diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index fc2431b..b99e33f 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -2391,7 +2391,7 @@
     CreateRecordInput input = VALUE_OR_RETURN_STATUS(CreateRecordInput::fromAidl(_input));
     CreateRecordOutput output;
 
-    sp<RecordThread::RecordTrack> recordTrack;
+    sp<IAfRecordTrack> recordTrack;
     sp<Client> client;
     status_t lStatus;
     audio_session_t sessionId = input.sessionId;
@@ -3946,7 +3946,7 @@
             // use an index mask here to create the PatchRecord.
             inChannelMask = audio_channel_mask_out_to_in_index_mask(track->channelMask());
         }
-        sp patchRecord = new RecordThread::PatchRecord(nullptr /* thread */,
+        sp<IAfPatchRecord> patchRecord = IAfPatchRecord::create(nullptr /* thread */,
                                                        track->sampleRate(),
                                                        inChannelMask,
                                                        track->format(),
@@ -3966,7 +3966,7 @@
         // for now, we exclude fast tracks by removing the Fast flag.
         const audio_output_flags_t outputFlags =
                 (audio_output_flags_t)(track->getOutputFlags() & ~AUDIO_OUTPUT_FLAG_FAST);
-        sp patchTrack = new PlaybackThread::PatchTrack(secondaryThread,
+        sp<IAfPatchTrack> patchTrack = IAfPatchTrack::create(secondaryThread,
                                                        track->streamType(),
                                                        track->sampleRate(),
                                                        track->channelMask(),
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 54e5cbc..ebff932 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -119,6 +119,7 @@
 #include "android/media/BnEffect.h"
 
 #include "Client.h"
+#include "ResamplerBufferProvider.h"
 
 // include AudioFlinger component interfaces
 #include "IAfEffect.h"
@@ -136,7 +137,6 @@
 class FastMixer;
 class IAudioManager;
 class PassthruBufferProvider;
-class RecordBufferConverter;
 class ServerProxy;
 
 // ----------------------------------------------------------------------------
@@ -571,8 +571,6 @@
     class AsyncCallbackThread;
     class BitPerfectThread;
 private:
-    class Track;
-    class RecordTrack;
     class DeviceEffectManager;
     // TODO(b/288339104) these should be separate files
 public:
@@ -581,8 +579,9 @@
 private:
     struct AudioStreamIn;
     struct TeePatch;
+public:
     using TeePatches = std::vector<TeePatch>;
-
+private:
 
     struct  stream_type_t {
         stream_type_t()
@@ -705,9 +704,12 @@
                                      PlaybackThread *srcThread,
                                      PlaybackThread *dstThread);
 
+public:
+    // TODO(b/288339104) cluster together
               status_t moveAuxEffectToIo(int EffectId,
                                          const sp<PlaybackThread>& dstThread,
                                          sp<PlaybackThread> *srcThread);
+private:
 
               // return thread associated with primary hardware device, or NULL
               PlaybackThread *primaryPlaybackThread_l() const;
@@ -728,7 +730,10 @@
 
                 void        removeClient_l(pid_t pid);
                 void        removeNotificationClient(pid_t pid);
+public:
+    // TODO(b/288339104) cluster together
                 bool isNonOffloadableGlobalEffectEnabled_l();
+private:
                 void onNonOffloadableGlobalEffectEnable();
                 bool isSessionAcquired_l(audio_session_t audioSession);
 
@@ -783,8 +788,8 @@
     };
 
     struct TeePatch {
-        sp<RecordThread::PatchRecord> patchRecord;
-        sp<PlaybackThread::PatchTrack> patchTrack;
+        sp<IAfPatchRecord> patchRecord;
+        sp<IAfPatchTrack> patchTrack;
     };
 
     // for mAudioSessionRefs only
@@ -797,11 +802,13 @@
         int         mCnt;
     };
 
+public:
+    // TODO(b/288339104) access by getter,
     mutable     Mutex                               mLock;
                 // protects mClients and mNotificationClients.
                 // must be locked after mLock and ThreadBase::mLock if both must be locked
                 // avoids acquiring AudioFlinger::mLock from inside thread loop.
-public:
+
     // TODO(b/288339104) access by getter,
     mutable     Mutex                               mClientLock;
 private:
@@ -910,9 +917,9 @@
                                       size_t rejectedKVPSize, const String8& rejectedKVPs,
                                       uid_t callingUid);
 
+public:
     sp<IAudioManager> getOrCreateAudioManager();
 
