diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 90f984e..87bf81e 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -435,7 +435,7 @@
         for (; i < mPlaybackThreads.size(); ++i) {
             PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
             Mutex::Autolock _tl(thread->mLock);
-            sp<PlaybackThread::Track> track = thread->getTrackById_l(trackId);
+            sp<IAfTrack> track = thread->getTrackById_l(trackId);
             if (track != nullptr) {
                 ALOGD("%s trackId: %u", __func__, trackId);
                 updateSecondaryOutputsForTrack_l(track.get(), thread, secondaryOutputs);
@@ -1107,7 +1107,7 @@
     CreateTrackInput input = VALUE_OR_RETURN_STATUS(CreateTrackInput::fromAidl(_input));
     CreateTrackOutput output;
 
-    sp<PlaybackThread::Track> track;
+    sp<IAfTrack> track;
     sp<Client> client;
     status_t lStatus;
     audio_stream_type_t streamType;
@@ -1297,7 +1297,7 @@
         AudioSystem::moveEffectsToIo(effectIds, effectThreadId);
     }
 
-    output.audioTrack = PlaybackThread::Track::createIAudioTrackAdapter(track);
+    output.audioTrack = IAfTrack::createIAudioTrackAdapter(track);
     _output = VALUE_OR_FATAL(output.toAidl());
 
 Exit:
@@ -2549,7 +2549,7 @@
     output.buffers = recordTrack->getBuffers();
     output.portId = portId;
 
-    output.audioRecord = RecordThread::RecordTrack::createIAudioRecordAdapter(recordTrack);
+    output.audioRecord = IAfRecordTrack::createIAudioRecordAdapter(recordTrack);
     _output = VALUE_OR_FATAL(output.toAidl());
 
 Exit:
@@ -3899,7 +3899,7 @@
 }
 
 void AudioFlinger::updateSecondaryOutputsForTrack_l(
-        PlaybackThread::Track* track,
+        IAfTrack* track,
         PlaybackThread* thread,
         const std::vector<audio_io_handle_t> &secondaryOutputs) const {
     TeePatches teePatches;
@@ -3989,14 +3989,14 @@
         patchTrack->setPeerProxy(patchRecord, true /* holdReference */);
         patchRecord->setPeerProxy(patchTrack, false /* holdReference */);
     }
-    track->setTeePatchesToUpdate(std::move(teePatches));
+    track->setTeePatchesToUpdate(&teePatches);  // TODO(b/288339104) void* to std::move()
 }
 
 sp<audioflinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
                                     audio_session_t triggerSession,
                                     audio_session_t listenerSession,
                                     const audioflinger::SyncEventCallback& callBack,
-                                    const wp<RefBase>& cookie)
+                                    const wp<IAfTrackBase>& cookie)
 {
     Mutex::Autolock _l(mLock);
 
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index ba6200f..614b452 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -121,6 +121,7 @@
 
 // include AudioFlinger component interfaces
 #include "IAfEffect.h"
+#include "IAfTrack.h"
 
 namespace android {
 
@@ -397,7 +398,7 @@
                                         audio_session_t triggerSession,
                                         audio_session_t listenerSession,
                                         const audioflinger::SyncEventCallback& callBack,
-                                        const wp<RefBase>& cookie);
+                                        const wp<IAfTrackBase>& cookie);
 
     bool        btNrecIsOff() const { return mBtNrecIsOff.load(); }
 
@@ -729,7 +730,7 @@
               ThreadBase *hapticPlaybackThread_l() const;
 
               void updateSecondaryOutputsForTrack_l(
-                      PlaybackThread::Track* track,
+                      IAfTrack* track,
                       PlaybackThread* thread,
                       const std::vector<audio_io_handle_t>& secondaryOutputs) const;
 
