------------ \n";
+        result << dumpPortConfig(patch.sources[i]);
+        result << "----------------- Dumping Source Port for id " << patch.sources[i].id
+               << " ------------ \n";
+        getPortById(patch.sources[i].id, port);
+        result << dumpPort(port);
+    }
+    for (uint32_t i = 0; i < patch.num_sinks; i++) {
+        result << "----------------- Dumping Sink Port Config @ index " << i << " ------------ \n";
+        result << dumpPortConfig(patch.sinks[i]);
+        result << "----------------- Dumping Sink Port for id " << patch.sinks[i].id
+               << " ------------ \n";
+        getPortById(patch.sinks[i].id, port);
+        result << dumpPort(port);
+    }
+    return result.str();
+}
+
+std::string dumpPort(const audio_port_v7& port) {
+    std::ostringstream result;
+    std::string deviceInfo;
+    if (port.type == AUDIO_PORT_TYPE_DEVICE) {
+        if (port.ext.device.type & AUDIO_DEVICE_BIT_IN) {
+            InputDeviceConverter::maskToString(port.ext.device.type, deviceInfo);
+        } else {
+            OutputDeviceConverter::maskToString(port.ext.device.type, deviceInfo);
+        }
+        deviceInfo += std::string(", address = ") + port.ext.device.address;
+    }
+    result << "audio_port_handle_t = " << port.id << ", "
+           << "Role = " << (port.role == AUDIO_PORT_ROLE_SOURCE ? "source" : "sink") << ", "
+           << "Type = " << (port.type == AUDIO_PORT_TYPE_DEVICE ? "device" : "mix") << ", "
+           << "deviceInfo = " << (port.type == AUDIO_PORT_TYPE_DEVICE ? deviceInfo : "") << ", "
+           << "Name = " << port.name << ", "
+           << "num profiles = " << port.num_audio_profiles << ", "
+           << "mix io handle = " << (port.type == AUDIO_PORT_TYPE_DEVICE ? 0 : port.ext.mix.handle)
+           << ", ";
+    for (int i = 0; i < port.num_audio_profiles; i++) {
+        result << "AudioProfile = " << i << " {";
+        result << "format = " << port.audio_profiles[i].format << ", ";
+        result << "samplerates = ";
+        for (int j = 0; j < port.audio_profiles[i].num_sample_rates; j++) {
+            result << port.audio_profiles[i].sample_rates[j] << ", ";
+        }
+        result << "channelmasks = ";
+        for (int j = 0; j < port.audio_profiles[i].num_channel_masks; j++) {
+            result << "0x" << std::hex << port.audio_profiles[i].channel_masks[j] << std::dec
+                   << ", ";
+        }
+        result << "} ";
+    }
+    result << dumpPortConfig(port.active_config);
+    return result.str();
+}
diff --git a/media/libaudioclient/tests/audio_test_utils.h b/media/libaudioclient/tests/audio_test_utils.h
new file mode 100644
index 0000000..526d5c4
--- /dev/null
+++ b/media/libaudioclient/tests/audio_test_utils.h
@@ -0,0 +1,188 @@
+/*
+ * Copyright (C) 2021 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 AUDIO_TEST_UTILS_H_
+#define AUDIO_TEST_UTILS_H_
+
+#include <sys/stat.h>
+#include <unistd.h>
+#include <atomic>
+#include <chrono>
+#include <cinttypes>
+#include <deque>
+#include <memory>
+#include <mutex>
+#include <thread>
+
+#include <binder/MemoryDealer.h>
+#include <media/AidlConversion.h>
+#include <media/AudioRecord.h>
+#include <media/AudioTrack.h>
+
+#define RECORD_TO_FILE 0
+
+using namespace android;
+
+void CreateRandomFile(int& fd);
+status_t listAudioPorts(std::vector<audio_port_v7>& portsVec);
+status_t listAudioPatches(std::vector<struct audio_patch>& patchesVec);
+status_t getPortByAttributes(audio_port_role_t role, audio_port_type_t type,
+                             audio_devices_t deviceType, audio_port_v7& port);
+status_t getPatchForOutputMix(audio_io_handle_t audioIo, audio_patch& patch);
+status_t getPatchForInputMix(audio_io_handle_t audioIo, audio_patch& patch);
+bool patchContainsOutputDevice(audio_port_handle_t deviceId, audio_patch patch);
+bool patchContainsInputDevice(audio_port_handle_t deviceId, audio_patch patch);
+bool checkPatchPlayback(audio_io_handle_t audioIo, audio_port_handle_t deviceId);
+bool checkPatchCapture(audio_io_handle_t audioIo, audio_port_handle_t deviceId);
+std::string dumpPort(const audio_port_v7& port);
+std::string dumpPortConfig(const audio_port_config& port);
+std::string dumpPatch(const audio_patch& patch);
+
+class OnAudioDeviceUpdateNotifier : public AudioSystem::AudioDeviceCallback {
+  public:
+    audio_io_handle_t mAudioIo = AUDIO_IO_HANDLE_NONE;
+    audio_port_handle_t mDeviceId = AUDIO_PORT_HANDLE_NONE;
+    std::mutex mMutex;
+    std::condition_variable mCondition;
+
+    void onAudioDeviceUpdate(audio_io_handle_t audioIo, audio_port_handle_t deviceId);
+    status_t waitForAudioDeviceCb();
+};
+
+// Simple AudioPlayback class.
+class AudioPlayback : public AudioTrack::IAudioTrackCallback {
+    friend sp<AudioPlayback>;
+    AudioPlayback(uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask,
+                  audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+                  audio_session_t sessionId = AUDIO_SESSION_NONE,
+                  AudioTrack::transfer_type transferType = AudioTrack::TRANSFER_SHARED,
+                  audio_attributes_t* attributes = nullptr, audio_offload_info_t* info = nullptr);
+
+  public:
+    status_t loadResource(const char* name);
+    status_t create();
+    sp<AudioTrack> getAudioTrackHandle();
+    status_t start();
+    status_t waitForConsumption(bool testSeek = false);
+    status_t fillBuffer();
+    status_t onProcess(bool testSeek = false);
+    virtual void onBufferEnd() override;
+    void stop();
+
+    bool mStopPlaying;
+    std::mutex mMutex;
+    std::condition_variable mCondition;
+
+    enum State {
+        PLAY_NO_INIT,
+        PLAY_READY,
+        PLAY_STARTED,
+        PLAY_STOPPED,
+    };
+
+  private:
+    ~AudioPlayback();
+    const uint32_t mSampleRate;
+    const audio_format_t mFormat;
+    const audio_channel_mask_t mChannelMask;
+    const audio_output_flags_t mFlags;
+    const audio_session_t mSessionId;
+    const AudioTrack::transfer_type mTransferType;
+    const audio_attributes_t* mAttributes;
+    const audio_offload_info_t* mOffloadInfo;
+
+    size_t mBytesUsedSoFar;
+    State mState;
+    size_t mMemCapacity;
+    sp<MemoryDealer> mMemoryDealer;
+    sp<IMemory> mMemory;
+
+    sp<AudioTrack> mTrack;
+};
+
+// hold pcm data sent by AudioRecord
+class RawBuffer {
+  public:
+    RawBuffer(int64_t ptsPipeline = -1, int64_t ptsManual = -1, int32_t capacity = 0);
+
+    std::unique_ptr<uint8_t[]> mData;
+    int64_t mPtsPipeline;
+    int64_t mPtsManual;
+    int32_t mCapacity;
+};
+
+// Simple AudioCapture
+class AudioCapture : public AudioRecord::IAudioRecordCallback {
+  public:
+    AudioCapture(audio_source_t inputSource, uint32_t sampleRate, audio_format_t format,
+                 audio_channel_mask_t channelMask,
+                 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
+                 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
+                 AudioRecord::transfer_type transferType = AudioRecord::TRANSFER_CALLBACK);
+    ~AudioCapture();
+    size_t onMoreData(const AudioRecord::Buffer& buffer) override;
+    void onOverrun() override;
+    void onMarker(uint32_t markerPosition) override;
+    void onNewPos(uint32_t newPos) override;
+    void onNewIAudioRecord() override;
+    status_t create();
+    sp<AudioRecord> getAudioRecordHandle();
+    status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
+                   audio_session_t triggerSession = AUDIO_SESSION_NONE);
+    status_t obtainBufferCb(RawBuffer& buffer);
+    status_t obtainBuffer(RawBuffer& buffer);
+    status_t audioProcess();
+    status_t stop();
+
+    uint32_t mFrameCount;
+    uint32_t mNotificationFrames;
+    int64_t mNumFramesToRecord;
+    int64_t mNumFramesReceived;
+    int64_t mNumFramesLost;
+    uint32_t mMarkerPosition;
+    uint32_t mMarkerPeriod;
+    uint32_t mReceivedCbMarkerAtPosition;
+    uint32_t mReceivedCbMarkerCount;
+    bool mBufferOverrun;
+
+    enum State {
+        REC_NO_INIT,
+        REC_READY,
+        REC_STARTED,
+        REC_STOPPED,
+    };
+
+  private:
+    const audio_source_t mInputSource;
+    const uint32_t mSampleRate;
+    const audio_format_t mFormat;
+    const audio_channel_mask_t mChannelMask;
+    const audio_input_flags_t mFlags;
+    const audio_session_t mSessionId;
+    const AudioRecord::transfer_type mTransferType;
+
+    size_t mMaxBytesPerCallback = 2048;
+    sp<AudioRecord> mRecord;
+    State mState;
+    bool mStopRecording;
+    int mOutFileFd = -1;
+
+    std::mutex mMutex;
+    std::condition_variable mCondition;
+    std::deque<RawBuffer> mBuffersReceived;
+};
+
+#endif  // AUDIO_TEST_UTILS_H_
diff --git a/media/libaudioclient/tests/audioclient_serialization_tests.cpp b/media/libaudioclient/tests/audioclient_serialization_tests.cpp
new file mode 100644
index 0000000..93baefd6
--- /dev/null
+++ b/media/libaudioclient/tests/audioclient_serialization_tests.cpp
@@ -0,0 +1,310 @@
+/*
+ * Copyright (C) 2021 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 "AudioClientSerializationUnitTests"
+
+#include <cstdint>
+#include <cstdlib>
+#include <ctime>
+
+#include <gtest/gtest.h>
+
+#include <android_audio_policy_configuration_V7_0-enums.h>
+#include <xsdc/XsdcSupport.h>
+
+#include "audio_test_utils.h"
+
+using namespace android;
+namespace xsd {
+using namespace ::android::audio::policy::configuration::V7_0;
+}
+
+template <typename T, typename X, typename FUNC>
+std::vector<T> getFlags(const xsdc_enum_range<X>& range, const FUNC& func,
+                        const std::string& findString = {}) {
+    std::vector<T> vec;
+    for (const auto& xsdEnumVal : range) {
+        T enumVal;
+        std::string enumString = toString(xsdEnumVal);
+        if (enumString.find(findString) != std::string::npos &&
+            func(enumString.c_str(), &enumVal)) {
+            vec.push_back(enumVal);
+        }
+    }
+    return vec;
+}
+
+static const std::vector<audio_usage_t> kUsages =
+        getFlags<audio_usage_t, xsd::AudioUsage, decltype(audio_usage_from_string)>(
+                xsdc_enum_range<xsd::AudioUsage>{}, audio_usage_from_string);
+
+static const std::vector<audio_content_type_t> kContentType =
+        getFlags<audio_content_type_t, xsd::AudioContentType,
+                 decltype(audio_content_type_from_string)>(xsdc_enum_range<xsd::AudioContentType>{},
+                                                           audio_content_type_from_string);
+
+static const std::vector<audio_source_t> kInputSources =
+        getFlags<audio_source_t, xsd::AudioSource, decltype(audio_source_from_string)>(
+                xsdc_enum_range<xsd::AudioSource>{}, audio_source_from_string);
+
+static const std::vector<audio_stream_type_t> kStreamtypes =
+        getFlags<audio_stream_type_t, xsd::AudioStreamType,
+                 decltype(audio_stream_type_from_string)>(xsdc_enum_range<xsd::AudioStreamType>{},
+                                                          audio_stream_type_from_string);
+
+static const std::vector<uint32_t> kMixMatchRules = {
+        RULE_MATCH_ATTRIBUTE_USAGE,
+        RULE_EXCLUDE_ATTRIBUTE_USAGE,
+        RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET,
+        RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET,
+        RULE_MATCH_UID,
+        RULE_EXCLUDE_UID,
+        RULE_MATCH_USERID,
+        RULE_EXCLUDE_USERID,
+};
+
+// Generates a random string.
+std::string CreateRandomString(size_t n) {
+    std::string data =
+            "abcdefghijklmnopqrstuvwxyz"
+            "ABCDEFGHIJKLMNOPQRSTUVWXYZ"
+            "0123456789";
+    srand(static_cast<unsigned int>(time(0)));
+    std::string s(n, ' ');
+    for (size_t i = 0; i < n; ++i) {
+        s[i] = data[rand() % data.size()];
+    }
+    return s;
+}
+
+class FillAudioAttributes {
+  public:
+    void fillAudioAttributes(audio_attributes_t& attr);
+
+    unsigned int mSeed;
+};
+
+void FillAudioAttributes::fillAudioAttributes(audio_attributes_t& attr) {
+    attr.content_type = kContentType[rand() % kContentType.size()];
+    attr.usage = kUsages[rand() % kUsages.size()];
+    attr.source = kInputSources[rand() % kInputSources.size()];
+    // attr.flags -> [0, (1 << (CAPTURE_PRIVATE + 1) - 1)]
+    attr.flags = static_cast<audio_flags_mask_t>(rand() & 0x3fff);
+    sprintf(attr.tags, "%s",
+            CreateRandomString((int)rand() % (AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - 1)).c_str());
+}
+
+class SerializationTest : public FillAudioAttributes, public ::testing::Test {
+    void SetUp() override {
+        mSeed = static_cast<unsigned int>(time(0));
+        srand(mSeed);
+    }
+};
+
+// UNIT TESTS
+TEST_F(SerializationTest, AudioProductStrategyBinderization) {
+    for (int j = 0; j < 512; j++) {
+        const std::string name{"Test APSBinderization for seed::" + std::to_string(mSeed)};
+        std::vector<AudioAttributes> audioattributesvector;
+        for (auto i = 0; i < 16; i++) {
+            audio_attributes_t attributes;
+            fillAudioAttributes(attributes);
+            AudioAttributes audioattributes{static_cast<volume_group_t>(rand()),
+                                            kStreamtypes[rand() % kStreamtypes.size()], attributes};
+            audioattributesvector.push_back(audioattributes);
+        }
+        product_strategy_t psId = static_cast<product_strategy_t>(rand());
+        AudioProductStrategy aps{name, audioattributesvector, psId};
+
+        Parcel p;
+        EXPECT_EQ(NO_ERROR, aps.writeToParcel(&p)) << name;
+
+        AudioProductStrategy apsCopy;
+        p.setDataPosition(0);
+        EXPECT_EQ(NO_ERROR, apsCopy.readFromParcel(&p)) << name;
+        EXPECT_EQ(apsCopy.getName(), name) << name;
+        EXPECT_EQ(apsCopy.getId(), psId) << name;
+        auto avec = apsCopy.getAudioAttributes();
+        EXPECT_EQ(avec.size(), audioattributesvector.size()) << name;
+        for (int i = 0; i < audioattributesvector.size(); i++) {
+            EXPECT_EQ(avec[i].getGroupId(), audioattributesvector[i].getGroupId()) << name;
+            EXPECT_EQ(avec[i].getStreamType(), audioattributesvector[i].getStreamType()) << name;
+            EXPECT_TRUE(avec[i].getAttributes() == audioattributesvector[i].getAttributes())