-public:
     // These methods read variables atomically without mLock,
     // though the variables are updated with mLock.
     bool    isLowRamDevice() const { return mIsLowRamDevice; }
diff --git a/services/audioflinger/IAfTrack.h b/services/audioflinger/IAfTrack.h
index bd2228b..541be0a 100644
--- a/services/audioflinger/IAfTrack.h
+++ b/services/audioflinger/IAfTrack.h
@@ -223,6 +223,9 @@
 // Common interface for Playback tracks.
 class IAfTrack : public virtual IAfTrackBase {
 public:
+    // FillingStatus is used for suppressing volume ramp at begin of playing
+    enum FillingStatus { FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE };
+
     // 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).
@@ -230,6 +233,31 @@
     // Only one AIDL IAudioTrack interface adapter should be created per Track.
     static sp<media::IAudioTrack> createIAudioTrackAdapter(const sp<IAfTrack>& track);
 
+    static sp<IAfTrack> create( // TODO(b/288339104) void*
+            void* /* AudioFlinger::PlaybackThread */ thread,
+            const sp<Client>& client,
+            audio_stream_type_t streamType,
+            const audio_attributes_t& attr,
+            uint32_t sampleRate,
+            audio_format_t format,
+            audio_channel_mask_t channelMask,
+            size_t frameCount,
+            void* buffer,
+            size_t bufferSize,
+            const sp<IMemory>& sharedBuffer,
+            audio_session_t sessionId,
+            pid_t creatorPid,
+            const AttributionSourceState& attributionSource,
+            audio_output_flags_t flags,
+            track_type type,
+            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE,
+            /** default behaviour is to start when there are as many frames
+              * ready as possible (aka. Buffer is full). */
+            size_t frameCountToBeReady = SIZE_MAX,
+            float speed = 1.0f,
+            bool isSpatialized = false,
+            bool isBitPerfect = false);
+
     virtual void pause() = 0;
     virtual void flush() = 0;
     virtual audio_stream_type_t streamType() const = 0;
@@ -316,11 +344,66 @@
     virtual void disable() = 0;
     virtual int& fastIndex() = 0;
     virtual bool isPlaybackRestricted() const = 0;
+
+    // Used by thread only
+
+    virtual bool isPausing() const = 0;
+    virtual bool isPaused() const = 0;
+    virtual bool isResuming() const = 0;
+    virtual bool isReady() const = 0;
+    virtual void setPaused() = 0;
+    virtual void reset() = 0;
+    virtual bool isFlushPending() const = 0;
+    virtual void flushAck() = 0;
+    virtual bool isResumePending() const = 0;
+    virtual void resumeAck() = 0;
+    // For direct or offloaded tracks ensure that the pause state is acknowledged
+    // by the playback thread in case of an immediate flush.
+    virtual bool isPausePending() const = 0;
+    virtual void pauseAck() = 0;
+    virtual void updateTrackFrameInfo(
+            int64_t trackFramesReleased, int64_t sinkFramesWritten, uint32_t halSampleRate,
+            const ExtendedTimestamp& timeStamp) = 0;
+    virtual sp<IMemory> sharedBuffer() const = 0;
+
+    // Dup with ExtendedAudioBufferProvider
+    virtual size_t framesReady() const = 0;
+
+    // presentationComplete checked by frames. (Mixed Tracks).
+    // framesWritten is cumulative, never reset, and is shared all tracks
+    // audioHalFrames is derived from output latency
+    virtual bool presentationComplete(int64_t framesWritten, size_t audioHalFrames) = 0;
+
+    // presentationComplete checked by time. (Direct Tracks).
+    virtual bool presentationComplete(uint32_t latencyMs) = 0;
+
+    virtual void resetPresentationComplete() = 0;
+
+    virtual bool hasVolumeController() const = 0;
+    virtual void setHasVolumeController(bool hasVolumeController) = 0;
+    virtual const sp<AudioTrackServerProxy>& audioTrackServerProxy() const = 0;
+    virtual void setCachedVolume(float volume) = 0;
+    virtual void setResetDone(bool resetDone) = 0;
+
+    virtual ExtendedAudioBufferProvider* asExtendedAudioBufferProvider() = 0;
+    virtual VolumeProvider* asVolumeProvider() = 0;
+
+    // TODO(b/288339104) split into getter/setter
+    virtual FillingStatus& fillingStatus() = 0;
+    virtual int8_t& retryCount() = 0;
+    virtual FastTrackUnderruns& fastTrackUnderruns() = 0;
 };
 