diff --git a/services/audioflinger/IAfTrack.h b/services/audioflinger/IAfTrack.h
new file mode 100644
index 0000000..65d265a
--- /dev/null
+++ b/services/audioflinger/IAfTrack.h
@@ -0,0 +1,379 @@
+/*
+ * Copyright (C) 2023 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+namespace android {
+
+// Common interface to all Playback and Record tracks.
+class IAfTrackBase : public virtual RefBase {
+public:
+    enum track_state : int32_t {
+        IDLE,
+        FLUSHED,  // for PlaybackTracks only
+        STOPPED,
+        // next 2 states are currently used for fast tracks
+        // and offloaded tracks only
+        STOPPING_1,  // waiting for first underrun
+        STOPPING_2,  // waiting for presentation complete
+        RESUMING,    // for PlaybackTracks only
+        ACTIVE,
+        PAUSING,
+        PAUSED,
+        STARTING_1,  // for RecordTrack only
+        STARTING_2,  // for RecordTrack only
+    };
+
+    // where to allocate the data buffer
+    enum alloc_type {
+        ALLOC_CBLK,      // allocate immediately after control block
+        ALLOC_READONLY,  // allocate from a separate read-only heap per thread
+        ALLOC_PIPE,      // do not allocate; use the pipe buffer
+        ALLOC_LOCAL,     // allocate a local buffer
+        ALLOC_NONE,      // do not allocate:use the buffer passed to TrackBase constructor
+    };
+
+    enum track_type {
+        TYPE_DEFAULT,
+        TYPE_OUTPUT,
+        TYPE_PATCH,
+    };
+
+    virtual status_t initCheck() const = 0;
+    virtual status_t start(
+            AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
+            audio_session_t triggerSession = AUDIO_SESSION_NONE) = 0;
+    virtual void stop() = 0;
+    virtual sp<IMemory> getCblk() const = 0;
+    virtual audio_track_cblk_t* cblk() const = 0;
+    virtual audio_session_t sessionId() const = 0;
+    virtual uid_t uid() const = 0;
+    virtual pid_t creatorPid() const = 0;
+    virtual uint32_t sampleRate() const = 0;
+    virtual size_t frameSize() const = 0;
+    virtual audio_port_handle_t portId() const = 0;
+    virtual status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) = 0;
+    virtual track_state state() const = 0;
+    virtual void setState(track_state state) = 0;
+    virtual sp<IMemory> getBuffers() const = 0;
+    virtual void* buffer() const = 0;
+    virtual size_t bufferSize() const = 0;
+    virtual bool isFastTrack() const = 0;
+    virtual bool isDirect() const = 0;
+    virtual bool isOutputTrack() const = 0;
+    virtual bool isPatchTrack() const = 0;
+    virtual bool isExternalTrack() const = 0;
+
+    virtual void invalidate() = 0;
+    virtual bool isInvalid() const = 0;
+
+    virtual void terminate() = 0;
+    virtual bool isTerminated() const = 0;
+
+    virtual audio_attributes_t attributes() const = 0;
+    virtual bool isSpatialized() const = 0;
+    virtual bool isBitPerfect() const = 0;
+
+    // not currently implemented in TrackBase, but overridden.
+    virtual void destroy() {};  // MmapTrack doesn't implement.
+    virtual void appendDumpHeader(String8& result) const = 0;
+    virtual void appendDump(String8& result, bool active) const = 0;
+
+    // Dup with AudioBufferProvider interface
+    virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0;
+    virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer) = 0;
+
+    // Added for RecordTrack and OutputTrack
+    // TODO(b/288339104) type
+    virtual wp<Thread> thread() const = 0;
+    virtual const sp<ServerProxy>& serverProxy() const = 0;
+
+    // TEE_SINK
+    virtual void dumpTee(int fd __unused, const std::string& reason __unused) const {};
+
+    /** returns the buffer contents size converted to time in milliseconds
+     * for PCM Playback or Record streaming tracks. The return value is zero for
+     * PCM static tracks and not defined for non-PCM tracks.
+     *
+     * This may be called without the thread lock.
+     */
+    virtual double bufferLatencyMs() const = 0;
+
+    /** returns whether the track supports server latency computation.
+     * This is set in the constructor and constant throughout the track lifetime.
+     */
+    virtual bool isServerLatencySupported() const = 0;
+
+    /** computes the server latency for PCM Playback or Record track
+     * to the device sink/source.  This is the time for the next frame in the track buffer
+     * written or read from the server thread to the device source or sink.
+     *
+     * This may be called without the thread lock, but latencyMs and fromTrack
+     * may be not be synchronized. For example PatchPanel may not obtain the
+     * thread lock before calling.
+     *
+     * \param latencyMs on success is set to the latency in milliseconds of the
+     *        next frame written/read by the server thread to/from the track buffer
+     *        from the device source/sink.
+     * \param fromTrack on success is set to true if latency was computed directly
+     *        from the track timestamp; otherwise set to false if latency was
+     *        estimated from the server timestamp.
+     *        fromTrack may be nullptr or omitted if not required.
+     *
+     * \returns OK or INVALID_OPERATION on failure.
+     */
+    virtual status_t getServerLatencyMs(double* latencyMs, bool* fromTrack = nullptr) const = 0;
+
+    /** computes the total client latency for PCM Playback or Record tracks
+     * for the next client app access to the device sink/source; i.e. the
+     * server latency plus the buffer latency.
+     *
+     * This may be called without the thread lock, but latencyMs and fromTrack
+     * may be not be synchronized. For example PatchPanel may not obtain the
+     * thread lock before calling.
+     *
+     * \param latencyMs on success is set to the latency in milliseconds of the
+     *        next frame written/read by the client app to/from the track buffer
+     *        from the device sink/source.
+     * \param fromTrack on success is set to true if latency was computed directly
+     *        from the track timestamp; otherwise set to false if latency was
+     *        estimated from the server timestamp.
+     *        fromTrack may be nullptr or omitted if not required.
+     *
+     * \returns OK or INVALID_OPERATION on failure.
+     */
+    virtual status_t getTrackLatencyMs(double* latencyMs, bool* fromTrack = nullptr) const = 0;
+
+    // TODO: Consider making this external.
+    struct FrameTime {
+        int64_t frames;
+        int64_t timeNs;
+    };
+
+    // KernelFrameTime is updated per "mix" period even for non-pcm tracks.
+    virtual void getKernelFrameTime(FrameTime* ft) const = 0;
+
+    virtual audio_format_t format() const = 0;
+    virtual int id() const = 0;
+
+    virtual const char* getTrackStateAsString() const = 0;
+
+    // Called by the PlaybackThread to indicate that the track is becoming active
+    // and a new interval should start with a given device list.
+    virtual void logBeginInterval(const std::string& devices) = 0;
+
+    // Called by the PlaybackThread to indicate the track is no longer active.
+    virtual void logEndInterval() = 0;
+
+    // Called to tally underrun frames in playback.
+    virtual void tallyUnderrunFrames(size_t frames) = 0;
+
+    virtual audio_channel_mask_t channelMask() const = 0;
+
+    /** @return true if the track has changed (metadata or volume) since
+     *          the last time this function was called,
+     *          true if this function was never called since the track creation,
+     *          false otherwise.
+     *  Thread safe.
+     */
+    virtual bool readAndClearHasChanged() = 0;
+
+    /** Set that a metadata has changed and needs to be notified to backend. Thread safe. */
+    virtual void setMetadataHasChanged() = 0;
+
+    /**
+     * For RecordTrack
+     * TODO(b/288339104) either use this or add asRecordTrack or asTrack etc.
+     */
+    virtual void handleSyncStartEvent(const sp<audioflinger::SyncEvent>& event __unused){};
+
+    // For Thread use, fast tracks and offloaded tracks only
+    // TODO(b/288339104) rearrange to IAfTrack.
+    virtual bool isStopped() const = 0;
+    virtual bool isStopping() const = 0;
+    virtual bool isStopping_1() const = 0;
+    virtual bool isStopping_2() const = 0;
+};
+
+// Common interface for Playback tracks.
+class IAfTrack : public virtual IAfTrackBase {
+public:
+    // createIAudioTrackAdapter() is a static constructor which creates an
+    // IAudioTrack AIDL interface adapter from the Track object that
+    // may be passed back to the client (if needed).
+    //
+    // Only one AIDL IAudioTrack interface adapter should be created per Track.
+    static sp<media::IAudioTrack> createIAudioTrackAdapter(const sp<IAfTrack>& track);
+
+    virtual void pause() = 0;
+    virtual void flush() = 0;
+    virtual audio_stream_type_t streamType() const = 0;
+    virtual bool isOffloaded() const = 0;
+    virtual bool isOffloadedOrDirect() const = 0;
+    virtual bool isStatic() const = 0;
+    virtual status_t setParameters(const String8& keyValuePairs) = 0;
+    virtual status_t selectPresentation(int presentationId, int programId) = 0;
+    virtual status_t attachAuxEffect(int EffectId) = 0;
+    virtual void setAuxBuffer(int EffectId, int32_t* buffer) = 0;
+    virtual int32_t* auxBuffer() const = 0;
+    virtual void setMainBuffer(float* buffer) = 0;
+    virtual float* mainBuffer() const = 0;
+    virtual int auxEffectId() const = 0;
+    virtual status_t getTimestamp(AudioTimestamp& timestamp) = 0;
+    virtual void signal() = 0;
+    virtual status_t getDualMonoMode(audio_dual_mono_mode_t* mode) const = 0;
+    virtual status_t setDualMonoMode(audio_dual_mono_mode_t mode) = 0;
+    virtual status_t getAudioDescriptionMixLevel(float* leveldB) const = 0;
+    virtual status_t setAudioDescriptionMixLevel(float leveldB) = 0;
+    virtual status_t getPlaybackRateParameters(audio_playback_rate_t* playbackRate) const = 0;
+    virtual status_t setPlaybackRateParameters(const audio_playback_rate_t& playbackRate) = 0;
+
+    // implement FastMixerState::VolumeProvider interface
+    virtual gain_minifloat_packed_t getVolumeLR() const = 0;
+
+    // implement volume handling.
+    virtual media::VolumeShaper::Status applyVolumeShaper(
+            const sp<media::VolumeShaper::Configuration>& configuration,
+            const sp<media::VolumeShaper::Operation>& operation) = 0;
+    virtual sp<media::VolumeShaper::State> getVolumeShaperState(int id) const = 0;
+    virtual sp<media::VolumeHandler> getVolumeHandler() const = 0;
+    /** Set the computed normalized final volume of the track.
+     * !masterMute * masterVolume * streamVolume * averageLRVolume */
+    virtual void setFinalVolume(float volumeLeft, float volumeRight) = 0;
+    virtual float getFinalVolume() const = 0;
+    virtual void getFinalVolume(float* left, float* right) const = 0;
+
+    using SourceMetadatas = std::vector<playback_track_metadata_v7_t>;
+    using MetadataInserter = std::back_insert_iterator<SourceMetadatas>;
+    /** Copy the track metadata in the provided iterator. Thread safe. */
+    virtual void copyMetadataTo(MetadataInserter& backInserter) const = 0;
+
+    /** Return haptic playback of the track is enabled or not, used in mixer. */
+    virtual bool getHapticPlaybackEnabled() const = 0;
+    /** Set haptic playback of the track is enabled or not, should be
+     * set after query or get callback from vibrator service */
+    virtual void setHapticPlaybackEnabled(bool hapticPlaybackEnabled) = 0;
+    /** Return at what intensity to play haptics, used in mixer. */
+    virtual os::HapticScale getHapticIntensity() const = 0;
+    /** Return the maximum amplitude allowed for haptics data, used in mixer. */
+    virtual float getHapticMaxAmplitude() const = 0;
+    /** Set intensity of haptic playback, should be set after querying vibrator service. */
+    virtual void setHapticIntensity(os::HapticScale hapticIntensity) = 0;
+    /** Set maximum amplitude allowed for haptic data, should be set after querying
+     *  vibrator service.
+     */
+    virtual void setHapticMaxAmplitude(float maxAmplitude) = 0;
+    virtual sp<os::ExternalVibration> getExternalVibration() const = 0;
+
+    // This function should be called with holding thread lock.
+    virtual void updateTeePatches() = 0;
+
+    // TODO(b/288339104) type
+    virtual void setTeePatchesToUpdate(
+            const void* teePatchesToUpdate /* TeePatches& teePatchesToUpdate */) = 0;
+
+    static bool checkServerLatencySupported(audio_format_t format, audio_output_flags_t flags) {
+        return audio_is_linear_pcm(format) && (flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) == 0;
+    }
+
+    virtual audio_output_flags_t getOutputFlags() const = 0;
+    virtual float getSpeed() const = 0;
+
+    /**
+     * Updates the mute state and notifies the audio service. Call this only when holding player
+     * thread lock.
+     */
+    virtual void processMuteEvent_l(
+            const sp<IAudioManager>& audioManager, mute_state_t muteState) = 0;
+
+    virtual void triggerEvents(AudioSystem::sync_event_t type) = 0;
+
+    virtual void disable() = 0;
+    virtual int& fastIndex() = 0;
+    virtual bool isPlaybackRestricted() const = 0;
+};
+
+// playback track, used by DuplicatingThread
+class IAfOutputTrack : public virtual IAfTrack {
+public:
+    virtual ssize_t write(void* data, uint32_t frames) = 0;
+    virtual bool bufferQueueEmpty() const = 0;
+    virtual bool isActive() const = 0;
+
+    /** Set the metadatas of the upstream tracks. Thread safe. */
+    virtual void setMetadatas(const SourceMetadatas& metadatas) = 0;
+    /** returns client timestamp to the upstream duplicating thread. */
+    virtual ExtendedTimestamp getClientProxyTimestamp() const = 0;
+};
+
+class IAfMmapTrack : public virtual IAfTrackBase {
+public:
+    // protected by MMapThread::mLock
+    virtual void setSilenced_l(bool silenced) = 0;
+    // protected by MMapThread::mLock
+    virtual bool isSilenced_l() const = 0;
+    // protected by MMapThread::mLock
+    virtual bool getAndSetSilencedNotified_l() = 0;
+
+    /**
+     * Updates the mute state and notifies the audio service. Call this only when holding player
+     * thread lock.
+     */
+    virtual void processMuteEvent_l(  // see IAfTrack
+            const sp<IAudioManager>& audioManager, mute_state_t muteState) = 0;
+};
+
+class IAfRecordTrack : public virtual IAfTrackBase {
+public:
+    // createIAudioRecordAdapter() is a static constructor which creates an
+    // IAudioRecord AIDL interface adapter from the RecordTrack object that
+    // may be passed back to the client (if needed).
+    //
+    // Only one AIDL IAudioRecord interface adapter should be created per RecordTrack.
+    static sp<media::IAudioRecord> createIAudioRecordAdapter(const sp<IAfRecordTrack>& recordTrack);
+
+    // clear the buffer overflow flag
+    virtual void clearOverflow() = 0;
+    // set the buffer overflow flag and return previous value
+    virtual bool setOverflow() = 0;
+
+    // TODO(b/288339104) handleSyncStartEvent in IAfTrackBase should move here.
+    virtual void clearSyncStartEvent() = 0;
+    virtual void updateTrackFrameInfo(
+            int64_t trackFramesReleased, int64_t sourceFramesRead, uint32_t halSampleRate,
+            const ExtendedTimestamp& timestamp) = 0;
+
+    virtual void setSilenced(bool silenced) = 0;
+    virtual bool isSilenced() const = 0;
+    virtual status_t getActiveMicrophones(
+            std::vector<media::MicrophoneInfoFw>* activeMicrophones) const = 0;
+
+    virtual status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction) = 0;
+    virtual status_t setPreferredMicrophoneFieldDimension(float zoom) = 0;
+    virtual status_t shareAudioHistory(
+            const std::string& sharedAudioPackageName, int64_t sharedAudioStartMs) = 0;
+    virtual int32_t startFrames() const = 0;
+
+    static bool checkServerLatencySupported(audio_format_t format, audio_input_flags_t flags) {
+        return audio_is_linear_pcm(format) && (flags & AUDIO_INPUT_FLAG_HW_AV_SYNC) == 0;
+    }
+
+    using SinkMetadatas = std::vector<record_track_metadata_v7_t>;
+    using MetadataInserter = std::back_insert_iterator<SinkMetadatas>;
+    virtual void copyMetadataTo(MetadataInserter& backInserter) const = 0; // see IAfTrack
+};
+
+}  // namespace android
diff --git a/services/audioflinger/MmapTracks.h b/services/audioflinger/MmapTracks.h
index cb46c52..0ab1adb 100644
--- a/services/audioflinger/MmapTracks.h
+++ b/services/audioflinger/MmapTracks.h
@@ -20,7 +20,7 @@
 #endif
 