+                    << name;
+        }
+    }
+}
+
+TEST_F(SerializationTest, AudioVolumeGroupBinderization) {
+    for (int j = 0; j < 512; j++) {
+        const std::string name{"Test AVGBinderization for seed::" + std::to_string(mSeed)};
+        volume_group_t groupId = static_cast<volume_group_t>(rand());
+        std::vector<audio_attributes_t> attributesvector;
+        for (auto i = 0; i < 16; i++) {
+            audio_attributes_t attributes;
+            fillAudioAttributes(attributes);
+            attributesvector.push_back(attributes);
+        }
+        std::vector<audio_stream_type_t> streamsvector;
+        for (auto i = 0; i < 8; i++) {
+            streamsvector.push_back(kStreamtypes[rand() % kStreamtypes.size()]);
+        }
+        AudioVolumeGroup avg{name, groupId, attributesvector, streamsvector};
+
+        Parcel p;
+        EXPECT_EQ(NO_ERROR, avg.writeToParcel(&p));
+
+        AudioVolumeGroup avgCopy;
+        p.setDataPosition(0);
+        EXPECT_EQ(NO_ERROR, avgCopy.readFromParcel(&p)) << name;
+        EXPECT_EQ(avgCopy.getName(), name) << name;
+        EXPECT_EQ(avgCopy.getId(), groupId) << name;
+        auto avec = avgCopy.getAudioAttributes();
+        EXPECT_EQ(avec.size(), attributesvector.size()) << name;
+        for (int i = 0; i < avec.size(); i++) {
+            EXPECT_TRUE(avec[i] == attributesvector[i]) << name;
+        }
+        StreamTypeVector svec = avgCopy.getStreamTypes();
+        EXPECT_EQ(svec.size(), streamsvector.size()) << name;
+        for (int i = 0; i < svec.size(); i++) {
+            EXPECT_EQ(svec[i], streamsvector[i]) << name;
+        }
+    }
+}
+
+TEST_F(SerializationTest, AudioMixBinderization) {
+    for (int j = 0; j < 512; j++) {
+        const std::string msg{"Test AMBinderization for seed::" + std::to_string(mSeed)};
+        Vector<AudioMixMatchCriterion> criteria;
+        for (int i = 0; i < 16; i++) {
+            AudioMixMatchCriterion ammc{kUsages[rand() % kUsages.size()],
+                                        kInputSources[rand() % kInputSources.size()],
+                                        kMixMatchRules[rand() % kMixMatchRules.size()]};
+            criteria.add(ammc);
+        }
+        audio_config_t config{};
+        config.sample_rate = 48000;
+        config.channel_mask = AUDIO_CHANNEL_IN_MONO;
+        config.format = AUDIO_FORMAT_PCM_16_BIT;
+        config.offload_info = AUDIO_INFO_INITIALIZER;
+        config.frame_count = 4800;
+        AudioMix am{criteria,
+                    static_cast<uint32_t>(rand()),
+                    config,
+                    static_cast<uint32_t>(rand()),
+                    String8(msg.c_str()),
+                    static_cast<uint32_t>(rand())};
+
+        Parcel p;
+        EXPECT_EQ(NO_ERROR, am.writeToParcel(&p)) << msg;
+
+        AudioMix amCopy;
+        p.setDataPosition(0);
+        EXPECT_EQ(NO_ERROR, amCopy.readFromParcel(&p)) << msg;
+        EXPECT_EQ(amCopy.mMixType, am.mMixType) << msg;
+        EXPECT_EQ(amCopy.mFormat.sample_rate, am.mFormat.sample_rate) << msg;
+        EXPECT_EQ(amCopy.mFormat.channel_mask, am.mFormat.channel_mask) << msg;
+        EXPECT_EQ(amCopy.mFormat.format, am.mFormat.format) << msg;
+        EXPECT_EQ(amCopy.mRouteFlags, am.mRouteFlags) << msg;
+        EXPECT_EQ(amCopy.mDeviceAddress, am.mDeviceAddress) << msg;
+        EXPECT_EQ(amCopy.mCbFlags, am.mCbFlags) << msg;
+        EXPECT_EQ(amCopy.mCriteria.size(), am.mCriteria.size()) << msg;
+        for (auto i = 0; i < amCopy.mCriteria.size(); i++) {
+            EXPECT_EQ(amCopy.mCriteria[i].mRule, am.mCriteria[i].mRule) << msg;
+            EXPECT_EQ(amCopy.mCriteria[i].mValue.mUserId, am.mCriteria[i].mValue.mUserId) << msg;
+        }
+    }
+}
+
+using MMCTestParams = std::tuple<audio_usage_t, audio_source_t, uint32_t>;
+
+class MMCParameterizedTest : public FillAudioAttributes,
+                             public ::testing::TestWithParam<MMCTestParams> {
+  public:
+    MMCParameterizedTest()
+        : mAudioUsage(std::get<0>(GetParam())),
+          mAudioSource(std::get<1>(GetParam())),
+          mAudioMixMatchRules(std::get<2>(GetParam())){};
+
+    const audio_usage_t mAudioUsage;
+    const audio_source_t mAudioSource;
+    const uint32_t mAudioMixMatchRules;
+
+    void SetUp() override {
+        mSeed = static_cast<unsigned int>(time(0));
+        srand(mSeed);
+    }
+};
+
+TEST_P(MMCParameterizedTest, AudioMixMatchCriterionBinderization) {
+    const std::string msg{"Test AMMCBinderization for seed::" + std::to_string(mSeed)};
+    AudioMixMatchCriterion ammc{mAudioUsage, mAudioSource, mAudioMixMatchRules};
+
+    Parcel p;
+    EXPECT_EQ(NO_ERROR, ammc.writeToParcel(&p)) << msg;
+
+    AudioMixMatchCriterion ammcCopy;
+    p.setDataPosition(0);
+    EXPECT_EQ(NO_ERROR, ammcCopy.readFromParcel(&p)) << msg;
+    EXPECT_EQ(ammcCopy.mRule, ammc.mRule) << msg;
+    EXPECT_EQ(ammcCopy.mValue.mUserId, ammc.mValue.mUserId) << msg;
+}
+
+// audioUsage, audioSource, audioMixMatchRules
+INSTANTIATE_TEST_SUITE_P(SerializationParameterizedTests, MMCParameterizedTest,
+                         ::testing::Combine(testing::ValuesIn(kUsages),
+                                            testing::ValuesIn(kInputSources),
+                                            testing::ValuesIn(kMixMatchRules)));
+
+using AudioAttributesTestParams = std::tuple<audio_stream_type_t>;
+
+class AudioAttributesParameterizedTest
+    : public FillAudioAttributes,
+      public ::testing::TestWithParam<AudioAttributesTestParams> {
+  public:
+    AudioAttributesParameterizedTest() : mAudioStream(std::get<0>(GetParam())){};
+
+    const audio_stream_type_t mAudioStream;
+
+    void SetUp() override {
+        mSeed = static_cast<unsigned int>(time(0));
+        srand(mSeed);
+    }
+};
+
+TEST_P(AudioAttributesParameterizedTest, AudioAttributesBinderization) {
+    const std::string msg{"Test AABinderization for seed::" + std::to_string(mSeed)};
+    volume_group_t groupId = static_cast<volume_group_t>(rand());
+    audio_stream_type_t stream = mAudioStream;
+    audio_attributes_t attributes;
+    fillAudioAttributes(attributes);
+    AudioAttributes audioattributes{groupId, stream, attributes};
+
+    Parcel p;
+    EXPECT_EQ(NO_ERROR, audioattributes.writeToParcel(&p)) << msg;
+
+    AudioAttributes audioattributesCopy;
+    p.setDataPosition(0);
+    EXPECT_EQ(NO_ERROR, audioattributesCopy.readFromParcel(&p)) << msg;
+    EXPECT_EQ(audioattributesCopy.getGroupId(), audioattributes.getGroupId()) << msg;
+    EXPECT_EQ(audioattributesCopy.getStreamType(), audioattributes.getStreamType()) << msg;
+    EXPECT_TRUE(audioattributesCopy.getAttributes() == attributes) << msg;
+}
+
+// audioStream
+INSTANTIATE_TEST_SUITE_P(SerializationParameterizedTests, AudioAttributesParameterizedTest,
+                         ::testing::Combine(testing::ValuesIn(kStreamtypes)));
diff --git a/media/libaudioclient/tests/audioeffect_tests.cpp b/media/libaudioclient/tests/audioeffect_tests.cpp
new file mode 100644
index 0000000..93fe306
--- /dev/null
+++ b/media/libaudioclient/tests/audioeffect_tests.cpp
@@ -0,0 +1,335 @@
+/*
+ * Copyright (C) 2021 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 "AudioEffectUnitTests"
+
+#include <gtest/gtest.h>
+#include <media/AudioEffect.h>
+#include <system/audio_effects/effect_visualizer.h>
+
+#include "audio_test_utils.h"
+
+using namespace android;
+
+static constexpr int kDefaultInputEffectPriority = -1;
+static constexpr int kDefaultOutputEffectPriority = 0;
+
+static const char* gPackageName = "AudioEffectTest";
+
+bool isEffectExistsOnAudioSession(const effect_uuid_t* type, int priority,
+                                  audio_session_t sessionId) {
+    std::string packageName{gPackageName};
+    AttributionSourceState attributionSource;
+    attributionSource.packageName = packageName;
+    attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
+    attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
+    attributionSource.token = sp<BBinder>::make();
+    sp<AudioEffect> effect = new AudioEffect(attributionSource);
+    effect->set(type, nullptr /* uid */, priority, nullptr /* callback */, sessionId);
+    return effect->initCheck() == ALREADY_EXISTS;
+}
+
+bool isEffectDefaultOnRecord(const effect_uuid_t* type, const sp<AudioRecord>& audioRecord) {
+    effect_descriptor_t descriptors[AudioEffect::kMaxPreProcessing];
+    uint32_t numEffects = AudioEffect::kMaxPreProcessing;
+    status_t ret = AudioEffect::queryDefaultPreProcessing(audioRecord->getSessionId(), descriptors,
+                                                          &numEffects);
+    if (ret != OK) {
+        return false;
+    }
+    for (int i = 0; i < numEffects; i++) {
+        if (memcmp(&descriptors[i].type, type, sizeof(effect_uuid_t)) == 0) {
+            return true;
+        }
+    }
+    return false;
+}
+
+void listEffectsAvailable(std::vector<effect_descriptor_t>& descriptors) {
+    uint32_t numEffects = 0;
+    if (NO_ERROR == AudioEffect::queryNumberEffects(&numEffects)) {
+        for (auto i = 0; i < numEffects; i++) {
+            effect_descriptor_t des;
+            if (NO_ERROR == AudioEffect::queryEffect(i, &des)) descriptors.push_back(des);
+        }
+    }
+}
+
+bool isPreprocessing(effect_descriptor_t& descriptor) {
+    return ((descriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC);
+}
+
+bool isInsert(effect_descriptor_t& descriptor) {
+    return ((descriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT);
+}
+
+bool isAux(effect_descriptor_t& descriptor) {
+    return ((descriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY);
+}
+
+bool isFastCompatible(effect_descriptor_t& descriptor) {
+    return !(((descriptor.flags & EFFECT_FLAG_HW_ACC_MASK) == 0) &&
+             ((descriptor.flags & EFFECT_FLAG_NO_PROCESS) == 0));
+}
+
+// UNIT TESTS
+TEST(AudioEffectTest, getEffectDescriptor) {
+    effect_uuid_t randomType = {
+            0x81781c08, 0x93dd, 0x11ec, 0xb909, {0x02, 0x42, 0xac, 0x12, 0x00, 0x02}};
+    effect_uuid_t randomUuid = {
+            0x653730e1, 0x1be1, 0x438e, 0xa35a, {0xfc, 0x9b, 0xa1, 0x2a, 0x5e, 0xc9}};
+    effect_uuid_t empty = EFFECT_UUID_INITIALIZER;
+
+    effect_descriptor_t descriptor;
+    EXPECT_EQ(NAME_NOT_FOUND, AudioEffect::getEffectDescriptor(&randomUuid, &randomType,
+                                                               EFFECT_FLAG_TYPE_MASK, &descriptor));
+
+    std::vector<effect_descriptor_t> descriptors;
+    listEffectsAvailable(descriptors);
+
+    for (auto i = 0; i < descriptors.size(); i++) {
+        EXPECT_EQ(NO_ERROR,
+                  AudioEffect::getEffectDescriptor(&descriptors[i].uuid, &descriptors[i].type,
+                                                   EFFECT_FLAG_TYPE_MASK, &descriptor));
+        EXPECT_EQ(0, memcmp(&descriptor, &descriptors[i], sizeof(effect_uuid_t)));
+    }
+    // negative tests
+    if (descriptors.size() > 0) {
+        EXPECT_EQ(BAD_VALUE,
+                  AudioEffect::getEffectDescriptor(&descriptors[0].uuid, &descriptors[0].type,
+                                                   EFFECT_FLAG_TYPE_MASK, nullptr));
+    }
+    EXPECT_EQ(BAD_VALUE, AudioEffect::getEffectDescriptor(nullptr, nullptr,
+                                                          EFFECT_FLAG_TYPE_PRE_PROC, &descriptor));
+    EXPECT_EQ(BAD_VALUE, AudioEffect::getEffectDescriptor(&empty, &randomType,
+                                                          EFFECT_FLAG_TYPE_MASK, nullptr));
+    EXPECT_EQ(BAD_VALUE, AudioEffect::getEffectDescriptor(nullptr, &randomType,
+                                                          EFFECT_FLAG_TYPE_POST_PROC, &descriptor));
+    EXPECT_EQ(BAD_VALUE, AudioEffect::getEffectDescriptor(&randomUuid, nullptr,
+                                                          EFFECT_FLAG_TYPE_INSERT, &descriptor));
+}
+
+TEST(AudioEffectTest, DISABLED_GetSetParameterForEffect) {
+    std::string packageName{gPackageName};
+    AttributionSourceState attributionSource;
+    attributionSource.packageName = packageName;
+    attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
+    attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
+    attributionSource.token = sp<BBinder>::make();
+    sp<AudioEffect> visualizer = new AudioEffect(attributionSource);
+    ASSERT_NE(visualizer, nullptr) << "effect not created";
+    visualizer->set(SL_IID_VISUALIZATION);
+    status_t status = visualizer->initCheck();
+    ASSERT_TRUE(status == NO_ERROR || status == ALREADY_EXISTS) << "Init check error";
+    ASSERT_EQ(NO_ERROR, visualizer->setEnabled(true)) << "visualizer not enabled";
+
+    uint32_t buf32[3][sizeof(effect_param_t) / sizeof(uint32_t) + 2];
+    effect_param_t* vis_none = (effect_param_t*)(buf32[0]);
+    effect_param_t* vis_rms = (effect_param_t*)(buf32[1]);
+    effect_param_t* vis_tmp = (effect_param_t*)(buf32[2]);
+
+    // Visualizer::setMeasurementMode()
+    vis_none->psize = sizeof(uint32_t);
+    vis_none->vsize = sizeof(uint32_t);
+    *(int32_t*)vis_none->data = VISUALIZER_PARAM_MEASUREMENT_MODE;
+    *((int32_t*)vis_none->data + 1) = MEASUREMENT_MODE_NONE;
+    EXPECT_EQ(NO_ERROR, visualizer->setParameter(vis_none))
+            << "setMeasurementMode doesn't report success";
+
+    // Visualizer::getMeasurementMode()
+    vis_tmp->psize = sizeof(uint32_t);
+    vis_tmp->vsize = sizeof(uint32_t);
+    *(int32_t*)vis_tmp->data = VISUALIZER_PARAM_MEASUREMENT_MODE;
+    *((int32_t*)vis_tmp->data + 1) = 23;
+    EXPECT_EQ(NO_ERROR, visualizer->getParameter(vis_tmp))
+            << "getMeasurementMode doesn't report success";
+    EXPECT_EQ(*((int32_t*)vis_tmp->data + 1), *((int32_t*)vis_none->data + 1))
+            << "target mode does not match set mode";
+
+    // Visualizer::setMeasurementModeDeferred()
+    vis_rms->psize = sizeof(uint32_t);
+    vis_rms->vsize = sizeof(uint32_t);
+    *(int32_t*)vis_rms->data = VISUALIZER_PARAM_MEASUREMENT_MODE;
+    *((int32_t*)vis_rms->data + 1) = MEASUREMENT_MODE_PEAK_RMS;
+    EXPECT_EQ(NO_ERROR, visualizer->setParameterDeferred(vis_rms))
+            << "setMeasurementModeDeferred doesn't report success";
+
+    *((int32_t*)vis_tmp->data + 1) = 23;
+    EXPECT_EQ(NO_ERROR, visualizer->getParameter(vis_tmp))