 // playback track, used by DuplicatingThread
 class IAfOutputTrack : public virtual IAfTrack {
 public:
+    // TODO(b/288339104) void*
+    static sp<IAfOutputTrack> create(
+            void* /* AudioFlinger::PlaybackThread */ playbackThread,
+            void* /* AudioFlinger::DuplicatingThread */ sourceThread, uint32_t sampleRate,
+            audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount,
+            const AttributionSourceState& attributionSource);
+
     virtual ssize_t write(void* data, uint32_t frames) = 0;
     virtual bool bufferQueueEmpty() const = 0;
     virtual bool isActive() const = 0;
@@ -333,6 +416,18 @@
 
 class IAfMmapTrack : public virtual IAfTrackBase {
 public:
+    // TODO(b/288339104) void*
+    static sp<IAfMmapTrack> create(void* /*AudioFlinger::ThreadBase */ thread,
+            const audio_attributes_t& attr,
+            uint32_t sampleRate,
+            audio_format_t format,
+            audio_channel_mask_t channelMask,
+            audio_session_t sessionId,
+            bool isOut,
+            const android::content::AttributionSourceState& attributionSource,
+            pid_t creatorPid,
+            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
+
     // protected by MMapThread::mLock
     virtual void setSilenced_l(bool silenced) = 0;
     // protected by MMapThread::mLock
@@ -348,6 +443,8 @@
             const sp<IAudioManager>& audioManager, mute_state_t muteState) = 0;
 };
 
+class RecordBufferConverter;
+
 class IAfRecordTrack : public virtual IAfTrackBase {
 public:
     // createIAudioRecordAdapter() is a static constructor which creates an
@@ -357,6 +454,24 @@
     // Only one AIDL IAudioRecord interface adapter should be created per RecordTrack.
     static sp<media::IAudioRecord> createIAudioRecordAdapter(const sp<IAfRecordTrack>& recordTrack);
 
+    // TODO(b/288339104) void*
+    static sp<IAfRecordTrack> create(void* /* AudioFlinger::RecordThread */ thread,
+            const sp<Client>& client,
+            const audio_attributes_t& attr,
+            uint32_t sampleRate,
+            audio_format_t format,
+            audio_channel_mask_t channelMask,
+            size_t frameCount,
+            void* buffer,
+            size_t bufferSize,
+            audio_session_t sessionId,
+            pid_t creatorPid,
+            const AttributionSourceState& attributionSource,
+            audio_input_flags_t flags,
+            track_type type,
+            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE,
+            int32_t startFrames = -1);
+
     // clear the buffer overflow flag
     virtual void clearOverflow() = 0;
     // set the buffer overflow flag and return previous value
@@ -386,6 +501,12 @@
     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
+
+    // private to Threads
+    virtual AudioBufferProvider::Buffer& sinkBuffer() = 0;
+    virtual audioflinger::SynchronizedRecordState& synchronizedRecordState() = 0;
+    virtual RecordBufferConverter* recordBufferConverter() const = 0;
+    virtual ResamplerBufferProvider* resamplerBufferProvider() const = 0;
 };
 
 // PatchProxyBufferProvider interface is implemented by PatchTrack and PatchRecord.
@@ -401,13 +522,32 @@
 
 class IAfPatchTrackBase : public virtual RefBase {
 public:
+    using Timeout = std::optional<std::chrono::nanoseconds>;
+
     virtual void setPeerTimeout(std::chrono::nanoseconds timeout) = 0;
     virtual void setPeerProxy(const sp<IAfPatchTrackBase>& proxy, bool holdReference) = 0;
     virtual void clearPeerProxy() = 0;
     virtual PatchProxyBufferProvider* asPatchProxyBufferProvider() = 0;
 };
 
-class IAfPatchTrack : public virtual IAfTrack, public virtual IAfPatchTrackBase {};
+class IAfPatchTrack : public virtual IAfTrack, public virtual IAfPatchTrackBase {
+public:
+    static sp<IAfPatchTrack> create(
+            void * /* PlaybackThread */ playbackThread, // TODO(b/288339104)
+            audio_stream_type_t streamType,
+            uint32_t sampleRate,
+            audio_channel_mask_t channelMask,
+            audio_format_t format,
+            size_t frameCount,
+            void *buffer,
+            size_t bufferSize,
+            audio_output_flags_t flags,
+            const Timeout& timeout = {},
+            size_t frameCountToBeReady = 1 /** Default behaviour is to start
+                                             *  as soon as possible to have
+                                             *  the lowest possible latency
+                                             *  even if it might glitch. */);
+};
 