 // playback track
-class MmapTrack : public TrackBase {
+class MmapTrack : public TrackBase, public IAfMmapTrack {
 public:
                 MmapTrack(ThreadBase *thread,
                             const audio_attributes_t& attr,
@@ -32,26 +32,25 @@
                             const android::content::AttributionSourceState& attributionSource,
                             pid_t creatorPid,
                             audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
-    virtual             ~MmapTrack();
+    ~MmapTrack() override;
 
-                        // TrackBase virtual
-    virtual status_t    initCheck() const;
-    virtual status_t    start(AudioSystem::sync_event_t event,
-                              audio_session_t triggerSession);
-    virtual void        stop();
-    virtual bool        isFastTrack() const { return false; }
-            bool        isDirect() const override { return true; }
+    status_t initCheck() const final;
+    status_t start(
+            AudioSystem::sync_event_t event, audio_session_t triggerSession) final;
+    void stop() final;
+    bool isFastTrack() const final { return false; }
+    bool isDirect() const final { return true; }
 
-            void        appendDumpHeader(String8& result);
-            void        appendDump(String8& result, bool active);
+    void appendDumpHeader(String8& result) const final;
+    void appendDump(String8& result, bool active) const final;
 
                         // protected by MMapThread::mLock
-            void        setSilenced_l(bool silenced) { mSilenced = silenced;
+    void setSilenced_l(bool silenced) final { mSilenced = silenced;
                                                        mSilencedNotified = false;}
                         // protected by MMapThread::mLock
-            bool        isSilenced_l() const { return mSilenced; }
+    bool isSilenced_l() const final { return mSilenced; }
                         // protected by MMapThread::mLock
-            bool        getAndSetSilencedNotified_l() { bool silencedNotified = mSilencedNotified;
+    bool getAndSetSilencedNotified_l() final { bool silencedNotified = mSilencedNotified;
                                                         mSilencedNotified = true;
                                                         return silencedNotified; }
 
@@ -61,7 +60,7 @@
      */
     void processMuteEvent_l(const sp<IAudioManager>& audioManager,
                             mute_state_t muteState)
-                            REQUIRES(AudioFlinger::MmapPlaybackThread::mLock);
+                            REQUIRES(AudioFlinger::MmapPlaybackThread::mLock) final;
 private:
     friend class MmapThread;
 
@@ -72,11 +71,11 @@
     // releaseBuffer() not overridden
 
     // ExtendedAudioBufferProvider interface
-    virtual size_t framesReady() const;
-    virtual int64_t framesReleased() const;
-    virtual void onTimestamp(const ExtendedTimestamp &timestamp);
+    size_t framesReady() const final;
+    int64_t framesReleased() const final;
+    void onTimestamp(const ExtendedTimestamp &timestamp) final;
 
-    pid_t mPid;
+    const pid_t mPid;
     bool  mSilenced;            // protected by MMapThread::mLock
     bool  mSilencedNotified;    // protected by MMapThread::mLock
 
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index 837d045..e323cdd 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -679,7 +679,7 @@
     // If so, do a frame diff and time difference computation to estimate
     // the total patch latency. This requires that frame counts are reported by the
     // HAL are matched properly in the case of record overruns and playback underruns.
-    ThreadBase::TrackBase::FrameTime recordFT{}, playFT{};
+    IAfTrack::FrameTime recordFT{}, playFT{};
     recordTrack->getKernelFrameTime(&recordFT);
     playbackTrack->getKernelFrameTime(&playFT);
     if (recordFT.timeNs > 0 && playFT.timeNs > 0) {
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 7ff2394..ee2dd9b 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -67,15 +67,8 @@
 };
 
 // playback track
-class Track : public TrackBase, public VolumeProvider {
+class Track : public TrackBase, public virtual IAfTrack, public VolumeProvider {
 public:
-    // createIAudioTrackAdapter() is a static constructor which creates an
-    // IAudioTrack AIDL interface adapter from the Track object that
-    // may be passed back to the client (if needed).
-    //
-    // Only one AIDL IAudioTrack interface wrapper should be created per Track.
-    static sp<media::IAudioTrack> createIAudioTrackAdapter(const sp<Track>& track);
-
                         Track(  PlaybackThread *thread,
                                 const sp<Client>& client,
                                 audio_stream_type_t streamType,
@@ -99,72 +92,66 @@
                                 float speed = 1.0f,
                                 bool isSpatialized = false,
                                 bool isBitPerfect = false);
-    virtual             ~Track();
-    virtual status_t    initCheck() const;
-
-            void        appendDumpHeader(String8& result);
-            void        appendDump(String8& result, bool active);
-    virtual status_t    start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
-                              audio_session_t triggerSession = AUDIO_SESSION_NONE);
-    virtual void        stop();
-            void        pause();
-
-            void        flush();
-            void        destroy();
-
-    virtual uint32_t    sampleRate() const;
-
-            audio_stream_type_t streamType() const {
+    ~Track() override;
+    status_t initCheck() const final;
+    void appendDumpHeader(String8& result) const final;
+    void appendDump(String8& result, bool active) const final;
+    status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
+            audio_session_t triggerSession = AUDIO_SESSION_NONE) override;
+    void stop() override;
+    void pause() final;
+    void flush() final;
+    void destroy() final;
+    uint32_t sampleRate() const final;
+    audio_stream_type_t streamType() const final {
                 return mStreamType;
             }
-            bool        isOffloaded() const
+    bool isOffloaded() const final
                                 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }
-            bool        isDirect() const override
+    bool isDirect() const final
                                 { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; }
-            bool        isOffloadedOrDirect() const { return (mFlags
+    bool isOffloadedOrDirect() const final { return (mFlags
                             & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD
                                     | AUDIO_OUTPUT_FLAG_DIRECT)) != 0; }
-            bool        isStatic() const { return  mSharedBuffer.get() != nullptr; }
+    bool isStatic() const final { return  mSharedBuffer.get() != nullptr; }
 