+            << "getMeasurementMode doesn't report success";
+    EXPECT_EQ(*((int32_t*)vis_tmp->data + 1), *((int32_t*)vis_none->data + 1))
+            << "target mode does not match set mode";
+
+    // setParameterCommit
+    EXPECT_EQ(NO_ERROR, visualizer->setParameterCommit())
+            << "setMeasurementModeCommit does not report success";
+
+    // validate Params
+    *((int32_t*)vis_tmp->data + 1) = 23;
+    EXPECT_EQ(NO_ERROR, visualizer->getParameter(vis_tmp))
+            << "getMeasurementMode doesn't report success";
+    EXPECT_EQ(*((int32_t*)vis_tmp->data + 1), *((int32_t*)vis_rms->data + 1))
+            << "target mode does not match set mode";
+}
+
+TEST(AudioEffectTest, ManageSourceDefaultEffects) {
+    int32_t selectedEffect = -1;
+
+    const uint32_t sampleRate = 44100;
+    const audio_format_t format = AUDIO_FORMAT_PCM_16_BIT;
+    const audio_channel_mask_t channelMask = AUDIO_CHANNEL_IN_STEREO;
+    sp<AudioCapture> capture = nullptr;
+
+    std::vector<effect_descriptor_t> descriptors;
+    listEffectsAvailable(descriptors);
+    for (auto i = 0; i < descriptors.size(); i++) {
+        if (isPreprocessing(descriptors[i])) {
+            capture = new AudioCapture(AUDIO_SOURCE_MIC, sampleRate, format, channelMask);
+            ASSERT_NE(capture, nullptr) << "Unable to create Record Application";
+            EXPECT_EQ(NO_ERROR, capture->create());
+            EXPECT_EQ(NO_ERROR, capture->start());
+            if (!isEffectDefaultOnRecord(&descriptors[i].type, capture->getAudioRecordHandle())) {
+                selectedEffect = i;
+                break;
+            }
+        }
+    }
+    if (selectedEffect == -1) GTEST_SKIP() << " expected at least one preprocessing effect";
+    effect_uuid_t selectedEffectType = descriptors[selectedEffect].type;
+
+    char type[512];
+    AudioEffect::guidToString(&selectedEffectType, type, sizeof(type));
+
+    capture = new AudioCapture(AUDIO_SOURCE_MIC, sampleRate, format, channelMask);
+    ASSERT_NE(capture, nullptr) << "Unable to create Record Application";
+    EXPECT_EQ(NO_ERROR, capture->create());
+    EXPECT_EQ(NO_ERROR, capture->start());
+    EXPECT_FALSE(isEffectDefaultOnRecord(&selectedEffectType, capture->getAudioRecordHandle()))
+            << "Effect should not have been default on record. " << type;
+    EXPECT_FALSE(isEffectExistsOnAudioSession(&selectedEffectType, kDefaultInputEffectPriority - 1,
+                                              capture->getAudioRecordHandle()->getSessionId()))
+            << "Effect should not have been added. " << type;
+    EXPECT_EQ(OK, capture->audioProcess());
+    EXPECT_EQ(OK, capture->stop());
+
+    String16 name{gPackageName};
+    audio_unique_id_t effectId;
+    status_t status = AudioEffect::addSourceDefaultEffect(
+            type, name, nullptr, kDefaultInputEffectPriority, AUDIO_SOURCE_MIC, &effectId);
+    EXPECT_EQ(NO_ERROR, status) << "Adding default effect failed: " << type;
+
+    capture = new AudioCapture(AUDIO_SOURCE_MIC, sampleRate, format, channelMask);
+    ASSERT_NE(capture, nullptr) << "Unable to create Record Application";
+    EXPECT_EQ(NO_ERROR, capture->create());
+    EXPECT_EQ(NO_ERROR, capture->start());
+    EXPECT_TRUE(isEffectDefaultOnRecord(&selectedEffectType, capture->getAudioRecordHandle()))
+            << "Effect should have been default on record. " << type;
+    EXPECT_TRUE(isEffectExistsOnAudioSession(&selectedEffectType, kDefaultInputEffectPriority - 1,
+                                             capture->getAudioRecordHandle()->getSessionId()))
+            << "Effect should have been added. " << type;
+    EXPECT_EQ(OK, capture->audioProcess());
+    EXPECT_EQ(OK, capture->stop());
+
+    status = AudioEffect::removeSourceDefaultEffect(effectId);
+    EXPECT_EQ(NO_ERROR, status);
+    capture = new AudioCapture(AUDIO_SOURCE_MIC, sampleRate, format, channelMask);
+    ASSERT_NE(capture, nullptr) << "Unable to create Record Application";
+    EXPECT_EQ(NO_ERROR, capture->create());
+    EXPECT_EQ(NO_ERROR, capture->start());
+    EXPECT_FALSE(isEffectDefaultOnRecord(&selectedEffectType, capture->getAudioRecordHandle()))
+            << "Effect should not have been default on record. " << type;
+    EXPECT_FALSE(isEffectExistsOnAudioSession(&selectedEffectType, kDefaultInputEffectPriority - 1,
+                                              capture->getAudioRecordHandle()->getSessionId()))
+            << "Effect should not have been added. " << type;
+    EXPECT_EQ(OK, capture->audioProcess());
+    EXPECT_EQ(OK, capture->stop());
+}
+
+TEST(AudioEffectTest, ManageStreamDefaultEffects) {
+    int32_t selectedEffect = -1;
+
+    std::vector<effect_descriptor_t> descriptors;
+    listEffectsAvailable(descriptors);
+    for (auto i = 0; i < descriptors.size(); i++) {
+        if (isAux(descriptors[i])) {
+            selectedEffect = i;
+            break;
+        }
+    }
+    if (selectedEffect == -1) GTEST_SKIP() << " expected at least one Aux effect";
+    effect_uuid_t* selectedEffectType = &descriptors[selectedEffect].type;
+
+    char type[512];
+    AudioEffect::guidToString(selectedEffectType, type, sizeof(type));
+    // create track
+    audio_attributes_t attributes;
+    attributes.usage = AUDIO_USAGE_MEDIA;
+    attributes.content_type = AUDIO_CONTENT_TYPE_MUSIC;
+    auto playback = sp<AudioPlayback>::make(
+            44100 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+            AUDIO_OUTPUT_FLAG_NONE, AUDIO_SESSION_NONE, AudioTrack::TRANSFER_SHARED, &attributes);
+    ASSERT_NE(nullptr, playback);
+    ASSERT_EQ(NO_ERROR, playback->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"));
+    EXPECT_EQ(NO_ERROR, playback->create());
+    EXPECT_EQ(NO_ERROR, playback->start());
+    EXPECT_FALSE(isEffectExistsOnAudioSession(selectedEffectType, kDefaultOutputEffectPriority - 1,
+                                              playback->getAudioTrackHandle()->getSessionId()))
+            << "Effect should not have been added. " << type;
+    EXPECT_EQ(NO_ERROR, playback->waitForConsumption());
+    playback->stop();
+    playback.clear();
+
+    String16 name{gPackageName};
+    audio_unique_id_t id;
+    status_t status = AudioEffect::addStreamDefaultEffect(
+            type, name, nullptr, kDefaultOutputEffectPriority, AUDIO_USAGE_MEDIA, &id);
+    EXPECT_EQ(NO_ERROR, status) << "Adding default effect failed: " << type;
+
+    playback = sp<AudioPlayback>::make(
+            44100 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+            AUDIO_OUTPUT_FLAG_NONE, AUDIO_SESSION_NONE, AudioTrack::TRANSFER_SHARED, &attributes);
+    ASSERT_NE(nullptr, playback);
+    ASSERT_EQ(NO_ERROR, playback->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"));
+    EXPECT_EQ(NO_ERROR, playback->create());
+    float level = 0.2f, levelGot;
+    playback->getAudioTrackHandle()->setAuxEffectSendLevel(level);
+    EXPECT_EQ(NO_ERROR, playback->start());
+    EXPECT_TRUE(isEffectExistsOnAudioSession(selectedEffectType, kDefaultOutputEffectPriority - 1,
+                                             playback->getAudioTrackHandle()->getSessionId()))
+            << "Effect should have been added. " << type;
+    EXPECT_EQ(NO_ERROR, playback->waitForConsumption());
+    playback->getAudioTrackHandle()->getAuxEffectSendLevel(&levelGot);
+    EXPECT_EQ(level, levelGot);
+    playback->stop();
+    playback.clear();
+
+    status = AudioEffect::removeStreamDefaultEffect(id);
+    EXPECT_EQ(NO_ERROR, status);
+    playback = sp<AudioPlayback>::make(
+            44100 /*sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+            AUDIO_OUTPUT_FLAG_NONE, AUDIO_SESSION_NONE, AudioTrack::TRANSFER_SHARED, &attributes);
+    ASSERT_NE(nullptr, playback);
+    ASSERT_EQ(NO_ERROR, playback->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"));
+    EXPECT_EQ(NO_ERROR, playback->create());
+    EXPECT_EQ(NO_ERROR, playback->start());
+    EXPECT_FALSE(isEffectExistsOnAudioSession(selectedEffectType, kDefaultOutputEffectPriority - 1,
+                                              playback->getAudioTrackHandle()->getSessionId()))
+            << "Effect should not have been added. " << type;
+    EXPECT_EQ(NO_ERROR, playback->waitForConsumption());
+    playback->stop();
+    playback.clear();
+}
diff --git a/media/libaudioclient/tests/audiorecord_tests.cpp b/media/libaudioclient/tests/audiorecord_tests.cpp
new file mode 100644
index 0000000..754e6cc
--- /dev/null
+++ b/media/libaudioclient/tests/audiorecord_tests.cpp
@@ -0,0 +1,235 @@
+/*
+ * Copyright (C) 2021 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 "AudioRecordTest"
+
+#include <gtest/gtest.h>
+
+#include "audio_test_utils.h"
+
+using namespace android;
+
+class AudioRecordTest : public ::testing::Test {
+  public:
+    virtual void SetUp() override {
+        mAC = new AudioCapture(AUDIO_SOURCE_DEFAULT, 44100, AUDIO_FORMAT_PCM_16_BIT,
+                               AUDIO_CHANNEL_IN_FRONT);
+        ASSERT_NE(nullptr, mAC);
+        ASSERT_EQ(OK, mAC->create()) << "record creation failed";
+    }
+
+    virtual void TearDown() override {
+        if (mAC) ASSERT_EQ(OK, mAC->stop());
+    }
+
+    sp<AudioCapture> mAC;
+};
+
+class AudioRecordCreateTest
+    : public ::testing::TestWithParam<
+              std::tuple<uint32_t, audio_format_t, audio_channel_mask_t, audio_input_flags_t,
+                         audio_session_t, audio_source_t>> {
+  public:
+    AudioRecordCreateTest()
+        : mSampleRate(std::get<0>(GetParam())),
+          mFormat(std::get<1>(GetParam())),
+          mChannelMask(std::get<2>(GetParam())),
+          mFlags(std::get<3>(GetParam())),
+          mSessionId(std::get<4>(GetParam())),
+          mInputSource(std::get<5>(GetParam())){};
+
+    const uint32_t mSampleRate;
+    const audio_format_t mFormat;
+    const audio_channel_mask_t mChannelMask;
+    const audio_input_flags_t mFlags;
+    const audio_session_t mSessionId;
+    const audio_source_t mInputSource;
+    const AudioRecord::transfer_type mTransferType = AudioRecord::TRANSFER_OBTAIN;
+
+    sp<AudioCapture> mAC;
+
+    virtual void SetUp() override {
+        mAC = new AudioCapture(mInputSource, mSampleRate, mFormat, mChannelMask, mFlags, mSessionId,
+                               mTransferType);
+        ASSERT_NE(nullptr, mAC);
+        ASSERT_EQ(OK, mAC->create()) << "record creation failed";
+    }
+
+    virtual void TearDown() override {
+        if (mAC) ASSERT_EQ(OK, mAC->stop());
+    }
+};
+
+TEST_F(AudioRecordTest, TestSimpleRecord) {
+    EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+}
+
+TEST_F(AudioRecordTest, TestAudioCbNotifier) {
+    EXPECT_EQ(BAD_VALUE, mAC->getAudioRecordHandle()->addAudioDeviceCallback(nullptr));
+    sp<OnAudioDeviceUpdateNotifier> cb = new OnAudioDeviceUpdateNotifier();
+    sp<OnAudioDeviceUpdateNotifier> cbOld = new OnAudioDeviceUpdateNotifier();
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->addAudioDeviceCallback(cbOld));
+    EXPECT_EQ(INVALID_OPERATION, mAC->getAudioRecordHandle()->addAudioDeviceCallback(cbOld));
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->addAudioDeviceCallback(cb));
+    EXPECT_EQ(OK, mAC->start()) << "record creation failed";
+    EXPECT_EQ(OK, cb->waitForAudioDeviceCb());
+    EXPECT_EQ(AUDIO_IO_HANDLE_NONE, cbOld->mAudioIo);
+    EXPECT_EQ(AUDIO_PORT_HANDLE_NONE, cbOld->mDeviceId);
+    EXPECT_NE(AUDIO_IO_HANDLE_NONE, cb->mAudioIo);
+    EXPECT_NE(AUDIO_PORT_HANDLE_NONE, cb->mDeviceId);
+    EXPECT_EQ(BAD_VALUE, mAC->getAudioRecordHandle()->removeAudioDeviceCallback(nullptr));
+    EXPECT_EQ(INVALID_OPERATION, mAC->getAudioRecordHandle()->removeAudioDeviceCallback(cbOld));
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->removeAudioDeviceCallback(cb));
+    mAC->stop();
+}
+
+TEST_F(AudioRecordTest, TestEventRecordTrackPause) {
+    const auto playback = sp<AudioPlayback>::make(
+            8000 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_MONO);
+    ASSERT_EQ(OK, playback->loadResource("/data/local/tmp/bbb_1ch_8kHz_s16le.raw"))
+            << "Unable to open Resource";
+    EXPECT_EQ(OK, playback->create()) << "AudioTrack Creation failed";
+    audio_session_t audioTrackSession = playback->getAudioTrackHandle()->getSessionId();
+    EXPECT_EQ(OK, mAC->start(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE, audioTrackSession))
+            << "record creation failed";
+    EXPECT_EQ(OK, playback->start());
+    RawBuffer buffer;
+    status_t status = mAC->obtainBufferCb(buffer);
+    EXPECT_EQ(status, TIMED_OUT) << "Not expecting any callbacks until track sends Sync event";
+    playback->getAudioTrackHandle()->pause();
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+    playback->stop();
+}
+
+TEST_F(AudioRecordTest, TestEventRecordTrackStop) {
+    const auto playback = sp<AudioPlayback>::make(
+            8000 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_MONO);
+    ASSERT_EQ(OK, playback->loadResource("/data/local/tmp/bbb_1ch_8kHz_s16le.raw"))
+            << "Unable to open Resource";
+    EXPECT_EQ(OK, playback->create()) << "AudioTrack Creation failed";
+    audio_session_t audioTrackSession = playback->getAudioTrackHandle()->getSessionId();
+    EXPECT_EQ(OK, mAC->start(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE, audioTrackSession))
+            << "record creation failed";
+    EXPECT_EQ(OK, playback->start());
+    RawBuffer buffer;
+    status_t status = mAC->obtainBufferCb(buffer);
+    EXPECT_EQ(status, TIMED_OUT) << "Not expecting any callbacks until track sends Sync event";
+    playback->stop();
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+}
+
+TEST_F(AudioRecordTest, TestGetSetMarker) {
+    mAC->mMarkerPosition = (mAC->mNotificationFrames << 3) + (mAC->mNotificationFrames >> 1);
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->setMarkerPosition(mAC->mMarkerPosition))
+            << "setMarkerPosition() failed";
+    uint32_t marker;
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getMarkerPosition(&marker))
+            << "getMarkerPosition() failed";
+    EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+    EXPECT_EQ(marker, mAC->mMarkerPosition)