 // Abstraction for the Audio Source for the RecordThread (HAL or PassthruPatchRecord).
 struct Source {
@@ -420,6 +560,27 @@
 
 class IAfPatchRecord : public virtual IAfRecordTrack, public virtual IAfPatchTrackBase {
 public:
+    static sp<IAfPatchRecord> create(
+            void* /* RecordThread */ recordThread, // TODO(b/288339104)
+            uint32_t sampleRate,
+            audio_channel_mask_t channelMask,
+            audio_format_t format,
+            size_t frameCount,
+            void* buffer,
+            size_t bufferSize,
+            audio_input_flags_t flags,
+            const Timeout& timeout = {},
+            audio_source_t source = AUDIO_SOURCE_DEFAULT);
+
+    static sp<IAfPatchRecord> createPassThru(
+            void* /* RecordThread */ recordThread, // TODO(b/288339104)
+            uint32_t sampleRate,
+            audio_channel_mask_t channelMask,
+            audio_format_t format,
+            size_t frameCount,
+            audio_input_flags_t flags,
+            audio_source_t source = AUDIO_SOURCE_DEFAULT);
+
     virtual Source* getSource() = 0;
     virtual size_t writeFrames(const void* src, size_t frameCount, size_t frameSize) = 0;
 };
diff --git a/services/audioflinger/MmapTracks.h b/services/audioflinger/MmapTracks.h
index 0ab1adb..081af74 100644
--- a/services/audioflinger/MmapTracks.h
+++ b/services/audioflinger/MmapTracks.h
@@ -15,14 +15,14 @@
 ** limitations under the License.
 */
 
-#ifndef INCLUDING_FROM_AUDIOFLINGER_H
-    #error This header file should only be included from AudioFlinger.h
-#endif
+#pragma once
+
+namespace android {
 
 // playback track
 class MmapTrack : public TrackBase, public IAfMmapTrack {
 public:
-                MmapTrack(ThreadBase *thread,
+                MmapTrack(AudioFlinger::ThreadBase* thread,
                             const audio_attributes_t& attr,
                             uint32_t sampleRate,
                             audio_format_t format,
@@ -87,3 +87,4 @@
             GUARDED_BY(AudioFlinger::MmapPlaybackThread::mLock);
 };  // end of Track
 
+} // namespace android
\ No newline at end of file
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index 7063ff0..cb74292 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -546,7 +546,7 @@
         outputFlags = (audio_output_flags_t) (outputFlags & ~AUDIO_OUTPUT_FLAG_FAST);
     }
 
-    sp<RecordThread::PatchRecord> tempRecordTrack;
+    sp<IAfPatchRecord> tempRecordTrack;
     const bool usePassthruPatchRecord =
             (inputFlags & AUDIO_INPUT_FLAG_DIRECT) && (outputFlags & AUDIO_OUTPUT_FLAG_DIRECT);
     const size_t playbackFrameCount = mPlayback.thread()->frameCount();
@@ -558,7 +558,7 @@
         frameCount = std::max(playbackFrameCount, recordFrameCount);
         ALOGV("%s() playframeCount %zu recordFrameCount %zu frameCount %zu",
             __func__, playbackFrameCount, recordFrameCount, frameCount);
-        tempRecordTrack = new RecordThread::PassthruPatchRecord(
+        tempRecordTrack = IAfPatchRecord::createPassThru(
                                                  mRecord.thread().get(),
                                                  sampleRate,
                                                  inChannelMask,
@@ -577,7 +577,7 @@
         ALOGV("%s() playframeCount %zu recordFrameCount %zu frameCount %zu",
             __func__, playbackFrameCount, recordFrameCount, frameCount);
 
-        tempRecordTrack = new RecordThread::PatchRecord(
+        tempRecordTrack = IAfPatchRecord::create(
                                                  mRecord.thread().get(),
                                                  sampleRate,
                                                  inChannelMask,
@@ -602,7 +602,7 @@
     // Disable this behavior for FM Tuner source if no fast capture/mixer available.
     const bool isFmBridge = mAudioPatch.sources[0].ext.device.type == AUDIO_DEVICE_IN_FM_TUNER;
     const size_t frameCountToBeReady = isFmBridge && !usePassthruPatchRecord ? frameCount / 4 : 1;
-    sp<PlaybackThread::PatchTrack> tempPatchTrack = new PlaybackThread::PatchTrack(
+    sp<IAfPatchTrack> tempPatchTrack = IAfPatchTrack::create(
                                            mPlayback.thread().get(),
                                            streamType,
                                            sampleRate,
diff --git a/services/audioflinger/PatchPanel.h b/services/audioflinger/PatchPanel.h
index 2c5e47c..e693486 100644
--- a/services/audioflinger/PatchPanel.h
+++ b/services/audioflinger/PatchPanel.h
@@ -216,9 +216,9 @@
         // the objects are created by createConnections() and released by clearConnections()
         // playback thread is created if no existing playback thread can be used
         // connects playback thread output to sink device
-        Endpoint<PlaybackThread, PlaybackThread::PatchTrack> mPlayback;
+        Endpoint<PlaybackThread, IAfPatchTrack> mPlayback;
         // connects source device to record thread input
-        Endpoint<RecordThread, RecordThread::PatchRecord> mRecord;
+        Endpoint<RecordThread, IAfPatchRecord> mRecord;
 