-            status_t    setParameters(const String8& keyValuePairs);
-            status_t    selectPresentation(int presentationId, int programId);
-            status_t    attachAuxEffect(int EffectId);
-            void        setAuxBuffer(int EffectId, int32_t *buffer);
-            int32_t     *auxBuffer() const { return mAuxBuffer; }
-            void        setMainBuffer(float *buffer) { mMainBuffer = buffer; }
-            float       *mainBuffer() const { return mMainBuffer; }
-            int         auxEffectId() const { return mAuxEffectId; }
-    virtual status_t    getTimestamp(AudioTimestamp& timestamp);
-            void        signal();
-            status_t    getDualMonoMode(audio_dual_mono_mode_t* mode);
-            status_t    setDualMonoMode(audio_dual_mono_mode_t mode);
-            status_t    getAudioDescriptionMixLevel(float* leveldB);
-            status_t    setAudioDescriptionMixLevel(float leveldB);
-            status_t    getPlaybackRateParameters(audio_playback_rate_t* playbackRate);
-            status_t    setPlaybackRateParameters(const audio_playback_rate_t& playbackRate);
+    status_t setParameters(const String8& keyValuePairs) final;
+    status_t selectPresentation(int presentationId, int programId) final;
+    status_t attachAuxEffect(int EffectId) final;
+    void setAuxBuffer(int EffectId, int32_t* buffer) final;
+    int32_t* auxBuffer() const final { return mAuxBuffer; }
+    void setMainBuffer(float* buffer) final { mMainBuffer = buffer; }
+    float* mainBuffer() const final { return mMainBuffer; }
+    int auxEffectId() const final { return mAuxEffectId; }
+    status_t getTimestamp(AudioTimestamp& timestamp) final;
+    void signal() final;
+    status_t getDualMonoMode(audio_dual_mono_mode_t* mode) const final;
+    status_t setDualMonoMode(audio_dual_mono_mode_t mode) final;
+    status_t getAudioDescriptionMixLevel(float* leveldB) const final;
+    status_t setAudioDescriptionMixLevel(float leveldB) final;
+    status_t getPlaybackRateParameters(audio_playback_rate_t* playbackRate) const final;
+    status_t setPlaybackRateParameters(const audio_playback_rate_t& playbackRate) final;
 
-// implement FastMixerState::VolumeProvider interface
-    virtual gain_minifloat_packed_t getVolumeLR();
+    // implement FastMixerState::VolumeProvider interface
+    gain_minifloat_packed_t getVolumeLR() const final;
 
-            status_t    setSyncEvent(const sp<audioflinger::SyncEvent>& event) override;
-
-    virtual bool        isFastTrack() const { return (mFlags & AUDIO_OUTPUT_FLAG_FAST) != 0; }
-
-            double      bufferLatencyMs() const override {
+    status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) final;
+    bool isFastTrack() const final { return (mFlags & AUDIO_OUTPUT_FLAG_FAST) != 0; }
+    double bufferLatencyMs() const final {
                             return isStatic() ? 0. : TrackBase::bufferLatencyMs();
                         }
 
-// implement volume handling.
+    // implement volume handling.
     media::VolumeShaper::Status applyVolumeShaper(
                                 const sp<media::VolumeShaper::Configuration>& configuration,
                                 const sp<media::VolumeShaper::Operation>& operation);
-    sp<media::VolumeShaper::State> getVolumeShaperState(int id);
-    sp<media::VolumeHandler>   getVolumeHandler() { return mVolumeHandler; }
+    sp<media::VolumeShaper::State> getVolumeShaperState(int id) const final;
+    sp<media::VolumeHandler> getVolumeHandler() const final{ return mVolumeHandler; }
     /** Set the computed normalized final volume of the track.
      * !masterMute * masterVolume * streamVolume * averageLRVolume */
-    void                setFinalVolume(float volumeLeft, float volumeRight);
-    float               getFinalVolume() const { return mFinalVolume; }
-    void                getFinalVolume(float* left, float* right) const {
+    void setFinalVolume(float volumeLeft, float volumeRight) final;
+    float getFinalVolume() const final { return mFinalVolume; }
+    void getFinalVolume(float* left, float* right) const final {
                             *left = mFinalVolumeLeft;
                             *right = mFinalVolumeRight;
     }
@@ -172,21 +159,22 @@
     using SourceMetadatas = std::vector<playback_track_metadata_v7_t>;
     using MetadataInserter = std::back_insert_iterator<SourceMetadatas>;
     /** Copy the track metadata in the provided iterator. Thread safe. */
-    virtual void    copyMetadataTo(MetadataInserter& backInserter) const;
+    void copyMetadataTo(MetadataInserter& backInserter) const override;
+
 
             /** Return haptic playback of the track is enabled or not, used in mixer. */
-            bool    getHapticPlaybackEnabled() const { return mHapticPlaybackEnabled; }
+    bool getHapticPlaybackEnabled() const final { return mHapticPlaybackEnabled; }
             /** Set haptic playback of the track is enabled or not, should be
              *  set after query or get callback from vibrator service */
-            void    setHapticPlaybackEnabled(bool hapticPlaybackEnabled) {
+    void setHapticPlaybackEnabled(bool hapticPlaybackEnabled) final {
                 mHapticPlaybackEnabled = hapticPlaybackEnabled;
             }
             /** Return at what intensity to play haptics, used in mixer. */
-            os::HapticScale getHapticIntensity() const { return mHapticIntensity; }
+    os::HapticScale getHapticIntensity() const final { return mHapticIntensity; }
             /** Return the maximum amplitude allowed for haptics data, used in mixer. */
-            float getHapticMaxAmplitude() const { return mHapticMaxAmplitude; }
+    float getHapticMaxAmplitude() const final { return mHapticMaxAmplitude; }
             /** Set intensity of haptic playback, should be set after querying vibrator service. */
-            void    setHapticIntensity(os::HapticScale hapticIntensity) {
+    void setHapticIntensity(os::HapticScale hapticIntensity) final {
                 if (os::isValidHapticScale(hapticIntensity)) {
                     mHapticIntensity = hapticIntensity;
                     setHapticPlaybackEnabled(mHapticIntensity != os::HapticScale::MUTE);
@@ -195,16 +183,19 @@
             /** Set maximum amplitude allowed for haptic data, should be set after querying
              *  vibrator service.
              */
-            void    setHapticMaxAmplitude(float maxAmplitude) {
+    void setHapticMaxAmplitude(float maxAmplitude) final {
                 mHapticMaxAmplitude = maxAmplitude;
             }
-            sp<os::ExternalVibration> getExternalVibration() const { return mExternalVibration; }
+    sp<os::ExternalVibration> getExternalVibration() const final { return mExternalVibration; }
 
             // This function should be called with holding thread lock.
-            void    updateTeePatches();
-            void    setTeePatchesToUpdate(TeePatches teePatchesToUpdate);
+    void updateTeePatches() final;
+    void setTeePatchesToUpdate(const void* teePatchesToUpdate) final {
+        setTeePatchesToUpdate(*reinterpret_cast<const TeePatches*>(teePatchesToUpdate));
+    }
+    void setTeePatchesToUpdate(TeePatches teePatchesToUpdate);
 
-    void tallyUnderrunFrames(size_t frames) override {
+    void tallyUnderrunFrames(size_t frames) final {
        if (isOut()) { // we expect this from output tracks only
            mAudioTrackServerProxy->tallyUnderrunFrames(frames);
            // Fetch absolute numbers from AudioTrackShared as it counts
@@ -215,22 +206,16 @@
        }
     }
 
-    static bool checkServerLatencySupported(
-            audio_format_t format, audio_output_flags_t flags) {
-        return audio_is_linear_pcm(format)
-                && (flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) == 0;
-    }
-
-    audio_output_flags_t getOutputFlags() const { return mFlags; }
-    float getSpeed() const { return mSpeed; }
-    bool isSpatialized() const override { return mIsSpatialized; }
-    bool isBitPerfect() const override { return mIsBitPerfect; }
+    audio_output_flags_t getOutputFlags() const final { return mFlags; }
+    float getSpeed() const final { return mSpeed; }
+    bool isSpatialized() const final { return mIsSpatialized; }
+    bool isBitPerfect() const final { return mIsBitPerfect; }
 