+            << "configured marker and received marker are different";
+    EXPECT_EQ(mAC->mReceivedCbMarkerAtPosition, mAC->mMarkerPosition)
+            << "configured marker and received cb marker are different";
+}
+
+TEST_F(AudioRecordTest, TestGetSetMarkerPeriodical) {
+    mAC->mMarkerPeriod = (mAC->mNotificationFrames << 3) + (mAC->mNotificationFrames >> 1);
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->setPositionUpdatePeriod(mAC->mMarkerPeriod))
+            << "setPositionUpdatePeriod() failed";
+    uint32_t marker;
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getPositionUpdatePeriod(&marker))
+            << "getPositionUpdatePeriod() failed";
+    EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+    EXPECT_EQ(marker, mAC->mMarkerPeriod) << "configured marker and received marker are different";
+    EXPECT_EQ(mAC->mReceivedCbMarkerCount, mAC->mNumFramesToRecord / mAC->mMarkerPeriod)
+            << "configured marker and received cb marker are different";
+}
+
+TEST_F(AudioRecordTest, TestGetPosition) {
+    uint32_t position;
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getPosition(&position)) << "getPosition() failed";
+    EXPECT_EQ(0, position);
+    EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+    EXPECT_EQ(OK, mAC->stop());
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getPosition(&position)) << "getPosition() failed";
+}
+
+// TODO: Add checkPatchCapture(), verify the information of patch via dumpPort() and dumpPatch()
+TEST_P(AudioRecordCreateTest, TestCreateRecord) {
+    EXPECT_EQ(mFormat, mAC->getAudioRecordHandle()->format());
+    EXPECT_EQ(audio_channel_count_from_in_mask(mChannelMask),
+              mAC->getAudioRecordHandle()->channelCount());
+    if (mAC->mFrameCount != 0)
+        EXPECT_LE(mAC->mFrameCount, mAC->getAudioRecordHandle()->frameCount());
+    EXPECT_EQ(mInputSource, mAC->getAudioRecordHandle()->inputSource());
+    if (mSampleRate != 0) EXPECT_EQ(mSampleRate, mAC->getAudioRecordHandle()->getSampleRate());
+    if (mSessionId != AUDIO_SESSION_NONE)
+        EXPECT_EQ(mSessionId, mAC->getAudioRecordHandle()->getSessionId());
+    if (mTransferType != AudioRecord::TRANSFER_CALLBACK) {
+        uint32_t marker;
+        mAC->mMarkerPosition = (mAC->mNotificationFrames << 3) + (mAC->mNotificationFrames >> 1);
+        EXPECT_EQ(INVALID_OPERATION,
+                  mAC->getAudioRecordHandle()->setMarkerPosition(mAC->mMarkerPosition));
+        EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getMarkerPosition(&marker));
+        EXPECT_EQ(INVALID_OPERATION,
+                  mAC->getAudioRecordHandle()->setPositionUpdatePeriod(mAC->mMarkerPosition));
+        EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getPositionUpdatePeriod(&marker));
+    }
+    EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+}
+
+// for port primary input
+INSTANTIATE_TEST_SUITE_P(AudioRecordPrimaryInput, AudioRecordCreateTest,
+                         ::testing::Combine(::testing::Values(8000, 11025, 12000, 16000, 22050,
+                                                              24000, 32000, 44100, 48000),
+                                            ::testing::Values(AUDIO_FORMAT_PCM_8_24_BIT),
+                                            ::testing::Values(AUDIO_CHANNEL_IN_MONO,
+                                                              AUDIO_CHANNEL_IN_STEREO,
+                                                              AUDIO_CHANNEL_IN_FRONT_BACK),
+                                            ::testing::Values(AUDIO_INPUT_FLAG_NONE),
+                                            ::testing::Values(AUDIO_SESSION_NONE),
+                                            ::testing::Values(AUDIO_SOURCE_DEFAULT)));
+
+// for port fast input
+INSTANTIATE_TEST_SUITE_P(AudioRecordFastInput, AudioRecordCreateTest,
+                         ::testing::Combine(::testing::Values(8000, 11025, 12000, 16000, 22050,
+                                                              24000, 32000, 44100, 48000),
+                                            ::testing::Values(AUDIO_FORMAT_PCM_8_24_BIT),
+                                            ::testing::Values(AUDIO_CHANNEL_IN_MONO,
+                                                              AUDIO_CHANNEL_IN_STEREO,
+                                                              AUDIO_CHANNEL_IN_FRONT_BACK),
+                                            ::testing::Values(AUDIO_INPUT_FLAG_FAST),
+                                            ::testing::Values(AUDIO_SESSION_NONE),
+                                            ::testing::Values(AUDIO_SOURCE_DEFAULT)));
+
+// misc
+INSTANTIATE_TEST_SUITE_P(AudioRecordMiscInput, AudioRecordCreateTest,
+                         ::testing::Combine(::testing::Values(48000),
+                                            ::testing::Values(AUDIO_FORMAT_PCM_16_BIT),
+                                            ::testing::Values(AUDIO_CHANNEL_IN_MONO),
+                                            ::testing::Values(AUDIO_INPUT_FLAG_NONE),
+                                            ::testing::Values(AUDIO_SESSION_NONE),
+                                            ::testing::Values(AUDIO_SOURCE_MIC,
+                                                              AUDIO_SOURCE_CAMCORDER,
+                                                              AUDIO_SOURCE_VOICE_RECOGNITION,
+                                                              AUDIO_SOURCE_VOICE_COMMUNICATION,
+                                                              AUDIO_SOURCE_UNPROCESSED)));
diff --git a/media/libaudioclient/tests/audiorouting_tests.cpp b/media/libaudioclient/tests/audiorouting_tests.cpp
new file mode 100644
index 0000000..32ba597
--- /dev/null
+++ b/media/libaudioclient/tests/audiorouting_tests.cpp
@@ -0,0 +1,253 @@
+/*
+ * Copyright (C) 2021 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
+
+#include <cutils/properties.h>
+#include <gtest/gtest.h>
+#include <libxml/parser.h>
+#include <libxml/xinclude.h>
+#include <string.h>
+#include <system/audio_config.h>
+
+#include "audio_test_utils.h"
+
+using namespace android;
+
+template <class T>
+constexpr void (*xmlDeleter)(T* t);
+template <>
+constexpr auto xmlDeleter<xmlDoc> = xmlFreeDoc;
+template <>
+constexpr auto xmlDeleter<xmlChar> = [](xmlChar* s) { xmlFree(s); };
+
+/** @return a unique_ptr with the correct deleter for the libxml2 object. */
+template <class T>
+constexpr auto make_xmlUnique(T* t) {
+    // Wrap deleter in lambda to enable empty base optimization
+    auto deleter = [](T* t) { xmlDeleter<T>(t); };
+    return std::unique_ptr<T, decltype(deleter)>{t, deleter};
+}
+
+std::string getXmlAttribute(const xmlNode* cur, const char* attribute) {
+    auto charPtr = make_xmlUnique(xmlGetProp(cur, reinterpret_cast<const xmlChar*>(attribute)));
+    if (charPtr == NULL) {
+        return "";
+    }
+    std::string value(reinterpret_cast<const char*>(charPtr.get()));
+    return value;
+}
+
+struct MixPort {
+    std::string name;
+    std::string role;
+    std::string flags;
+};
+
+struct Route {
+    std::string name;
+    std::string sources;
+    std::string sink;
+};
+
+status_t parse_audio_policy_configuration_xml(std::vector<std::string>& attachedDevices,
+                                              std::vector<MixPort>& mixPorts,
+                                              std::vector<Route>& routes) {
+    std::string path = audio_find_readable_configuration_file("audio_policy_configuration.xml");
+    if (path.length() == 0) return UNKNOWN_ERROR;
+    auto doc = make_xmlUnique(xmlParseFile(path.c_str()));
+    if (doc == nullptr) return UNKNOWN_ERROR;
+    xmlNode* root = xmlDocGetRootElement(doc.get());
+    if (root == nullptr) return UNKNOWN_ERROR;
+    if (xmlXIncludeProcess(doc.get()) < 0) return UNKNOWN_ERROR;
+    mixPorts.clear();
+    if (!xmlStrcmp(root->name, reinterpret_cast<const xmlChar*>("audioPolicyConfiguration"))) {
+        std::string raw{getXmlAttribute(root, "version")};
+        for (auto* child = root->xmlChildrenNode; child != nullptr; child = child->next) {
+            if (!xmlStrcmp(child->name, reinterpret_cast<const xmlChar*>("modules"))) {
+                xmlNode* root = child;
+                for (auto* child = root->xmlChildrenNode; child != nullptr; child = child->next) {
+                    if (!xmlStrcmp(child->name, reinterpret_cast<const xmlChar*>("module"))) {
+                        xmlNode* root = child;
+                        for (auto* child = root->xmlChildrenNode; child != nullptr;
+                             child = child->next) {
+                            if (!xmlStrcmp(child->name,
+                                           reinterpret_cast<const xmlChar*>("mixPorts"))) {
+                                xmlNode* root = child;
+                                for (auto* child = root->xmlChildrenNode; child != nullptr;
+                                     child = child->next) {
+                                    if (!xmlStrcmp(child->name,
+                                                   reinterpret_cast<const xmlChar*>("mixPort"))) {
+                                        MixPort mixPort;
+                                        xmlNode* root = child;
+                                        mixPort.name = getXmlAttribute(root, "name");
+                                        mixPort.role = getXmlAttribute(root, "role");
+                                        mixPort.flags = getXmlAttribute(root, "flags");
+                                        if (mixPort.role == "source") mixPorts.push_back(mixPort);
+                                    }
+                                }
+                            } else if (!xmlStrcmp(child->name, reinterpret_cast<const xmlChar*>(
+                                                                       "attachedDevices"))) {
+                                xmlNode* root = child;
+                                for (auto* child = root->xmlChildrenNode; child != nullptr;
+                                     child = child->next) {
+                                    if (!xmlStrcmp(child->name,
+                                                   reinterpret_cast<const xmlChar*>("item"))) {
+                                        auto xmlValue = make_xmlUnique(xmlNodeListGetString(
+                                                child->doc, child->xmlChildrenNode, 1));
+                                        if (xmlValue == nullptr) {
+                                            raw = "";
+                                        } else {
+                                            raw = reinterpret_cast<const char*>(xmlValue.get());
+                                        }
+                                        std::string& value = raw;
+                                        attachedDevices.push_back(std::move(value));
+                                    }
+                                }
+                            } else if (!xmlStrcmp(child->name,
+                                                  reinterpret_cast<const xmlChar*>("routes"))) {
+                                xmlNode* root = child;
+                                for (auto* child = root->xmlChildrenNode; child != nullptr;
+                                     child = child->next) {
+                                    if (!xmlStrcmp(child->name,
+                                                   reinterpret_cast<const xmlChar*>("route"))) {
+                                        Route route;
+                                        xmlNode* root = child;
+                                        route.name = getXmlAttribute(root, "name");
+                                        route.sources = getXmlAttribute(root, "sources");
+                                        route.sink = getXmlAttribute(root, "sink");
+                                        routes.push_back(route);
+                                    }
+                                }
+                            }
+                        }
+                    }
+                }
+            }
+        }
+    }
+    return OK;
+}
+
+// UNIT TEST
+TEST(AudioTrackTest, TestPerformanceMode) {
+    std::vector<std::string> attachedDevices;
+    std::vector<MixPort> mixPorts;
+    std::vector<Route> routes;
+    EXPECT_EQ(OK, parse_audio_policy_configuration_xml(attachedDevices, mixPorts, routes));
+    std::string output_flags_string[] = {"AUDIO_OUTPUT_FLAG_FAST", "AUDIO_OUTPUT_FLAG_DEEP_BUFFER"};
+    audio_output_flags_t output_flags[] = {AUDIO_OUTPUT_FLAG_FAST, AUDIO_OUTPUT_FLAG_DEEP_BUFFER};
+    audio_flags_mask_t flags[] = {AUDIO_FLAG_LOW_LATENCY, AUDIO_FLAG_DEEP_BUFFER};
+    bool hasFlag = false;
+    for (int i = 0; i < sizeof(flags) / sizeof(flags[0]); i++) {
+        hasFlag = false;
+        for (int j = 0; j < mixPorts.size() && !hasFlag; j++) {
+            MixPort port = mixPorts[j];
+            if (port.role == "source" && port.flags.find(output_flags_string[i]) != -1) {
+                for (int k = 0; k < routes.size() && !hasFlag; k++) {
+                    if (routes[k].sources.find(port.name) != -1 &&
+                        std::find(attachedDevices.begin(), attachedDevices.end(), routes[k].sink) !=
+                                attachedDevices.end()) {
+                        hasFlag = true;
+                        std::cerr << "found port with flag " << output_flags_string[i] << "@ "
+                                  << " port :: name : " << port.name << " role : " << port.role
+                                  << " port :: flags : " << port.flags
+                                  << " connected via route name : " << routes[k].name
+                                  << " route sources : " << routes[k].sources
+                                  << " route sink : " << routes[k].sink << std::endl;
+                    }
+                }
+            }
+        }
+        if (!hasFlag) continue;
+        audio_attributes_t attributes = AUDIO_ATTRIBUTES_INITIALIZER;
+        attributes.usage = AUDIO_USAGE_MEDIA;
+        attributes.content_type = AUDIO_CONTENT_TYPE_MUSIC;
+        attributes.flags = flags[i];
+        sp<AudioPlayback> ap = sp<AudioPlayback>::make(