         wp<ThreadBase> mThread;
         bool mIsEndpointPatch;
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 5467629..c549f3f 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -15,13 +15,12 @@
 ** limitations under the License.
 */
 
-#ifndef INCLUDING_FROM_AUDIOFLINGER_H
-    #error This header file should only be included from AudioFlinger.h
-#endif
-
+#pragma once
 #include <math.h>
 #include <sys/types.h>
 
+namespace android {
+
 // Checks and monitors OP_PLAY_AUDIO
 class OpPlayAudioMonitor : public RefBase {
     friend class sp<OpPlayAudioMonitor>;
@@ -69,7 +68,7 @@
 // playback track
 class Track : public TrackBase, public virtual IAfTrack, public VolumeProvider {
 public:
-                        Track(  PlaybackThread *thread,
+                        Track(AudioFlinger::PlaybackThread* thread,
                                 const sp<Client>& client,
                                 audio_stream_type_t streamType,
                                 const audio_attributes_t& attr,
@@ -191,9 +190,10 @@
             // This function should be called with holding thread lock.
     void updateTeePatches_l() final;
     void setTeePatchesToUpdate_l(const void* teePatchesToUpdate) final {
-        setTeePatchesToUpdate_l(*reinterpret_cast<const TeePatches*>(teePatchesToUpdate));
+        setTeePatchesToUpdate_l(  // TODO(b/288339104) void*
+                *reinterpret_cast<const AudioFlinger::TeePatches*>(teePatchesToUpdate));
     }
-    void setTeePatchesToUpdate_l(TeePatches teePatchesToUpdate);
+    void setTeePatchesToUpdate_l(AudioFlinger::TeePatches teePatchesToUpdate);
 
     void tallyUnderrunFrames(size_t frames) final {
        if (isOut()) { // we expect this from output tracks only
@@ -235,34 +235,35 @@
     int64_t framesReleased() const override;
     void onTimestamp(const ExtendedTimestamp &timestamp) override;
 
-    bool isPausing() const { return mState == PAUSING; }
-    bool isPaused() const { return mState == PAUSED; }
-    bool isResuming() const { return mState == RESUMING; }
-    bool isReady() const;
-    void setPaused() { mState = PAUSED; }
-    void reset();
-    bool isFlushPending() const { return mFlushHwPending; }
-    void flushAck();
-    bool isResumePending();
-    void resumeAck();
+    // Used by thread
+    bool isPausing() const final { return mState == PAUSING; }
+    bool isPaused() const final { return mState == PAUSED; }
+    bool isResuming() const final { return mState == RESUMING; }
+    bool isReady() const final;
+    void setPaused() final { mState = PAUSED; }
+    void reset() final;
+    bool isFlushPending() const final { return mFlushHwPending; }
+    void flushAck() final;
+    bool isResumePending() const final;
+    void resumeAck() final;
     // For direct or offloaded tracks ensure that the pause state is acknowledged
     // by the playback thread in case of an immediate flush.
-    bool isPausePending() const { return mPauseHwPending; }
-    void pauseAck();
+    bool isPausePending() const final { return mPauseHwPending; }
+    void pauseAck() final;
     void updateTrackFrameInfo(int64_t trackFramesReleased, int64_t sinkFramesWritten,
-            uint32_t halSampleRate, const ExtendedTimestamp &timeStamp);
+            uint32_t halSampleRate, const ExtendedTimestamp& timeStamp) final;
 
-    sp<IMemory> sharedBuffer() const { return mSharedBuffer; }
+    sp<IMemory> sharedBuffer() const final { return mSharedBuffer; }
 
     // presentationComplete checked by frames. (Mixed Tracks).
     // framesWritten is cumulative, never reset, and is shared all tracks
     // audioHalFrames is derived from output latency
-    bool presentationComplete(int64_t framesWritten, size_t audioHalFrames);
+    bool presentationComplete(int64_t framesWritten, size_t audioHalFrames) final;
 