     /**
      * Updates the mute state and notifies the audio service. Call this only when holding player
      * thread lock.
      */
-    void processMuteEvent_l(const sp<IAudioManager>& audioManager, mute_state_t muteState);
+    void processMuteEvent_l(const sp<IAudioManager>& audioManager, mute_state_t muteState) final;
 
 protected:
     // for numerous
@@ -246,9 +231,9 @@
     void releaseBuffer(AudioBufferProvider::Buffer* buffer) override;
 
     // ExtendedAudioBufferProvider interface
-    virtual size_t framesReady() const;
-    virtual int64_t framesReleased() const;
-    virtual void onTimestamp(const ExtendedTimestamp &timestamp);
+    size_t framesReady() const override;
+    int64_t framesReleased() const override;
+    void onTimestamp(const ExtendedTimestamp &timestamp) override;
 
     bool isPausing() const { return mState == PAUSING; }
     bool isPaused() const { return mState == PAUSED; }
@@ -289,13 +274,11 @@
     void signalClientFlag(int32_t flag);
 
 public:
-    void triggerEvents(AudioSystem::sync_event_t type);
-    virtual void invalidate();
-    void disable();
-
-    int fastIndex() const { return mFastIndex; }
-
-    bool isPlaybackRestricted() const {
+    void triggerEvents(AudioSystem::sync_event_t type) final;
+    void invalidate() final;
+    void disable() final;
+    int& fastIndex() final { return mFastIndex; }
+    bool isPlaybackRestricted() const final {
         // The monitor is only created for tracks that can be silenced.
         return mOpPlayAudioMonitor ? !mOpPlayAudioMonitor->hasOpPlayAudio() : false; }
 
@@ -398,7 +381,7 @@
 
 
 // playback track, used by DuplicatingThread
-class OutputTrack : public Track {
+class OutputTrack : public Track, public IAfOutputTrack {
 public:
 
     class Buffer : public AudioBufferProvider::Buffer {
@@ -413,22 +396,21 @@
                                 audio_channel_mask_t channelMask,
                                 size_t frameCount,
                                 const AttributionSourceState& attributionSource);
-    virtual             ~OutputTrack();
+    ~OutputTrack() override;
 
-    virtual status_t    start(AudioSystem::sync_event_t event =
+    status_t start(AudioSystem::sync_event_t event =
                                     AudioSystem::SYNC_EVENT_NONE,
-                             audio_session_t triggerSession = AUDIO_SESSION_NONE);
-    virtual void        stop();
-            ssize_t     write(void* data, uint32_t frames);
-            bool        bufferQueueEmpty() const { return mBufferQueue.size() == 0; }
-            bool        isActive() const { return mActive; }
-    const wp<ThreadBase>& thread() const { return mThread; }
+                             audio_session_t triggerSession = AUDIO_SESSION_NONE) final;
+    void stop() final;
+    ssize_t write(void* data, uint32_t frames) final;
+    bool bufferQueueEmpty() const final { return mBufferQueue.size() == 0; }
+    bool isActive() const final { return mActive; }
 
-            void        copyMetadataTo(MetadataInserter& backInserter) const override;
+    void copyMetadataTo(MetadataInserter& backInserter) const final;
     /** Set the metadatas of the upstream tracks. Thread safe. */
-            void        setMetadatas(const SourceMetadatas& metadatas);
+    void setMetadatas(const SourceMetadatas& metadatas) final;
     /** returns client timestamp to the upstream duplicating thread. */
-    ExtendedTimestamp   getClientProxyTimestamp() const {
+    ExtendedTimestamp getClientProxyTimestamp() const final {
                             // server - kernel difference is not true latency when drained
                             // i.e. mServerProxy->isDrained().
                             ExtendedTimestamp timestamp;
@@ -439,7 +421,6 @@
                             // (with mTimeNs[] filled with -1's) is returned.
                             return timestamp;
                         }
-
 private:
     status_t            obtainBuffer(AudioBufferProvider::Buffer* buffer,
                                      uint32_t waitTimeMs);
@@ -491,22 +472,21 @@
                                                                     *  as soon as possible to have
                                                                     *  the lowest possible latency
                                                                     *  even if it might glitch. */);
-    virtual             ~PatchTrack();
+    ~PatchTrack() override;
 
-            size_t      framesReady() const override;
+    size_t framesReady() const final;
 
-    virtual status_t    start(AudioSystem::sync_event_t event =
+    status_t start(AudioSystem::sync_event_t event =
                                     AudioSystem::SYNC_EVENT_NONE,
-                             audio_session_t triggerSession = AUDIO_SESSION_NONE);
+                             audio_session_t triggerSession = AUDIO_SESSION_NONE) final;
 
     // AudioBufferProvider interface
-    virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
-    virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
+    status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) final;
+    void releaseBuffer(AudioBufferProvider::Buffer* buffer) final;
 
     // PatchProxyBufferProvider interface
-    virtual status_t    obtainBuffer(Proxy::Buffer* buffer,
-                                     const struct timespec *timeOut = NULL);
-    virtual void        releaseBuffer(Proxy::Buffer* buffer);
+    status_t obtainBuffer(Proxy::Buffer* buffer, const struct timespec* timeOut = nullptr) final;
+    void releaseBuffer(Proxy::Buffer* buffer) final;
 
 private:
             void restartIfDisabled();
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index 61f36a1..d98b214 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -22,15 +22,8 @@
 #endif
 
 // record track
-class RecordTrack : public TrackBase {
+class RecordTrack : public TrackBase, public IAfRecordTrack {
 public:
-    // createIAudioRecordAdapter() is a static constructor which creates an
-    // IAudioRecord AIDL interface wrapper from the RecordTrack object that
-    // may be passed back to the client (if needed).
-    //
-    // Only one AIDL IAudioRecord interface wrapper should be created per RecordTrack.
-    static sp<media::IAudioRecord> createIAudioRecordAdapter(const sp<RecordTrack>& recordTrack);
-
                         RecordTrack(RecordThread *thread,
                                 const sp<Client>& client,
                                 const audio_attributes_t& attr,
@@ -47,57 +40,50 @@
                                 track_type type,
                                 audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE,
                                 int32_t startFrames = -1);
-    virtual             ~RecordTrack();
-    virtual status_t    initCheck() const;
+    ~RecordTrack() override;
+    status_t initCheck() const final;
 
-    virtual status_t    start(AudioSystem::sync_event_t event, audio_session_t triggerSession);
-    virtual void        stop();
+    status_t start(AudioSystem::sync_event_t event, audio_session_t triggerSession) final;
+    void stop() final;
+    void destroy() final;
+    void invalidate() final;
 
-            void        destroy();
-
-    virtual void        invalidate();
             // clear the buffer overflow flag
-            void        clearOverflow() { mOverflow = false; }
+    void clearOverflow() final { mOverflow = false; }
             // set the buffer overflow flag and return previous value
-            bool        setOverflow() { bool tmp = mOverflow; mOverflow = true;
+    bool setOverflow() final { bool tmp = mOverflow; mOverflow = true;
                                                 return tmp; }
 
-            void        appendDumpHeader(String8& result);
-            void        appendDump(String8& result, bool active);
+    void appendDumpHeader(String8& result) const final;
+    void appendDump(String8& result, bool active) const final;
 
-            void        handleSyncStartEvent(const sp<audioflinger::SyncEvent>& event);
-            void        clearSyncStartEvent();
+    void handleSyncStartEvent(const sp<audioflinger::SyncEvent>& event) final;
+    void clearSyncStartEvent() final;
 
-            void        updateTrackFrameInfo(int64_t trackFramesReleased,
+    void updateTrackFrameInfo(int64_t trackFramesReleased,
                                              int64_t sourceFramesRead,
                                              uint32_t halSampleRate,
-                                             const ExtendedTimestamp &timestamp);
+                                             const ExtendedTimestamp &timestamp) final;
 
-    virtual bool        isFastTrack() const { return (mFlags & AUDIO_INPUT_FLAG_FAST) != 0; }
-            bool        isDirect() const override
+    bool isFastTrack() const final { return (mFlags & AUDIO_INPUT_FLAG_FAST) != 0; }
+    bool isDirect() const final
                                 { return (mFlags & AUDIO_INPUT_FLAG_DIRECT) != 0; }
 
-            void        setSilenced(bool silenced) { if (!isPatchTrack()) mSilenced = silenced; }
-            bool        isSilenced() const { return mSilenced; }
+    void setSilenced(bool silenced) final { if (!isPatchTrack()) mSilenced = silenced; }
+    bool isSilenced() const final { return mSilenced; }
 