+                0 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+                AUDIO_OUTPUT_FLAG_NONE, AUDIO_SESSION_NONE, AudioTrack::TRANSFER_OBTAIN,
+                &attributes);
+        ASSERT_NE(nullptr, ap);
+        ASSERT_EQ(OK, ap->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
+                << "Unable to open Resource";
+        EXPECT_EQ(OK, ap->create()) << "track creation failed";
+        sp<OnAudioDeviceUpdateNotifier> cb = new OnAudioDeviceUpdateNotifier();
+        EXPECT_EQ(OK, ap->getAudioTrackHandle()->addAudioDeviceCallback(cb));
+        EXPECT_EQ(OK, ap->start()) << "audio track start failed";
+        EXPECT_EQ(OK, ap->onProcess());
+        EXPECT_EQ(OK, cb->waitForAudioDeviceCb());
+        EXPECT_TRUE(checkPatchPlayback(cb->mAudioIo, cb->mDeviceId));
+        EXPECT_NE(0, ap->getAudioTrackHandle()->getFlags() & output_flags[i]);
+        audio_patch patch;
+        EXPECT_EQ(OK, getPatchForOutputMix(cb->mAudioIo, patch));
+        for (auto j = 0; j < patch.num_sources; j++) {
+            if (patch.sources[j].type == AUDIO_PORT_TYPE_MIX &&
+                patch.sources[j].ext.mix.handle == cb->mAudioIo) {
+                if ((patch.sources[j].flags.output & output_flags[i]) == 0) {
+                    ADD_FAILURE() << "expected output flag " << output_flags[i] << " is absent";
+                    std::cerr << dumpPortConfig(patch.sources[j]);
+                }
+            }
+        }
+        ap->stop();
+        ap->getAudioTrackHandle()->removeAudioDeviceCallback(cb);
+    }
+}
+
+TEST(AudioTrackTest, TestRemoteSubmix) {
+    std::vector<std::string> attachedDevices;
+    std::vector<MixPort> mixPorts;
+    std::vector<Route> routes;
+    EXPECT_EQ(OK, parse_audio_policy_configuration_xml(attachedDevices, mixPorts, routes));
+    bool hasFlag = false;
+    for (int j = 0; j < attachedDevices.size() && !hasFlag; j++) {
+        if (attachedDevices[j].find("Remote Submix") != -1) hasFlag = true;
+    }
+    if (!hasFlag) GTEST_SKIP() << " Device does not have Remote Submix port.";
+    sp<AudioCapture> capture = new AudioCapture(AUDIO_SOURCE_REMOTE_SUBMIX, 48000,
+                                                AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO);
+    ASSERT_NE(nullptr, capture);
+    ASSERT_EQ(OK, capture->create()) << "record creation failed";
+
+    sp<AudioPlayback> playback = sp<AudioPlayback>::make(
+            48000 /* sampleRate */, 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"))
+            << "Unable to open Resource";
+    ASSERT_EQ(OK, playback->create()) << "track creation failed";
+
+    audio_port_v7 port;
+    status_t status = getPortByAttributes(AUDIO_PORT_ROLE_SOURCE, AUDIO_PORT_TYPE_DEVICE,
+                                          AUDIO_DEVICE_IN_REMOTE_SUBMIX, port);
+    EXPECT_EQ(OK, status) << "Could not find port";
+
+    EXPECT_EQ(OK, capture->start()) << "start recording failed";
+    EXPECT_EQ(port.id, capture->getAudioRecordHandle()->getRoutedDeviceId())
+            << "Capture NOT routed on expected port";
+
+    status = getPortByAttributes(AUDIO_PORT_ROLE_SINK, AUDIO_PORT_TYPE_DEVICE,
+                                 AUDIO_DEVICE_OUT_REMOTE_SUBMIX, port);
+    EXPECT_EQ(OK, status) << "Could not find port";
+
+    EXPECT_EQ(OK, playback->start()) << "audio track start failed";
+    EXPECT_EQ(OK, playback->onProcess());
+    ASSERT_EQ(port.id, playback->getAudioTrackHandle()->getRoutedDeviceId())
+            << "Playback NOT routed on expected port";
+    capture->stop();
+    playback->stop();
+}
diff --git a/media/libaudioclient/tests/audiotrack_tests.cpp b/media/libaudioclient/tests/audiotrack_tests.cpp
new file mode 100644
index 0000000..1b42a49
--- /dev/null
+++ b/media/libaudioclient/tests/audiotrack_tests.cpp
@@ -0,0 +1,211 @@
+/*
+ * Copyright (C) 2021 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
+
+#include <gtest/gtest.h>
+
+#include "audio_test_utils.h"
+
+using namespace android;
+
+TEST(AudioTrackTest, TestPlayTrack) {
+    const auto ap = sp<AudioPlayback>::make(44100 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT,
+                                            AUDIO_CHANNEL_OUT_STEREO, AUDIO_OUTPUT_FLAG_NONE,
+                                            AUDIO_SESSION_NONE, AudioTrack::TRANSFER_OBTAIN);
+    ASSERT_NE(nullptr, ap);
+    ASSERT_EQ(OK, ap->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
+            << "Unable to open Resource";
+    EXPECT_EQ(OK, ap->create()) << "track creation failed";
+    EXPECT_EQ(OK, ap->start()) << "audio track start failed";
+    EXPECT_EQ(OK, ap->onProcess());
+    ap->stop();
+}
+
+TEST(AudioTrackTest, TestSeek) {
+    const auto ap = sp<AudioPlayback>::make(
+            44100 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO);
+    ASSERT_NE(nullptr, ap);
+    ASSERT_EQ(OK, ap->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
+            << "Unable to open Resource";
+    EXPECT_EQ(OK, ap->create()) << "track creation failed";
+    EXPECT_EQ(OK, ap->start()) << "audio track start failed";
+    EXPECT_EQ(OK, ap->onProcess(true));
+    ap->stop();
+}
+
+TEST(AudioTrackTest, OffloadOrDirectPlayback) {
+    audio_offload_info_t info = AUDIO_INFO_INITIALIZER;
+    info.sample_rate = 44100;
+    info.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+    info.format = AUDIO_FORMAT_MP3;
+    info.stream_type = AUDIO_STREAM_MUSIC;
+    info.bit_rate = 192;
+    info.duration_us = 120 * 1000000;  // 120 sec
+
+    audio_config_base_t config = {/* .sample_rate = */ info.sample_rate,
+                                  /* .channel_mask = */ info.channel_mask,
+                                  /* .format = */ AUDIO_FORMAT_PCM_16_BIT};
+    audio_attributes_t attributes = AUDIO_ATTRIBUTES_INITIALIZER;
+    attributes.content_type = AUDIO_CONTENT_TYPE_MUSIC;
+    attributes.usage = AUDIO_USAGE_MEDIA;
+    attributes.flags = AUDIO_FLAG_NONE;
+
+    if (!AudioTrack::isDirectOutputSupported(config, attributes) &&
+        AUDIO_OFFLOAD_NOT_SUPPORTED == AudioSystem::getOffloadSupport(info)) {
+        GTEST_SKIP() << "offload or direct playback is not supported";
+    }
+    sp<AudioPlayback> ap = nullptr;
+    if (AUDIO_OFFLOAD_NOT_SUPPORTED != AudioSystem::getOffloadSupport(info)) {
+        ap = sp<AudioPlayback>::make(info.sample_rate, info.format, info.channel_mask,
+                                     AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, AUDIO_SESSION_NONE,
+                                     AudioTrack::TRANSFER_OBTAIN, nullptr, &info);
+    } else {
+        ap = sp<AudioPlayback>::make(config.sample_rate, config.format, config.channel_mask,
+                                     AUDIO_OUTPUT_FLAG_DIRECT, AUDIO_SESSION_NONE,
+                                     AudioTrack::TRANSFER_OBTAIN);
+    }
+    ASSERT_NE(nullptr, ap);
+    EXPECT_EQ(OK, ap->create()) << "track creation failed";
+    audio_dual_mono_mode_t mode;
+    if (OK != ap->getAudioTrackHandle()->getDualMonoMode(&mode)) {
+        std::cerr << "no dual mono presentation is available" << std::endl;
+    }
+    if (OK != ap->getAudioTrackHandle()->setDualMonoMode(AUDIO_DUAL_MONO_MODE_LR)) {
+        std::cerr << "no dual mono presentation is available" << std::endl;
+    } else {
+        EXPECT_EQ(OK, ap->getAudioTrackHandle()->getDualMonoMode(&mode));
+        EXPECT_EQ(AUDIO_DUAL_MONO_MODE_LR, mode);
+    }
+    float leveldB;
+    if (OK != ap->getAudioTrackHandle()->getAudioDescriptionMixLevel(&leveldB)) {
+        std::cerr << "Audio Description mixing is unavailable" << std::endl;
+    }
+    if (OK != ap->getAudioTrackHandle()->setAudioDescriptionMixLevel(3.14f)) {
+        std::cerr << "Audio Description mixing is unavailable" << std::endl;
+    } else {
+        EXPECT_EQ(OK, ap->getAudioTrackHandle()->getAudioDescriptionMixLevel(&leveldB));
+        EXPECT_EQ(3.14f, leveldB);
+    }
+    AudioPlaybackRate audioRate;
+    audioRate = ap->getAudioTrackHandle()->getPlaybackRate();
+    std::cerr << "playback speed :: " << audioRate.mSpeed << std::endl
+              << "playback pitch :: " << audioRate.mPitch << std::endl;
+    audioRate.mSpeed = 2.0f;
+    audioRate.mPitch = 2.0f;
+    audioRate.mStretchMode = AUDIO_TIMESTRETCH_STRETCH_VOICE;
+    audioRate.mFallbackMode = AUDIO_TIMESTRETCH_FALLBACK_MUTE;
+    EXPECT_TRUE(isAudioPlaybackRateValid(audioRate));
+    if (OK != ap->getAudioTrackHandle()->setPlaybackRate(audioRate)) {
+        std::cerr << "unable to set playback rate parameters" << std::endl;
+    } else {
+        AudioPlaybackRate audioRateLocal;
+        audioRateLocal = ap->getAudioTrackHandle()->getPlaybackRate();
+        EXPECT_TRUE(isAudioPlaybackRateEqual(audioRate, audioRateLocal));
+    }
+    ap->stop();
+}
+
+TEST(AudioTrackTest, TestAudioCbNotifier) {
+    const auto ap = sp<AudioPlayback>::make(0 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT,
+                                            AUDIO_CHANNEL_OUT_STEREO, AUDIO_OUTPUT_FLAG_FAST,
+                                            AUDIO_SESSION_NONE, AudioTrack::TRANSFER_SHARED);
+    ASSERT_NE(nullptr, ap);
+    ASSERT_EQ(OK, ap->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
+            << "Unable to open Resource";
+    EXPECT_EQ(OK, ap->create()) << "track creation failed";
+    EXPECT_EQ(BAD_VALUE, ap->getAudioTrackHandle()->addAudioDeviceCallback(nullptr));
+    sp<OnAudioDeviceUpdateNotifier> cb = new OnAudioDeviceUpdateNotifier();
+    sp<OnAudioDeviceUpdateNotifier> cbOld = new OnAudioDeviceUpdateNotifier();
+    EXPECT_EQ(OK, ap->getAudioTrackHandle()->addAudioDeviceCallback(cbOld));
+    EXPECT_EQ(INVALID_OPERATION, ap->getAudioTrackHandle()->addAudioDeviceCallback(cbOld));
+    EXPECT_EQ(OK, ap->getAudioTrackHandle()->addAudioDeviceCallback(cb));
+    EXPECT_EQ(OK, ap->start()) << "audio track start failed";
+    EXPECT_EQ(OK, ap->onProcess());
+    EXPECT_EQ(OK, cb->waitForAudioDeviceCb());
+    EXPECT_EQ(AUDIO_IO_HANDLE_NONE, cbOld->mAudioIo);
+    EXPECT_EQ(AUDIO_PORT_HANDLE_NONE, cbOld->mDeviceId);
+    EXPECT_NE(AUDIO_IO_HANDLE_NONE, cb->mAudioIo);
+    EXPECT_NE(AUDIO_PORT_HANDLE_NONE, cb->mDeviceId);
+    EXPECT_EQ(cb->mAudioIo, ap->getAudioTrackHandle()->getOutput());
+    EXPECT_EQ(cb->mDeviceId, ap->getAudioTrackHandle()->getRoutedDeviceId());
+    String8 keys;
+    keys = ap->getAudioTrackHandle()->getParameters(keys);
+    if (!keys.isEmpty()) {
+        std::cerr << "track parameters :: " << keys << std::endl;
+    }
+    EXPECT_TRUE(checkPatchPlayback(cb->mAudioIo, cb->mDeviceId));
+    EXPECT_EQ(BAD_VALUE, ap->getAudioTrackHandle()->removeAudioDeviceCallback(nullptr));
+    EXPECT_EQ(INVALID_OPERATION, ap->getAudioTrackHandle()->removeAudioDeviceCallback(cbOld));
+    EXPECT_EQ(OK, ap->getAudioTrackHandle()->removeAudioDeviceCallback(cb));
+    ap->stop();
+}
+
+class AudioTrackCreateTest
+    : public ::testing::TestWithParam<std::tuple<uint32_t, audio_format_t, audio_channel_mask_t,
+                                                 audio_output_flags_t, audio_session_t>> {
+  public:
+    AudioTrackCreateTest()
+        : mSampleRate(std::get<0>(GetParam())),
+          mFormat(std::get<1>(GetParam())),
+          mChannelMask(std::get<2>(GetParam())),
+          mFlags(std::get<3>(GetParam())),
+          mSessionId(std::get<4>(GetParam())){};
+
+    const uint32_t mSampleRate;
+    const audio_format_t mFormat;
+    const audio_channel_mask_t mChannelMask;
+    const audio_output_flags_t mFlags;
+    const audio_session_t mSessionId;
+
+    sp<AudioPlayback> mAP;
+
+    virtual void SetUp() override {
+        mAP = sp<AudioPlayback>::make(mSampleRate, mFormat, mChannelMask, mFlags,
+                                              mSessionId);
+        ASSERT_NE(nullptr, mAP);
+        ASSERT_EQ(OK, mAP->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
+                << "Unable to open Resource";
+        ASSERT_EQ(OK, mAP->create()) << "track creation failed";
+    }
+
+    virtual void TearDown() override {
+        if (mAP) mAP->stop();
+    }
+};
+
+TEST_P(AudioTrackCreateTest, TestCreateTrack) {
+    EXPECT_EQ(mFormat, mAP->getAudioTrackHandle()->format());
+    EXPECT_EQ(audio_channel_count_from_out_mask(mChannelMask),
+              mAP->getAudioTrackHandle()->channelCount());
+    if (mSampleRate != 0) EXPECT_EQ(mSampleRate, mAP->getAudioTrackHandle()->getSampleRate());
+    if (mSessionId != AUDIO_SESSION_NONE)
+        EXPECT_EQ(mSessionId, mAP->getAudioTrackHandle()->getSessionId());
+    EXPECT_EQ(mSampleRate, mAP->getAudioTrackHandle()->getOriginalSampleRate());
+    EXPECT_EQ(OK, mAP->start()) << "audio track start failed";
+    EXPECT_EQ(OK, mAP->onProcess());
+}
+
+// sampleRate, format, channelMask, flags, sessionId
+INSTANTIATE_TEST_SUITE_P(
+        AudioTrackParameterizedTest, AudioTrackCreateTest,
+        ::testing::Combine(::testing::Values(48000), ::testing::Values(AUDIO_FORMAT_PCM_16_BIT),
+                           ::testing::Values(AUDIO_CHANNEL_OUT_STEREO),
+                           ::testing::Values(AUDIO_OUTPUT_FLAG_NONE,
+                                             AUDIO_OUTPUT_FLAG_PRIMARY | AUDIO_OUTPUT_FLAG_FAST,
+                                             AUDIO_OUTPUT_FLAG_RAW | AUDIO_OUTPUT_FLAG_FAST,
+                                             AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
+                           ::testing::Values(AUDIO_SESSION_NONE)));
diff --git a/media/libaudioclient/tests/bbb_1ch_8kHz_s16le.raw b/media/libaudioclient/tests/bbb_1ch_8kHz_s16le.raw
new file mode 100644
index 0000000..2d1e4bf
--- /dev/null
+++ b/media/libaudioclient/tests/bbb_1ch_8kHz_s16le.raw
Binary files differ
diff --git a/media/libaudioclient/tests/bbb_2ch_24kHz_s16le.raw b/media/libaudioclient/tests/bbb_2ch_24kHz_s16le.raw
new file mode 100644
index 0000000..c8ac5f7
--- /dev/null
+++ b/media/libaudioclient/tests/bbb_2ch_24kHz_s16le.raw
Binary files differ
diff --git a/media/libaudioclient/tests/test_create_audiorecord.cpp b/media/libaudioclient/tests/test_create_audiorecord.cpp
index 2e0883b..277110b 100644
--- a/media/libaudioclient/tests/test_create_audiorecord.cpp
+++ b/media/libaudioclient/tests/test_create_audiorecord.cpp
@@ -29,14 +29,13 @@
 