     // presentationComplete checked by time. (Direct Tracks).
-    bool presentationComplete(uint32_t latencyMs);
+    bool presentationComplete(uint32_t latencyMs) final;
 
-    void resetPresentationComplete() {
+    void resetPresentationComplete() final {
         mPresentationCompleteFrames = 0;
         mPresentationCompleteTimeNs = 0;
     }
@@ -273,7 +274,6 @@
 
     void signalClientFlag(int32_t flag);
 
-public:
     void triggerEvents(AudioSystem::sync_event_t type) final;
     void invalidate() final;
     void disable() final;
@@ -282,11 +282,32 @@
         // The monitor is only created for tracks that can be silenced.
         return mOpPlayAudioMonitor ? !mOpPlayAudioMonitor->hasOpPlayAudio() : false; }
 
-protected:
+    const sp<AudioTrackServerProxy>& audioTrackServerProxy() const final {
+        return mAudioTrackServerProxy;
+    }
+    bool hasVolumeController() const final { return mHasVolumeController; }
+    void setHasVolumeController(bool hasVolumeController) final {
+        mHasVolumeController = hasVolumeController;
+    }
+    void setCachedVolume(float volume) final {
+        mCachedVolume = volume;
+    }
+    void setResetDone(bool resetDone) final {
+        mResetDone = resetDone;
+    }
+    ExtendedAudioBufferProvider* asExtendedAudioBufferProvider() final {
+        return this;
+    }
+    VolumeProvider* asVolumeProvider() final {
+        return this;
+    }
 
-    // FILLED state is used for suppressing volume ramp at begin of playing
-    enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE};
-    mutable uint8_t     mFillingUpStatus;
+    FillingStatus& fillingStatus() final { return mFillingStatus; }
+    int8_t& retryCount() final { return mRetryCount; }
+    FastTrackUnderruns& fastTrackUnderruns() final { return mObservedUnderruns; }
+
+protected:
+    mutable FillingStatus mFillingStatus;
     int8_t              mRetryCount;
 
     // see comment at AudioFlinger::PlaybackThread::Track::~Track for why this can't be const
@@ -368,8 +389,8 @@
     bool                mFlushHwPending; // track requests for thread flush
     bool                mPauseHwPending = false; // direct/offload track request for thread pause
     audio_output_flags_t mFlags;
-    TeePatches  mTeePatches;
-    std::optional<TeePatches> mTeePatchesToUpdate;
+    AudioFlinger::TeePatches  mTeePatches;
+    std::optional<AudioFlinger::TeePatches> mTeePatchesToUpdate;
     const float         mSpeed;
     const bool          mIsSpatialized;
     const bool          mIsBitPerfect;
@@ -390,8 +411,8 @@
         void *mBuffer;
     };
 
-                        OutputTrack(PlaybackThread *thread,
-                                DuplicatingThread *sourceThread,
+                        OutputTrack(AudioFlinger::PlaybackThread* thread,
+                                AudioFlinger::DuplicatingThread* sourceThread,
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
@@ -436,7 +457,7 @@
     Vector < Buffer* >          mBufferQueue;
     AudioBufferProvider::Buffer mOutBuffer;
     bool                        mActive;
-    DuplicatingThread* const    mSourceThread; // for waitTimeMs() in write()
+    AudioFlinger::DuplicatingThread* const mSourceThread; // for waitTimeMs() in write()
     sp<AudioTrackClientProxy>   mClientProxy;
 
     /** Attributes of the source tracks.
@@ -458,8 +479,7 @@
 // playback track, used by PatchPanel
 class PatchTrack : public Track, public PatchTrackBase, public IAfPatchTrack {
 public:
-
-                        PatchTrack(PlaybackThread *playbackThread,
+                        PatchTrack(AudioFlinger::PlaybackThread* playbackThread,
                                    audio_stream_type_t streamType,
                                    uint32_t sampleRate,
                                    audio_channel_mask_t channelMask,
@@ -492,3 +512,5 @@
 private:
             void restartIfDisabled();
 };  // end of PatchTrack
+
+} // namespace android
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index ad627d7..9d25ba4 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -15,16 +15,16 @@
 ** limitations under the License.
 */
 
+#pragma once
+
 #include <android/content/AttributionSourceState.h>
 
-#ifndef INCLUDING_FROM_AUDIOFLINGER_H
-    #error This header file should only be included from AudioFlinger.h
-#endif
+namespace android {
 