-            status_t    getActiveMicrophones(
-                    std::vector<media::MicrophoneInfoFw>* activeMicrophones);
+    status_t getActiveMicrophones(
+            std::vector<media::MicrophoneInfoFw>* activeMicrophones) const final;
 
-            status_t    setPreferredMicrophoneDirection(audio_microphone_direction_t direction);
-            status_t    setPreferredMicrophoneFieldDimension(float zoom);
-            status_t    shareAudioHistory(const std::string& sharedAudioPackageName,
-                                          int64_t sharedAudioStartMs);
-            int32_t     startFrames() { return mStartFrames; }
+    status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction) final;
+    status_t setPreferredMicrophoneFieldDimension(float zoom) final;
+    status_t shareAudioHistory(const std::string& sharedAudioPackageName,
+            int64_t sharedAudioStartMs) final;
+    int32_t startFrames() const final { return mStartFrames; }
 
-    static  bool        checkServerLatencySupported(
-                                audio_format_t format, audio_input_flags_t flags) {
-                            return audio_is_linear_pcm(format)
-                                    && (flags & AUDIO_INPUT_FLAG_HW_AV_SYNC) == 0;
-                        }
-
-            using SinkMetadatas = std::vector<record_track_metadata_v7_t>;
-            using MetadataInserter = std::back_insert_iterator<SinkMetadatas>;
-            virtual void    copyMetadataTo(MetadataInserter& backInserter) const;
+    using SinkMetadatas = std::vector<record_track_metadata_v7_t>;
+    using MetadataInserter = std::back_insert_iterator<SinkMetadatas>;
+    void copyMetadataTo(MetadataInserter& backInserter) const final;
 
 private:
     friend class AudioFlinger;  // for mState
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 16f5a98..fc09927 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -7502,7 +7502,8 @@
     if (numTracks > 0) {
         ss << ":";
         for (const auto &track : mOutputTracks) {
-            const sp<ThreadBase> thread = track->thread().promote();
+            // TODO(b/288339104) type
+            const auto thread = sp<ThreadBase>::cast(track->thread().promote());
             ss << " (" << track->id() << " : ";
             if (thread.get() != nullptr) {
                 ss << thread.get() << ", " << thread->id();
@@ -7586,7 +7587,8 @@
 {
     mWaitTimeMs = UINT_MAX;
     for (size_t i = 0; i < mOutputTracks.size(); i++) {
-        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
+        // TODO(b/288339104) type
+        const auto strong = sp<ThreadBase>::cast(mOutputTracks[i]->thread().promote());
         if (strong != 0) {
             uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
             if (waitTimeMs < mWaitTimeMs) {
@@ -7599,7 +7601,8 @@
 bool AudioFlinger::DuplicatingThread::outputsReady()
 {
     for (size_t i = 0; i < outputTracks.size(); i++) {
-        sp<ThreadBase> thread = outputTracks[i]->thread().promote();
+        // TODO(b/288339104) type
+        const auto thread = sp<ThreadBase>::cast(outputTracks[i]->thread().promote());
         if (thread == 0) {
             ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p",
                     outputTracks[i].get());
@@ -8877,10 +8880,11 @@
     sp<audioflinger::SyncEvent> strongEvent = event.promote();
 
     if (strongEvent != 0) {
-        sp<RefBase> ptr = std::any_cast<const wp<RefBase>>(strongEvent->cookie()).promote();
-        if (ptr != 0) {
-            RecordTrack *recordTrack = (RecordTrack *)ptr.get();
-            recordTrack->handleSyncStartEvent(strongEvent);
+        sp<IAfTrackBase> ptr =
+                std::any_cast<const wp<IAfTrackBase>>(strongEvent->cookie()).promote();
+        if (ptr != nullptr) {
+            // TODO(b/288339104) handleSyncStartEvent is in IAfTrackBase not IAfRecordTrack.
+            ptr->handleSyncStartEvent(strongEvent);
         }
     }
 }
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index d5b6a98..4e62b57 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -20,40 +20,8 @@
 #endif
 
 // base for record and playback
-class TrackBase : public ExtendedAudioBufferProvider, public RefBase {
-
+class TrackBase : public ExtendedAudioBufferProvider, public virtual IAfTrackBase {
 public:
-    enum track_state : int32_t {
-        IDLE,
-        FLUSHED,        // for PlaybackTracks only
-        STOPPED,
-        // next 2 states are currently used for fast tracks
-        // and offloaded tracks only
-        STOPPING_1,     // waiting for first underrun
-        STOPPING_2,     // waiting for presentation complete
-        RESUMING,       // for PlaybackTracks only
-        ACTIVE,
-        PAUSING,
-        PAUSED,
-        STARTING_1,     // for RecordTrack only
-        STARTING_2,     // for RecordTrack only
-    };
-
-    // where to allocate the data buffer
-    enum alloc_type {
-        ALLOC_CBLK,     // allocate immediately after control block
-        ALLOC_READONLY, // allocate from a separate read-only heap per thread
-        ALLOC_PIPE,     // do not allocate; use the pipe buffer
-        ALLOC_LOCAL,    // allocate a local buffer
-        ALLOC_NONE,     // do not allocate:use the buffer passed to TrackBase constructor
-    };
-
-    enum track_type {
-        TYPE_DEFAULT,
-        TYPE_OUTPUT,
-        TYPE_PATCH,
-    };
-
                         TrackBase(ThreadBase *thread,
                                 const sp<Client>& client,
                                 const audio_attributes_t& mAttr,
@@ -71,87 +39,80 @@
                                 track_type type = TYPE_DEFAULT,
                                 audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE,
                                 std::string metricsId = {});
-    virtual             ~TrackBase();
-    virtual status_t    initCheck() const;
+    ~TrackBase() override;
+    status_t initCheck() const override;
+    sp<IMemory> getCblk() const final { return mCblkMemory; }
+    audio_track_cblk_t* cblk() const final { return mCblk; }
+    audio_session_t sessionId() const final { return mSessionId; }
+    uid_t uid() const final { return mUid; }
+    pid_t creatorPid() const final { return mCreatorPid; }
+    audio_port_handle_t portId() const final { return mPortId; }
+    status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) override;
+    track_state state() const final { return mState; }
+    void setState(track_state state) final { mState = state; }
+    sp<IMemory> getBuffers() const final { return mBufferMemory; }
+    void* buffer() const final { return mBuffer; }
+    size_t bufferSize() const final { return mBufferSize; }
 
-    virtual status_t    start(AudioSystem::sync_event_t event,
-                             audio_session_t triggerSession) = 0;
-    virtual void        stop() = 0;
-            sp<IMemory> getCblk() const { return mCblkMemory; }
-            audio_track_cblk_t* cblk() const { return mCblk; }
-            audio_session_t sessionId() const { return mSessionId; }
-            uid_t       uid() const { return mUid; }
-            pid_t       creatorPid() const { return mCreatorPid; }
-
-            audio_port_handle_t portId() const { return mPortId; }
-    virtual status_t    setSyncEvent(const sp<audioflinger::SyncEvent>& event);
-
-            sp<IMemory> getBuffers() const { return mBufferMemory; }
-            void*       buffer() const { return mBuffer; }
-            size_t      bufferSize() const { return mBufferSize; }
-    virtual bool        isFastTrack() const = 0;
-    virtual bool        isDirect() const = 0;
-            bool        isOutputTrack() const { return (mType == TYPE_OUTPUT); }
-            bool        isPatchTrack() const { return (mType == TYPE_PATCH); }
-            bool        isExternalTrack() const { return !isOutputTrack() && !isPatchTrack(); }
-
-    virtual void        invalidate() {
+    bool isOutputTrack() const final { return (mType == TYPE_OUTPUT); }
+    bool isPatchTrack() const final { return (mType == TYPE_PATCH); }
+    bool isExternalTrack() const final { return !isOutputTrack() && !isPatchTrack(); }
+    void invalidate() override {
                             if (mIsInvalid) return;
                             mTrackMetrics.logInvalidate();
                             mIsInvalid = true;
                         }
-            bool        isInvalid() const { return mIsInvalid; }
+    bool isInvalid() const final { return mIsInvalid; }
+    void terminate() final { mTerminated = true; }
+    bool isTerminated() const final { return mTerminated; }
+    audio_attributes_t attributes() const final { return mAttr; }
+    bool isSpatialized() const override { return false; }
+    bool isBitPerfect() const override { return false; }
 
-            void        terminate() { mTerminated = true; }
-            bool        isTerminated() const { return mTerminated; }
+    // TODO(b/288339104) type
+    wp<Thread> thread() const final { return mThread; }
 