 #define NUM_ARGUMENTS 8
 #define VERSION_VALUE "1.0"
-#define PACKAGE_NAME  "AudioRecord test"
+#define PACKAGE_NAME "AudioRecord test"
 
 namespace android {
 
 using android::content::AttributionSourceState;
 
-int testRecord(FILE *inputFile, int outputFileFd)
-{
+int testRecord(FILE* inputFile, int outputFileFd) {
     char line[MAX_INPUT_FILE_LINE_LENGTH];
     uint32_t testCount = 0;
     Vector<String16> args;
@@ -47,11 +46,9 @@
     attributionSource.token = sp<BBinder>::make();
 
     if (inputFile == nullptr) {
-        sp<AudioRecord> record = new AudioRecord(AUDIO_SOURCE_DEFAULT,
-                                              0 /* sampleRate */,
-                                              AUDIO_FORMAT_DEFAULT,
-                                              AUDIO_CHANNEL_IN_MONO,
-                                              attributionSource);
+        sp<AudioRecord> record =
+                new AudioRecord(AUDIO_SOURCE_DEFAULT, 0 /* sampleRate */, AUDIO_FORMAT_DEFAULT,
+                                AUDIO_CHANNEL_IN_MONO, attributionSource);
         if (record == 0 || record->initCheck() != NO_ERROR) {
             write(outputFileFd, "Error creating AudioRecord\n",
                   sizeof("Error creating AudioRecord\n"));
@@ -80,11 +77,10 @@
         char statusStr[MAX_OUTPUT_FILE_LINE_LENGTH];
         bool fast = false;
 
-        if (sscanf(line, " %u %x %x %zu %d %x %u %u",
-                   &sampleRate, &format, &channelMask,
-                   &frameCount, &notificationFrames,
-                   &flags, &sessionId, &inputSource) != NUM_ARGUMENTS) {
-            fprintf(stderr, "Malformed line for test #%u in input file\n", testCount+1);
+        if (sscanf(line, " %u %x %x %zu %d %x %u %u", &sampleRate, &format, &channelMask,
+                   &frameCount, &notificationFrames, &flags, &sessionId,
+                   &inputSource) != NUM_ARGUMENTS) {
+            fprintf(stderr, "Malformed line for test #%u in input file\n", testCount + 1);
             ret = 1;
             continue;
         }
@@ -100,21 +96,10 @@
         sp<AudioRecord> record = new AudioRecord(attributionSource);
         const auto emptyCallback = sp<AudioRecord::IAudioRecordCallback>::make();
 
-        record->set(AUDIO_SOURCE_DEFAULT,
-                   sampleRate,
-                   format,
-                   channelMask,
-                   frameCount,
-                   fast ? emptyCallback : nullptr,
-                   notificationFrames,
-                   false,
-                   sessionId,
-                   fast ? AudioRecord::TRANSFER_CALLBACK : AudioRecord::TRANSFER_DEFAULT,
-                   flags,
-                   getuid(),
-                   getpid(),
-                   &attributes,
-                   AUDIO_PORT_HANDLE_NONE);
+        record->set(AUDIO_SOURCE_DEFAULT, sampleRate, format, channelMask, frameCount,
+                    fast ? emptyCallback : nullptr, notificationFrames, false, sessionId,
+                    fast ? AudioRecord::TRANSFER_CALLBACK : AudioRecord::TRANSFER_DEFAULT, flags,
+                    getuid(), getpid(), &attributes, AUDIO_PORT_HANDLE_NONE);
         status = record->initCheck();
         sprintf(statusStr, "\n#### Test %u status %d\n", testCount, status);
         write(outputFileFd, statusStr, strlen(statusStr));
@@ -126,11 +111,8 @@
     return ret;
 }
 