 // record track
 class RecordTrack : public TrackBase, public virtual IAfRecordTrack {
 public:
-                        RecordTrack(RecordThread *thread,
+                        RecordTrack(AudioFlinger::RecordThread* thread,
                                 const sp<Client>& client,
                                 const audio_attributes_t& attr,
                                 uint32_t sampleRate,
@@ -85,15 +85,27 @@
     using MetadataInserter = std::back_insert_iterator<SinkMetadatas>;
     void copyMetadataTo(MetadataInserter& backInserter) const final;
 
+    AudioBufferProvider::Buffer& sinkBuffer() final { return mSink; }
+    audioflinger::SynchronizedRecordState& synchronizedRecordState() final {
+        return mSynchronizedRecordState;
+    }
+    RecordBufferConverter* recordBufferConverter() const final { return mRecordBufferConverter; }
+    ResamplerBufferProvider* resamplerBufferProvider() const final {
+        return mResamplerBufferProvider;
+    }
+
 private:
     friend class AudioFlinger;  // for mState
 
     DISALLOW_COPY_AND_ASSIGN(RecordTrack);
 
+protected:
     // AudioBufferProvider interface
-    virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
+    status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) override;
     // releaseBuffer() not overridden
 
+private:
+
     bool                mOverflow;  // overflow on most recent attempt to fill client buffer
 
             AudioBufferProvider::Buffer mSink;  // references client's buffer sink in shared memory
@@ -106,7 +118,7 @@
                     mSynchronizedRecordState{mSampleRate}; // sampleRate defined in base
 
             // used by resampler to find source frames
-            ResamplerBufferProvider            *mResamplerBufferProvider;
+            ResamplerBufferProvider* mResamplerBufferProvider;
 
             // used by the record thread to convert frames to proper destination format
             RecordBufferConverter              *mRecordBufferConverter;
@@ -121,8 +133,7 @@
 // playback track, used by PatchPanel
 class PatchRecord : public RecordTrack, public PatchTrackBase, public IAfPatchRecord {
 public:
-
-    PatchRecord(RecordThread *recordThread,
+    PatchRecord(AudioFlinger::RecordThread* recordThread,
                 uint32_t sampleRate,
                 audio_channel_mask_t channelMask,
                 audio_format_t format,
@@ -158,7 +169,7 @@
 
 class PassthruPatchRecord : public PatchRecord, public Source {
 public:
-    PassthruPatchRecord(RecordThread *recordThread,
+    PassthruPatchRecord(AudioFlinger::RecordThread* recordThread,
                         uint32_t sampleRate,
                         audio_channel_mask_t channelMask,
                         audio_format_t format,
@@ -201,7 +212,7 @@
         PassthruPatchRecord& mPassthru;
     };
 
-    sp<StreamInHalInterface> obtainStream(sp<ThreadBase>* thread);
+    sp<StreamInHalInterface> obtainStream(sp<AudioFlinger::ThreadBase>* thread);
 