-    audio_attributes_t  attributes() const { return mAttr; }
-
-    virtual bool        isSpatialized() const { return false; }
-
-    virtual bool        isBitPerfect() const { return false; }
+    const sp<ServerProxy>& serverProxy() const final { return mServerProxy; }
 
 #ifdef TEE_SINK
-           void         dumpTee(int fd, const std::string &reason) const {
-                                mTee.dump(fd, reason);
-                        }
+    void dumpTee(int fd, const std::string &reason) const final {
+        mTee.dump(fd, reason);
+    }
 #endif
-
-            /** returns the buffer contents size converted to time in milliseconds
-             * for PCM Playback or Record streaming tracks. The return value is zero for
-             * PCM static tracks and not defined for non-PCM tracks.
-             *
-             * This may be called without the thread lock.
-             */
-    virtual double      bufferLatencyMs() const {
+    /** returns the buffer contents size converted to time in milliseconds
+     * for PCM Playback or Record streaming tracks. The return value is zero for
+     * PCM static tracks and not defined for non-PCM tracks.
+     *
+     * This may be called without the thread lock.
+     */
+    double bufferLatencyMs() const override {
                             return mServerProxy->framesReadySafe() * 1000. / sampleRate();
                         }
 
-            /** returns whether the track supports server latency computation.
-             * This is set in the constructor and constant throughout the track lifetime.
-             */
+    /** returns whether the track supports server latency computation.
+     * This is set in the constructor and constant throughout the track lifetime.
+     */
+    bool isServerLatencySupported() const final { return mServerLatencySupported; }
 
-            bool        isServerLatencySupported() const { return mServerLatencySupported; }
-
-            /** computes the server latency for PCM Playback or Record track
-             * to the device sink/source.  This is the time for the next frame in the track buffer
-             * written or read from the server thread to the device source or sink.
-             *
-             * This may be called without the thread lock, but latencyMs and fromTrack
-             * may be not be synchronized. For example PatchPanel may not obtain the
-             * thread lock before calling.
-             *
-             * \param latencyMs on success is set to the latency in milliseconds of the
-             *        next frame written/read by the server thread to/from the track buffer
-             *        from the device source/sink.
-             * \param fromTrack on success is set to true if latency was computed directly
-             *        from the track timestamp; otherwise set to false if latency was
-             *        estimated from the server timestamp.
-             *        fromTrack may be nullptr or omitted if not required.
-             *
-             * \returns OK or INVALID_OPERATION on failure.
-             */
-            status_t    getServerLatencyMs(double *latencyMs, bool *fromTrack = nullptr) const {
+    /** computes the server latency for PCM Playback or Record track
+     * to the device sink/source.  This is the time for the next frame in the track buffer
+     * written or read from the server thread to the device source or sink.
+     *
+     * This may be called without the thread lock, but latencyMs and fromTrack
+     * may be not be synchronized. For example PatchPanel may not obtain the
+     * thread lock before calling.
+     *
+     * \param latencyMs on success is set to the latency in milliseconds of the
+     *        next frame written/read by the server thread to/from the track buffer
+     *        from the device source/sink.
+     * \param fromTrack on success is set to true if latency was computed directly
+     *        from the track timestamp; otherwise set to false if latency was
+     *        estimated from the server timestamp.
+     *        fromTrack may be nullptr or omitted if not required.
+     *
+     * \returns OK or INVALID_OPERATION on failure.
+     */
+    status_t getServerLatencyMs(double* latencyMs, bool* fromTrack = nullptr) const final {
                             if (!isServerLatencySupported()) {
                                 return INVALID_OPERATION;
                             }
@@ -170,25 +131,25 @@
                             return OK;
                         }
 
-            /** computes the total client latency for PCM Playback or Record tracks
-             * for the next client app access to the device sink/source; i.e. the
-             * server latency plus the buffer latency.
-             *
-             * This may be called without the thread lock, but latencyMs and fromTrack
-             * may be not be synchronized. For example PatchPanel may not obtain the
-             * thread lock before calling.
-             *
-             * \param latencyMs on success is set to the latency in milliseconds of the
-             *        next frame written/read by the client app to/from the track buffer
-             *        from the device sink/source.
-             * \param fromTrack on success is set to true if latency was computed directly
-             *        from the track timestamp; otherwise set to false if latency was
-             *        estimated from the server timestamp.
-             *        fromTrack may be nullptr or omitted if not required.
-             *
-             * \returns OK or INVALID_OPERATION on failure.
-             */
-            status_t    getTrackLatencyMs(double *latencyMs, bool *fromTrack = nullptr) const {
+    /** computes the total client latency for PCM Playback or Record tracks
+     * for the next client app access to the device sink/source; i.e. the
+     * server latency plus the buffer latency.
+     *
+     * This may be called without the thread lock, but latencyMs and fromTrack
+     * may be not be synchronized. For example PatchPanel may not obtain the
+     * thread lock before calling.
+     *
+     * \param latencyMs on success is set to the latency in milliseconds of the
+     *        next frame written/read by the client app to/from the track buffer
+     *        from the device sink/source.
+     * \param fromTrack on success is set to true if latency was computed directly
+     *        from the track timestamp; otherwise set to false if latency was
+     *        estimated from the server timestamp.
+     *        fromTrack may be nullptr or omitted if not required.
+     *
+     * \returns OK or INVALID_OPERATION on failure.
+     */
+    status_t getTrackLatencyMs(double* latencyMs, bool* fromTrack = nullptr) const {
                             double serverLatencyMs;
                             status_t status = getServerLatencyMs(&serverLatencyMs, fromTrack);
                             if (status == OK) {
@@ -197,21 +158,15 @@
                             return status;
                         }
 
-           // TODO: Consider making this external.
-           struct FrameTime {
-               int64_t frames;
-               int64_t timeNs;
-           };
-
-           // KernelFrameTime is updated per "mix" period even for non-pcm tracks.
-           void         getKernelFrameTime(FrameTime *ft) const {
+    // KernelFrameTime is updated per "mix" period even for non-pcm tracks.
+    void getKernelFrameTime(FrameTime* ft) const final {
                            *ft = mKernelFrameTime.load();
                         }
 
-           audio_format_t format() const { return mFormat; }
-           int id() const { return mId; }
+    audio_format_t format() const final { return mFormat; }
+    int id() const final { return mId; }
 
-    const char *getTrackStateAsString() const {
+    const char* getTrackStateAsString() const final {
         if (isTerminated()) {
             return "TERMINATED";
         }
@@ -245,19 +200,19 @@
 
     // Called by the PlaybackThread to indicate that the track is becoming active
     // and a new interval should start with a given device list.
-    void logBeginInterval(const std::string& devices) {
+    void logBeginInterval(const std::string& devices) final {
         mTrackMetrics.logBeginInterval(devices);
     }
 
     // Called by the PlaybackThread to indicate the track is no longer active.
-    void logEndInterval() {
+    void logEndInterval() final {
         mTrackMetrics.logEndInterval();
     }
 
     // Called to tally underrun frames in playback.
-    virtual void tallyUnderrunFrames(size_t /* frames */) {}
+    void tallyUnderrunFrames(size_t /* frames */) override {}
 
-    audio_channel_mask_t channelMask() const { return mChannelMask; }
+    audio_channel_mask_t channelMask() const final { return mChannelMask; }
 
     /** @return true if the track has changed (metadata or volume) since
      *          the last time this function was called,
@@ -265,10 +220,10 @@
      *          false otherwise.
      *  Thread safe.
      */
-    bool readAndClearHasChanged() { return !mChangeNotified.test_and_set(); }
+    bool readAndClearHasChanged() final { return !mChangeNotified.test_and_set(); }
 
     /** Set that a metadata has changed and needs to be notified to backend. Thread safe. */
-    void setMetadataHasChanged() { mChangeNotified.clear(); }
+    void setMetadataHasChanged() final { mChangeNotified.clear(); }
 
 protected:
     DISALLOW_COPY_AND_ASSIGN(TrackBase);
@@ -285,31 +240,31 @@
     }
 
     // AudioBufferProvider interface
-    virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0;
-    virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
+    // status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) override;
+    void releaseBuffer(AudioBufferProvider::Buffer* buffer) override;
 
     // ExtendedAudioBufferProvider interface is only needed for Track,
     // but putting it in TrackBase avoids the complexity of virtual inheritance
-    virtual size_t  framesReady() const { return SIZE_MAX; }
+    size_t framesReady() const override { return SIZE_MAX; } // MmapTrack doesn't implement.
 
     uint32_t channelCount() const { return mChannelCount; }
 
-    size_t frameSize() const { return mFrameSize; }
+    size_t frameSize() const final { return mFrameSize; }
 