-}; // namespace android
+};  // namespace android
 
-
-int main(int argc, char **argv)
-{
+int main(int argc, char** argv) {
     return android::main(argc, argv, android::testRecord);
 }
-
diff --git a/media/libaudioclient/tests/test_create_audiotrack.cpp b/media/libaudioclient/tests/test_create_audiotrack.cpp
index e7231d3..4e09e21 100644
--- a/media/libaudioclient/tests/test_create_audiotrack.cpp
+++ b/media/libaudioclient/tests/test_create_audiotrack.cpp
@@ -32,18 +32,15 @@
 
 namespace android {
 
-int testTrack(FILE *inputFile, int outputFileFd)
-{
+int testTrack(FILE* inputFile, int outputFileFd) {
     char line[MAX_INPUT_FILE_LINE_LENGTH];
     uint32_t testCount = 0;
     Vector<String16> args;
     int ret = 0;
 
     if (inputFile == nullptr) {
-        sp<AudioTrack> track = new AudioTrack(AUDIO_STREAM_DEFAULT,
-                                              0 /* sampleRate */,
-                                              AUDIO_FORMAT_DEFAULT,
-                                              AUDIO_CHANNEL_OUT_STEREO);
+        sp<AudioTrack> track = new AudioTrack(AUDIO_STREAM_DEFAULT, 0 /* sampleRate */,
+                                              AUDIO_FORMAT_DEFAULT, AUDIO_CHANNEL_OUT_STEREO);
         if (track == 0 || track->initCheck() != NO_ERROR) {
             write(outputFileFd, "Error creating AudioTrack\n",
                   sizeof("Error creating AudioTrack\n"));
@@ -78,11 +75,10 @@
         bool offload = false;
         bool fast = false;
 
-        if (sscanf(line, " %u %x %x %zu %d %u %x %u %u %u",
-                   &sampleRate, &format, &channelMask,
-                   &frameCount, &notificationFrames, &useSharedBuffer,
-                   &flags, &sessionId, &usage, &contentType) != NUM_ARGUMENTS) {
-            fprintf(stderr, "Malformed line for test #%u in input file\n", testCount+1);
+        if (sscanf(line, " %u %x %x %zu %d %u %x %u %u %u", &sampleRate, &format, &channelMask,
+                   &frameCount, &notificationFrames, &useSharedBuffer, &flags, &sessionId, &usage,
+                   &contentType) != NUM_ARGUMENTS) {
+            fprintf(stderr, "Malformed line for test #%u in input file\n", testCount + 1);
             ret = 1;
             continue;
         }
@@ -90,7 +86,7 @@
 
         if (useSharedBuffer != 0) {
             size_t heapSize = audio_channel_count_from_out_mask(channelMask) *
-                    audio_bytes_per_sample(format) * frameCount;
+                              audio_bytes_per_sample(format) * frameCount;
             heap = new MemoryDealer(heapSize, "AudioTrack Heap Base");
             sharedBuffer = heap->allocate(heapSize);
             frameCount = 0;
@@ -111,25 +107,13 @@
         attributes.usage = usage;
         sp<AudioTrack> track = new AudioTrack();
         const auto emptyCallback = sp<AudioTrack::IAudioTrackCallback>::make();
-        track->set(AUDIO_STREAM_DEFAULT,
-                   sampleRate,
-                   format,
-                   channelMask,
-                   frameCount,
-                   flags,
-                   (fast || offload) ? emptyCallback : nullptr,
-                   notificationFrames,
-                   sharedBuffer,
-                   false,
-                   sessionId,
-                   ((fast && sharedBuffer == 0) || offload) ?
-                           AudioTrack::TRANSFER_CALLBACK : AudioTrack::TRANSFER_DEFAULT,
-                   offload ? &offloadInfo : nullptr,
-                   AttributionSourceState(),
-                   &attributes,
-                   false,
-                   1.0f,
-                   AUDIO_PORT_HANDLE_NONE);
+        track->set(AUDIO_STREAM_DEFAULT, sampleRate, format, channelMask, frameCount, flags,
+                   (fast || offload) ? emptyCallback : nullptr, notificationFrames, sharedBuffer,
+                   false, sessionId,
+                   ((fast && sharedBuffer == 0) || offload) ? AudioTrack::TRANSFER_CALLBACK
+                                                            : AudioTrack::TRANSFER_DEFAULT,
+                   offload ? &offloadInfo : nullptr, AttributionSourceState(), &attributes, false,
+                   1.0f, AUDIO_PORT_HANDLE_NONE);
         status = track->initCheck();
         sprintf(statusStr, "\n#### Test %u status %d\n", testCount, status);
         write(outputFileFd, statusStr, strlen(statusStr));
@@ -141,11 +125,8 @@
     return ret;
 }
 
-}; // namespace android
+};  // namespace android
 
-
-int main(int argc, char **argv)
-{
+int main(int argc, char** argv) {
     return android::main(argc, argv, android::testTrack);
 }
-
diff --git a/media/libaudioclient/tests/test_create_utils.cpp b/media/libaudioclient/tests/test_create_utils.cpp
index caf5227..c2c2e8b 100644
--- a/media/libaudioclient/tests/test_create_utils.cpp
+++ b/media/libaudioclient/tests/test_create_utils.cpp
@@ -23,10 +23,10 @@
 
 namespace android {
 
-int readLine(FILE *inputFile, char *line, int size) {
+int readLine(FILE* inputFile, char* line, int size) {
     int ret = 0;
     while (true) {
-        char *str = fgets(line, size, inputFile);
+        char* str = fgets(line, size, inputFile);
         if (str == nullptr) {
             ret = -1;
             break;
@@ -42,8 +42,7 @@
     return ret;
 }
 
-bool checkVersion(FILE *inputFile, const char *version)
-{
+bool checkVersion(FILE* inputFile, const char* version) {
     char line[MAX_INPUT_FILE_LINE_LENGTH];
     char versionKey[MAX_INPUT_FILE_LINE_LENGTH];
     char versionValue[MAX_INPUT_FILE_LINE_LENGTH];
@@ -68,9 +67,8 @@
     return true;
 }
 
-int main(int argc, char **argv, test_func_t testFunc)
-{
-    FILE *inputFile = nullptr;
+int main(int argc, char** argv, test_func_t testFunc) {
+    FILE* inputFile = nullptr;
     int outputFileFd = STDOUT_FILENO;
     mode_t mode = S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH;
     int ret = 0;
@@ -96,7 +94,7 @@
         if (strcmp(*argv, "-o") == 0) {
             argv++;
             if (*argv) {
-                outputFileFd = open(*argv, O_WRONLY|O_CREAT, mode);
+                outputFileFd = open(*argv, O_WRONLY | O_CREAT, mode);
                 if (outputFileFd < 0) {
                     ret = 1;
                 }
@@ -126,5 +124,4 @@
     return ret;
 }
 
-}; // namespace android
-
+};  // namespace android
diff --git a/media/libaudioclient/tests/test_create_utils.h b/media/libaudioclient/tests/test_create_utils.h
index 9a6f9fa..110baf7 100644
--- a/media/libaudioclient/tests/test_create_utils.h
+++ b/media/libaudioclient/tests/test_create_utils.h
@@ -27,13 +27,12 @@
 
 namespace android {
 
-int readLine(FILE *inputFile, char *line, int size);
+int readLine(FILE* inputFile, char* line, int size);
 
-bool checkVersion(FILE *inputFile, const char *version);
+bool checkVersion(FILE* inputFile, const char* version);
 
+typedef int (*test_func_t)(FILE* inputFile, int outputFileFd);
 
-typedef int (*test_func_t)(FILE *inputFile, int outputFileFd);
+int main(int argc, char** argv, test_func_t testFunc);
 
-int main(int argc, char **argv, test_func_t testFunc);
-
-}; // namespace android
+};  // namespace android
diff --git a/media/libaudioclient/tests/trackplayerbase_tests.cpp b/media/libaudioclient/tests/trackplayerbase_tests.cpp
new file mode 100644
index 0000000..c9b704d
--- /dev/null
+++ b/media/libaudioclient/tests/trackplayerbase_tests.cpp
@@ -0,0 +1,161 @@
+/*
+ * Copyright (C) 2021 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 "TrackPlayerBaseTest"
+
+#include <gtest/gtest.h>
+
+#include <media/TrackPlayerBase.h>
+
+using namespace android;
+using namespace android::media;
+
+class TrackPlayer : public TrackPlayerBase, public AudioTrack::IAudioTrackCallback {
+  public:
+    // methods protected in base class
+    using TrackPlayerBase::playerPause;
+    using TrackPlayerBase::playerSetVolume;
+    using TrackPlayerBase::playerStart;
+    using TrackPlayerBase::playerStop;
+};
+
+class TrackPlayerBaseTest
+    : public ::testing::TestWithParam<std::tuple<double, double, uint32_t, uint32_t>> {
+  public:
+    TrackPlayerBaseTest()
+        : mDuration(std::get<0>(GetParam())),
+          mPulseFreq(std::get<1>(GetParam())),
+          mChannelCount(std::get<2>(GetParam())),
+          mSampleRate(std::get<3>(GetParam())){};
+
+    virtual void SetUp() override {
+        mFrameCount = mDuration * mSampleRate;
+        audio_channel_mask_t channelMask = audio_channel_out_mask_from_count(mChannelCount);
+        sp<AudioTrack> track = new AudioTrack(mStreamType, mSampleRate, mFormat, channelMask,
+                                              mFrameCount, mFlags, nullptr /* callback */,
+                                              0 /* notificationFrames */, AUDIO_SESSION_NONE);
+        ASSERT_EQ(track->initCheck(), NO_ERROR);
+
+        mPlayer = new TrackPlayer();
+        mPlayer->init(track.get(), mPlayer, PLAYER_TYPE_AAUDIO, AUDIO_USAGE_MEDIA,
+                      AUDIO_SESSION_NONE);
+        sp<AudioTrack> playerTrack = mPlayer->mAudioTrack;
+        ASSERT_EQ(playerTrack->initCheck(), NO_ERROR);
+
+        mBufferSize = mFrameCount * playerTrack->frameSize();
+        mBuffer.resize(mBufferSize, 0);
+
+        // populate buffer
+        ASSERT_NE(mPulseFreq, 0);
+        int32_t nPulseSamples = mSampleRate / mPulseFreq;
+        int32_t pulseSize = nPulseSamples * playerTrack->frameSize();
+
+        int32_t marker = 0;
+        while (marker + pulseSize <= mBufferSize) {
+            memset(mBuffer.data() + marker, 127, pulseSize / 2);
+            marker += pulseSize;
+        }
+    }
+
+    void playBuffer() {
+        bool blocking = true;
+        ssize_t nbytes = mPlayer->mAudioTrack->write(mBuffer.data(), mBufferSize, blocking);
+        EXPECT_EQ(nbytes, mBufferSize) << "Did not write all data in blocking mode";
+    }
+
+    const double mDuration;  // seconds
+    sp<TrackPlayer> mPlayer;
+
+  private:
+    const double mPulseFreq;
+    const uint32_t mChannelCount;
+    const uint32_t mSampleRate;
+
+    const audio_format_t mFormat = AUDIO_FORMAT_PCM_16_BIT;
+    const audio_output_flags_t mFlags = AUDIO_OUTPUT_FLAG_NONE;
+    const audio_stream_type_t mStreamType = AUDIO_STREAM_MUSIC;
+
+    int32_t mBufferSize;
+    int32_t mFrameCount;
+    std::vector<uint8_t> mBuffer;
+};
+
+class PlaybackTestParam : public TrackPlayerBaseTest {};
+
+TEST_P(PlaybackTestParam, PlaybackTest) {
+    // no-op implementation
+    EXPECT_TRUE(mPlayer->setStartDelayMs(0).isOk());
+
+    ASSERT_EQ(mPlayer->playerStart(), NO_ERROR);
+    ASSERT_NO_FATAL_FAILURE(playBuffer());
+    EXPECT_EQ(mPlayer->playerStop(), NO_ERROR);
+}
+
+INSTANTIATE_TEST_SUITE_P(TrackPlayerTest, PlaybackTestParam,
+                         ::testing::Values(std::make_tuple(2.5, 25.0, 2, 48000)));
+
+class ChangeVolumeTestParam : public TrackPlayerBaseTest {};
+
+TEST_P(ChangeVolumeTestParam, ChangeVolumeTest) {
+    float volume = 1.0f;
+    (void)mPlayer->setPlayerVolume(volume / 2, volume);
+
+    ASSERT_TRUE(mPlayer->start().isOk());
+    ASSERT_EQ(mPlayer->playerSetVolume(), NO_ERROR);
+
+    ASSERT_NO_FATAL_FAILURE(playBuffer());
+
+    EXPECT_TRUE(mPlayer->stop().isOk());
+
+    std::vector<float> setVol = {0.95f, 0.05f, 0.5f, 0.25f, -1.0f, 1.0f, 1.0f};
+    std::vector<float> setPan = {0.0f, 0.0f, 1.0f, -1.0f, -1.0f, 0.5f, -0.5f};
+
+    ASSERT_TRUE(mPlayer->start().isOk());
+
+    for (int32_t i = 0; i < setVol.size(); i++) {
+        EXPECT_TRUE(mPlayer->setVolume(setVol[i]).isOk());
+        EXPECT_TRUE(mPlayer->setPan(setPan[i]).isOk());
+        ASSERT_NO_FATAL_FAILURE(playBuffer());
+    }
+    EXPECT_TRUE(mPlayer->stop().isOk());
+}
+
+INSTANTIATE_TEST_SUITE_P(TrackPlayerTest, ChangeVolumeTestParam,
+                         ::testing::Values(std::make_tuple(1.0, 100.0, 1, 24000)));
+
+class PauseTestParam : public TrackPlayerBaseTest {};
+
+TEST_P(PauseTestParam, PauseTest) {
+    ASSERT_EQ(mPlayer->playerStart(), NO_ERROR);
+    ASSERT_NO_FATAL_FAILURE(playBuffer());
+
+    ASSERT_EQ(mPlayer->playerPause(), NO_ERROR);
+    ASSERT_EQ(mPlayer->playerStart(), NO_ERROR);
+
+    ASSERT_NO_FATAL_FAILURE(playBuffer());
+
+    EXPECT_EQ(mPlayer->playerStop(), NO_ERROR);
+
+    for (int32_t i = 0; i < 5; i++) {
+        ASSERT_TRUE(mPlayer->start().isOk());
+        ASSERT_NO_FATAL_FAILURE(playBuffer());
+        ASSERT_TRUE(mPlayer->pause().isOk());
+    }
+    EXPECT_TRUE(mPlayer->stop().isOk());
+}
+
+INSTANTIATE_TEST_SUITE_P(TrackPlayerTest, PauseTestParam,
+                         ::testing::Values(std::make_tuple(1.0, 75.0, 2, 24000)));
diff --git a/media/libaudiofoundation/include/media/AudioPort.h b/media/libaudiofoundation/include/media/AudioPort.h
index d6a098f..b1235f5 100644
--- a/media/libaudiofoundation/include/media/AudioPort.h
+++ b/media/libaudiofoundation/include/media/AudioPort.h
@@ -72,7 +72,7 @@
     AudioProfileVector &getAudioProfiles() { return mProfiles; }
 