-    virtual uint32_t sampleRate() const { return mSampleRate; }
+    uint32_t sampleRate() const override { return mSampleRate; }
 
-    bool isStopped() const {
+    bool isStopped() const final {
         return (mState == STOPPED || mState == FLUSHED);
     }
 
     // for fast tracks and offloaded tracks only
-    bool isStopping() const {
+    bool isStopping() const final {
         return mState == STOPPING_1 || mState == STOPPING_2;
     }
-    bool isStopping_1() const {
+    bool isStopping_1() const final {
         return mState == STOPPING_1;
     }
-    bool isStopping_2() const {
+    bool isStopping_2() const final {
         return mState == STOPPING_2;
     }
 
@@ -421,7 +376,7 @@
 {
 public:
 
-    virtual ~PatchProxyBufferProvider() {}
+    virtual ~PatchProxyBufferProvider() = default;
 
     virtual bool        producesBufferOnDemand() const = 0;
     virtual status_t    obtainBuffer(Proxy::Buffer* buffer,
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 9ec3ee3..20033dd 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -94,7 +94,7 @@
             track_type type,
             audio_port_handle_t portId,
             std::string metricsId)
-    :   RefBase(),
+    :
         mThread(thread),
         mAllocType(alloc),
         mClient(client),
@@ -339,7 +339,7 @@
 
 class TrackHandle : public android::media::BnAudioTrack {
 public:
-    explicit TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track);
+    explicit TrackHandle(const sp<IAfTrack>& track);
     ~TrackHandle() override;
 
     binder::Status getCblk(std::optional<media::SharedFileRegion>* _aidl_return) final;
@@ -373,16 +373,15 @@
             const media::audio::common::AudioPlaybackRate& playbackRate) final;
 
 private:
-    const sp<AudioFlinger::PlaybackThread::Track> mTrack;
+    const sp<IAfTrack> mTrack;
 };
 
 /* static */
-sp<media::IAudioTrack> AudioFlinger::PlaybackThread::Track::createIAudioTrackAdapter(
-        const sp<Track>& track) {
+sp<media::IAudioTrack> IAfTrack::createIAudioTrackAdapter(const sp<IAfTrack>& track) {
     return sp<TrackHandle>::make(track);
 }
 
-TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
+TrackHandle::TrackHandle(const sp<IAfTrack>& track)
     : BnAudioTrack(),
       mTrack(track)
 {
@@ -852,7 +851,7 @@
     forEachTeePatchTrack([](auto patchTrack) { patchTrack->destroy(); });
 }
 
-void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
+void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) const
 {
     result.appendFormat("Type     Id Active Client Session Port Id S  Flags "
                         "  Format Chn mask  SRate "
@@ -863,7 +862,7 @@
                         isServerLatencySupported() ? "   Latency" : "");
 }
 
-void AudioFlinger::PlaybackThread::Track::appendDump(String8& result, bool active)
+void AudioFlinger::PlaybackThread::Track::appendDump(String8& result, bool active) const
 {
     char trackType;
     switch (mType) {
@@ -1460,7 +1459,7 @@
     return status;
 }
 
-sp<VolumeShaper::State> AudioFlinger::PlaybackThread::Track::getVolumeShaperState(int id)
+sp<VolumeShaper::State> AudioFlinger::PlaybackThread::Track::getVolumeShaperState(int id) const
 {
     // Note: We don't check if Thread exists.
 
@@ -1751,7 +1750,7 @@
 
 // implement VolumeBufferProvider interface
 
-gain_minifloat_packed_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
+gain_minifloat_packed_t AudioFlinger::PlaybackThread::Track::getVolumeLR() const
 {
     // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
     ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
@@ -1824,7 +1823,7 @@
     }
 }
 
-status_t AudioFlinger::PlaybackThread::Track::getDualMonoMode(audio_dual_mono_mode_t* mode)
+status_t AudioFlinger::PlaybackThread::Track::getDualMonoMode(audio_dual_mono_mode_t* mode) const
 {
     status_t status = INVALID_OPERATION;
     if (isOffloadedOrDirect()) {
@@ -1857,7 +1856,7 @@
     return status;
 }
 
-status_t AudioFlinger::PlaybackThread::Track::getAudioDescriptionMixLevel(float* leveldB)
+status_t AudioFlinger::PlaybackThread::Track::getAudioDescriptionMixLevel(float* leveldB) const
 {
     status_t status = INVALID_OPERATION;
     if (isOffloadedOrDirect()) {
@@ -1891,7 +1890,7 @@
 }
 
 status_t AudioFlinger::PlaybackThread::Track::getPlaybackRateParameters(
-        audio_playback_rate_t* playbackRate)
+        audio_playback_rate_t* playbackRate) const
 {
     status_t status = INVALID_OPERATION;
     if (isOffloadedOrDirect()) {
@@ -2483,7 +2482,7 @@
 
 class RecordHandle : public android::media::BnAudioRecord {
 public:
-    explicit RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack);
+    explicit RecordHandle(const sp<IAfRecordTrack>& recordTrack);
     ~RecordHandle() override;
     binder::Status start(int /*AudioSystem::sync_event_t*/ event,
             int /*audio_session_t*/ triggerSession) final;
@@ -2497,20 +2496,20 @@
             const std::string& sharedAudioPackageName, int64_t sharedAudioStartMs) final;
 
 private:
-    const sp<AudioFlinger::RecordThread::RecordTrack> mRecordTrack;
+    const sp<IAfRecordTrack> mRecordTrack;
 
     // for use from destructor
     void stop_nonvirtual();
 };
 
 /* static */
-sp<media::IAudioRecord> AudioFlinger::RecordThread::RecordTrack::createIAudioRecordAdapter(
-        const sp<RecordTrack>& recordTrack) {
+sp<media::IAudioRecord> IAfRecordTrack::createIAudioRecordAdapter(
+        const sp<IAfRecordTrack>& recordTrack) {
     return sp<RecordHandle>::make(recordTrack);
 }
 
 RecordHandle::RecordHandle(
-        const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
+        const sp<IAfRecordTrack>& recordTrack)
     : BnAudioRecord(),
     mRecordTrack(recordTrack)
 {
@@ -2752,7 +2751,7 @@
 }
 
 
-void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
+void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) const
 {
     result.appendFormat("Active     Id Client Session Port Id  S  Flags  "
                         " Format Chn mask  SRate Source  "
@@ -2760,7 +2759,7 @@
                         isServerLatencySupported() ? "   Latency" : "");
 }
 
-void AudioFlinger::RecordThread::RecordTrack::appendDump(String8& result, bool active)
+void AudioFlinger::RecordThread::RecordTrack::appendDump(String8& result, bool active) const
 {
     result.appendFormat("%c%5s %6d %6u %7u %7u  %2s 0x%03X "
             "%08X %08X %6u %6X "
@@ -2859,7 +2858,7 @@
 }
 
 status_t AudioFlinger::RecordThread::RecordTrack::getActiveMicrophones(
-        std::vector<media::MicrophoneInfoFw>* activeMicrophones)
+        std::vector<media::MicrophoneInfoFw>* activeMicrophones) const
 {
     sp<ThreadBase> thread = mThread.promote();
     if (thread != 0) {
@@ -3331,13 +3330,13 @@
     }
 }
 
-void AudioFlinger::MmapThread::MmapTrack::appendDumpHeader(String8& result)
+void AudioFlinger::MmapThread::MmapTrack::appendDumpHeader(String8& result) const
 {
     result.appendFormat("Client Session Port Id  Format Chn mask  SRate Flags %s\n",
                         isOut() ? "Usg CT": "Source");
 }
 
-void AudioFlinger::MmapThread::MmapTrack::appendDump(String8& result, bool active __unused)
+void AudioFlinger::MmapThread::MmapTrack::appendDump(String8& result, bool active __unused) const
 {
     result.appendFormat("%6u %7u %7u %08X %08X %6u 0x%03X ",
             mPid,
diff --git a/services/audioflinger/fastpath/FastMixerState.h b/services/audioflinger/fastpath/FastMixerState.h
index c70e42a..8ab6d25 100644
--- a/services/audioflinger/fastpath/FastMixerState.h
+++ b/services/audioflinger/fastpath/FastMixerState.h
@@ -35,7 +35,7 @@
 class VolumeProvider {
 public:
     // The provider implementation is responsible for validating that the return value is in range.
-    virtual gain_minifloat_packed_t getVolumeLR() = 0;
+    virtual gain_minifloat_packed_t getVolumeLR() const = 0;
 protected:
     VolumeProvider() = default;
     virtual ~VolumeProvider() = default;