     void setExtraAudioDescriptors(
-            const std::vector<media::audio::common::ExtraAudioDescriptor> extraAudioDescriptors) {
+            const std::vector<media::audio::common::ExtraAudioDescriptor>& extraAudioDescriptors) {
         mExtraAudioDescriptors = extraAudioDescriptors;
     }
     std::vector<media::audio::common::ExtraAudioDescriptor> &getExtraAudioDescriptors() {
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 8546a7a..9351499 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -140,8 +140,6 @@
 status_t AudioPolicyManager::setDeviceConnectionStateInt(
         audio_policy_dev_state_t state, const android::media::audio::common::AudioPort& port,
         audio_format_t encodedFormat) {
-    // TODO: b/211601178 Forward 'port' to Audio HAL via mHwModules. For now, only device_type,
-    // device_address and device_name are forwarded.
     if (port.ext.getTag() != AudioPortExt::device) {
         return BAD_VALUE;
     }
@@ -160,7 +158,13 @@
     sp<DeviceDescriptor> device = mHwModules.getDeviceDescriptor(
             device_type, device_address.c_str(), device_name, encodedFormat,
             state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
-    return device ? setDeviceConnectionStateInt(device, state) : INVALID_OPERATION;
+    if (device == nullptr) {
+        return INVALID_OPERATION;
+    }
+    if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
+        device->setExtraAudioDescriptors(port.extraAudioDescriptors);
+    }
+    return setDeviceConnectionStateInt(device, state);
 }
 
 status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t deviceType,
diff --git a/services/audiopolicy/service/AudioPolicyEffects.cpp b/services/audiopolicy/service/AudioPolicyEffects.cpp
index 70fdfcb..c7a60c2 100644
--- a/services/audiopolicy/service/AudioPolicyEffects.cpp
+++ b/services/audiopolicy/service/AudioPolicyEffects.cpp
@@ -127,7 +127,8 @@
             attributionSource.packageName = "android";
             attributionSource.token = sp<BBinder>::make();
             sp<AudioEffect> fx = new AudioEffect(attributionSource);
-            fx->set(NULL, &effect->mUuid, -1, 0, 0, audioSession, input);
+            fx->set(nullptr /*type */, &effect->mUuid, -1 /* priority */, nullptr /* callback */,
+                    audioSession, input);
             status_t status = fx->initCheck();
             if (status != NO_ERROR && status != ALREADY_EXISTS) {
                 ALOGW("addInputEffects(): failed to create Fx %s on source %d",
@@ -279,7 +280,8 @@
             attributionSource.packageName = "android";
             attributionSource.token = sp<BBinder>::make();
             sp<AudioEffect> fx = new AudioEffect(attributionSource);
-            fx->set(NULL, &effect->mUuid, 0, 0, 0, audioSession, output);
+            fx->set(nullptr /* type */, &effect->mUuid, 0 /* priority */, nullptr /* callback */,
+                    audioSession, output);
             status_t status = fx->initCheck();
             if (status != NO_ERROR && status != ALREADY_EXISTS) {
                 ALOGE("addOutputSessionEffects(): failed to create Fx  %s on session %d",
@@ -984,8 +986,8 @@
             attributionSource.packageName = "android";
             attributionSource.token = sp<BBinder>::make();
             sp<AudioEffect> fx = new AudioEffect(attributionSource);
-            fx->set(EFFECT_UUID_NULL, &effectDesc->mUuid, 0, nullptr,
-                    nullptr, AUDIO_SESSION_DEVICE, AUDIO_IO_HANDLE_NONE,
+            fx->set(EFFECT_UUID_NULL, &effectDesc->mUuid, 0 /* priority */, nullptr /* callback */,
+                    AUDIO_SESSION_DEVICE, AUDIO_IO_HANDLE_NONE,
                     AudioDeviceTypeAddr{deviceEffects->getDeviceType(),
                                         deviceEffects->getDeviceAddress()});
             status_t status = fx->initCheck();
diff --git a/services/audiopolicy/tests/Android.bp b/services/audiopolicy/tests/Android.bp
index 2e220bc..e887798 100644
--- a/services/audiopolicy/tests/Android.bp
+++ b/services/audiopolicy/tests/Android.bp
@@ -30,6 +30,8 @@
     ],
 
     static_libs: [
+        "android.media.audio.common.types-V1-cpp",
+        "audioclient-types-aidl-cpp",
         "libaudiopolicycomponents",
         "libgmock",
     ],
diff --git a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
index 057fa58..96f58d2 100644
--- a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
@@ -103,8 +103,12 @@
         ++mAudioPortListUpdateCount;
     }
 
-    status_t setDeviceConnectedState(
-            const struct audio_port_v7 *port __unused, bool connected __unused) override {
+    status_t setDeviceConnectedState(const struct audio_port_v7 *port, bool connected) override {
+        if (connected) {
+            mConnectedDevicePorts.push_back(*port);
+        } else {
+            mDisconnectedDevicePorts.push_back(*port);
+        }
         return NO_ERROR;
     }
 
@@ -150,6 +154,30 @@
         return NO_ERROR;
     }
 
+    size_t getConnectedDevicePortCount() const {
+        return mConnectedDevicePorts.size();
+    }
+
+    const struct audio_port_v7 *getLastConnectedDevicePort() const {
+        if (mConnectedDevicePorts.empty()) {
+            return nullptr;
+        }
+        auto it = --mConnectedDevicePorts.end();
+        return &(*it);
+    }
+
+    size_t getDisconnectedDevicePortCount() const {
+        return mDisconnectedDevicePorts.size();
+    }
+
+    const struct audio_port_v7 *getLastDisconnectedDevicePort() const {
+        if (mDisconnectedDevicePorts.empty()) {
+            return nullptr;
+        }
+        auto it = --mDisconnectedDevicePorts.end();
+        return &(*it);
+    }
+
 private:
     audio_module_handle_t mNextModuleHandle = AUDIO_MODULE_HANDLE_NONE + 1;
     audio_io_handle_t mNextIoHandle = AUDIO_IO_HANDLE_NONE + 1;
@@ -158,6 +186,8 @@
     std::set<std::string> mAllowedModuleNames;
     size_t mAudioPortListUpdateCount = 0;
     size_t mRoutingUpdatedUpdateCount = 0;
+    std::vector<struct audio_port_v7> mConnectedDevicePorts;
+    std::vector<struct audio_port_v7> mDisconnectedDevicePorts;
 };
 
 } // namespace android
diff --git a/services/audiopolicy/tests/AudioPolicyTestManager.h b/services/audiopolicy/tests/AudioPolicyTestManager.h
index 7441f20..2a7a060 100644
--- a/services/audiopolicy/tests/AudioPolicyTestManager.h
+++ b/services/audiopolicy/tests/AudioPolicyTestManager.h
@@ -37,6 +37,7 @@
     using AudioPolicyManager::getDirectProfilesForAttributes;
     using AudioPolicyManager::setDeviceConnectionState;
     using AudioPolicyManager::deviceToAudioPort;
+    using AudioPolicyManager::handleDeviceConfigChange;
     uint32_t getAudioPortGeneration() const { return mAudioPortGeneration; }
 };
 
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index 43b1a2a..bb00c48 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -919,6 +919,30 @@
     EXPECT_TRUE(foundVoipTx);
 }
 
+TEST_F(AudioPolicyManagerTestWithConfigurationFile, HandleDeviceConfigChange) {
+    {
+        const auto prevCounter = mClient->getRoutingUpdatedCounter();
+
+        EXPECT_EQ(NO_ERROR, mManager->setDeviceConnectionState(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP,
+                                                               AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+                                                               "", "", AUDIO_FORMAT_LDAC));
+        const auto currCounter = mClient->getRoutingUpdatedCounter();
+        EXPECT_GT(currCounter, prevCounter);
+    }
+    {
+        const auto prevCounter = mClient->getRoutingUpdatedCounter();
+        // Update device configuration
+        EXPECT_EQ(NO_ERROR, mManager->handleDeviceConfigChange(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP,
+                                                               "" /*address*/, "" /*name*/,
+                                                               AUDIO_FORMAT_AAC));
+
+        // As mClient marks isReconfigA2dpSupported to false, device state needs to be toggled for
+        // config changes to take effect
+        const auto currCounter = mClient->getRoutingUpdatedCounter();
+        EXPECT_GT(currCounter, prevCounter);
+    }
+}
+
 using PolicyMixTuple = std::tuple<audio_usage_t, audio_source_t, uint32_t>;
 
 class AudioPolicyManagerTestDynamicPolicy : public AudioPolicyManagerTestWithConfigurationFile {
@@ -1700,6 +1724,45 @@
             address.c_str(), name.c_str(), AUDIO_FORMAT_DEFAULT));
 }
 
+android::media::audio::common::ExtraAudioDescriptor make_ExtraAudioDescriptor(
+        android::media::audio::common::AudioStandard audioStandard,
+        android::media::audio::common::AudioEncapsulationType audioEncapsulationType) {
+    android::media::audio::common::ExtraAudioDescriptor result;
+    result.standard = audioStandard;
+    result.audioDescriptor = {0xb4, 0xaf, 0x98, 0x1a};
+    result.encapsulationType = audioEncapsulationType;
+    return result;
+}
+
+TEST_P(AudioPolicyManagerTestDeviceConnection, PassingExtraAudioDescriptors) {
+    const audio_devices_t type = std::get<0>(GetParam());
+    if (!audio_device_is_digital(type)) {
+        // EADs are used only for HDMI devices.
+        GTEST_SKIP() << "Not a digital device type: " << audio_device_to_string(type);
+    }
+    const std::string name = std::get<1>(GetParam());
+    const std::string address = std::get<2>(GetParam());
+    android::media::AudioPort audioPort;
+    ASSERT_EQ(NO_ERROR,
+            mManager->deviceToAudioPort(type, address.c_str(), name.c_str(), &audioPort));
+    android::media::audio::common::AudioPort& port = audioPort.hal;
+    port.extraAudioDescriptors.push_back(make_ExtraAudioDescriptor(
+                    android::media::audio::common::AudioStandard::EDID,
+                    android::media::audio::common::AudioEncapsulationType::IEC61937));
+    const size_t lastConnectedDevicePortCount = mClient->getConnectedDevicePortCount();
+    const size_t lastDisconnectedDevicePortCount = mClient->getDisconnectedDevicePortCount();
+    EXPECT_EQ(NO_ERROR, mManager->setDeviceConnectionState(
+                    AUDIO_POLICY_DEVICE_STATE_AVAILABLE, port, AUDIO_FORMAT_DEFAULT));
+    EXPECT_EQ(lastConnectedDevicePortCount + 1, mClient->getConnectedDevicePortCount());
+    EXPECT_EQ(lastDisconnectedDevicePortCount, mClient->getDisconnectedDevicePortCount());
+    const audio_port_v7* devicePort = mClient->getLastConnectedDevicePort();
+    EXPECT_EQ(port.extraAudioDescriptors.size(), devicePort->num_extra_audio_descriptors);
+    EXPECT_EQ(AUDIO_STANDARD_EDID, devicePort->extra_audio_descriptors[0].standard);
+    EXPECT_EQ(AUDIO_ENCAPSULATION_TYPE_IEC61937,
+            devicePort->extra_audio_descriptors[0].encapsulation_type);
+    EXPECT_NE(0, devicePort->extra_audio_descriptors[0].descriptor[0]);
+}
+
 INSTANTIATE_TEST_CASE_P(
         DeviceConnectionState,
         AudioPolicyManagerTestDeviceConnection,
diff --git a/services/audiopolicy/tests/resources/test_audio_policy_configuration.xml b/services/audiopolicy/tests/resources/test_audio_policy_configuration.xml
index 5e1822a..d342aea 100644
--- a/services/audiopolicy/tests/resources/test_audio_policy_configuration.xml
+++ b/services/audiopolicy/tests/resources/test_audio_policy_configuration.xml
@@ -71,6 +71,9 @@
                 <devicePort tagName="BT SCO Headset Mic" type="AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET"
                             role="source" address="hfp_client_in">
                 </devicePort>
+                <devicePort tagName="BT A2DP Out" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP" role="sink"
+                            encodedFormats="AUDIO_FORMAT_LDAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_AAC AUDIO_FORMAT_SBC">
+                </devicePort>
             </devicePorts>
             <routes>
                 <route type="mix" sink="Speaker"
@@ -85,6 +88,8 @@
                        sources="mixport_bt_hfp_output,voip_rx"/>
                 <route type="mix" sink="mixport_bt_hfp_input"
                        sources="BT SCO Headset Mic"/>
+                <route type="mix" sink="BT A2DP Out"
+                       sources="primary output"/>
             </routes>
         </module>