     PatchRecordAudioBufferProvider mPatchRecordAudioBufferProvider;
     std::unique_ptr<void, decltype(free)*> mSinkBuffer;  // frame size aligned continuous buffer
@@ -213,3 +224,5 @@
     status_t mReadError = NO_ERROR; // GUARDED_BY(mReadLock)
     int64_t mLastReadFrames = 0;  // accessed on RecordThread only
 };
+
+} // namespace android
diff --git a/services/audioflinger/ResamplerBufferProvider.h b/services/audioflinger/ResamplerBufferProvider.h
new file mode 100644
index 0000000..b697743
--- /dev/null
+++ b/services/audioflinger/ResamplerBufferProvider.h
@@ -0,0 +1,66 @@
+/*
+ * 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 {
+
+class IAfRecordTrack;
+
+/* The ResamplerBufferProvider is used to retrieve recorded input data from the
+ * RecordThread.  It maintains local state on the relative position of the read
+ * position of the RecordTrack compared with the RecordThread.
+ */
+class ResamplerBufferProvider : public AudioBufferProvider
+{
+public:
+    explicit ResamplerBufferProvider(IAfRecordTrack* recordTrack) :
+        mRecordTrack(recordTrack) {}
+
+    // called to set the ResamplerBufferProvider to head of the RecordThread data buffer,
+    // skipping any previous data read from the hal.
+    void reset();
+
+    /* Synchronizes RecordTrack position with the RecordThread.
+     * Calculates available frames and handle overruns if the RecordThread
+     * has advanced faster than the ResamplerBufferProvider has retrieved data.
+     * TODO: why not do this for every getNextBuffer?
+     *
+     * Parameters
+     * framesAvailable:  pointer to optional output size_t to store record track
+     *                   frames available.
+     *      hasOverrun:  pointer to optional boolean, returns true if track has overrun.
+     */
+
+    void sync(size_t* framesAvailable = nullptr, bool* hasOverrun = nullptr);
+
+    // AudioBufferProvider interface
+    status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) final;
+    void releaseBuffer(AudioBufferProvider::Buffer* buffer) final;
+
+    int32_t getFront() const { return mRsmpInFront; }
+    void setFront(int32_t front) { mRsmpInFront = front; }
+
+private:
+    IAfRecordTrack* const mRecordTrack;
+    size_t mRsmpInUnrel = 0;   // unreleased frames remaining from
+                               // most recent getNextBuffer
+                               // for debug only
+    int32_t mRsmpInFront = 0;  // next available frame
+                               // rolling counter that is never cleared
+};
+
+} // namespace android
\ No newline at end of file
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 07f0373..8c68751 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -2208,7 +2208,7 @@
         result.append(prefix);
         mTracks[0]->appendDumpHeader(result);
         for (size_t i = 0; i < numtracks; ++i) {
-            sp<Track> track = mTracks[i];
+            sp<IAfTrack> track = mTracks[i];
             if (track != 0) {
                 bool active = mActiveTracks.indexOf(track) >= 0;
                 if (active) {
@@ -2228,7 +2228,7 @@
         result.append(prefix);
         mActiveTracks[0]->appendDumpHeader(result);
         for (size_t i = 0; i < numactive; ++i) {
-            sp<Track> track = mActiveTracks[i];
+            sp<IAfTrack> track = mActiveTracks[i];
             if (mTracks.indexOf(track) < 0) {
                 result.append(prefix);
                 track->appendDump(result, true /* active */);
@@ -2275,7 +2275,7 @@
 }
 
 // PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
-sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
+sp<IAfTrack> AudioFlinger::PlaybackThread::createTrack_l(
         const sp<Client>& client,
         audio_stream_type_t streamType,
         const audio_attributes_t& attr,
@@ -2300,7 +2300,7 @@
 {
     size_t frameCount = *pFrameCount;
     size_t notificationFrameCount = *pNotificationFrameCount;
-    sp<Track> track;
+    sp<IAfTrack> track;
     status_t lStatus;
     audio_output_flags_t outputFlags = mOutput->flags;
     audio_output_flags_t requestedFlags = *flags;
@@ -2593,7 +2593,7 @@
         // manager
         product_strategy_t strategy = getStrategyForStream(streamType);
         for (size_t i = 0; i < mTracks.size(); ++i) {
-            sp<Track> t = mTracks[i];
+            sp<IAfTrack> t = mTracks[i];
             if (t != 0 && t->isExternalTrack()) {
                 product_strategy_t actual = getStrategyForStream(t->streamType());
                 if (sessionId == t->sessionId() && strategy != actual) {
@@ -2615,11 +2615,11 @@
             trackFlags = static_cast<audio_output_flags_t>(trackFlags | AUDIO_OUTPUT_FLAG_DIRECT);
         }
 
-        track = new Track(this, client, streamType, attr, sampleRate, format,
+        track = IAfTrack::create(this, client, streamType, attr, sampleRate, format,
                           channelMask, frameCount,
                           nullptr /* buffer */, (size_t)0 /* bufferSize */, sharedBuffer,
                           sessionId, creatorPid, attributionSource, trackFlags,
-                          TrackBase::TYPE_DEFAULT, portId, SIZE_MAX /*frameCountToBeReady*/,
+                          IAfTrackBase::TYPE_DEFAULT, portId, SIZE_MAX /*frameCountToBeReady*/,
                           speed, isSpatialized, isBitPerfect);
 
         lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
@@ -2752,7 +2752,7 @@
 }
 
 // addTrack_l() must be called with ThreadBase::mLock held
-status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
+status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<IAfTrack>& track)
 NO_THREAD_SAFETY_ANALYSIS  // release and re-acquire mLock
 {
     status_t status = ALREADY_EXISTS;
@@ -2762,12 +2762,12 @@
         // buffers before playing. This is to ensure the client will
         // effectively get the latency it requested.
         if (track->isExternalTrack()) {
-            TrackBase::track_state state = track->mState;
+            IAfTrackBase::track_state state = track->state();
             mLock.unlock();
             status = AudioSystem::startOutput(track->portId());
             mLock.lock();
             // abort track was stopped/paused while we released the lock
-            if (state != track->mState) {
+            if (state != track->state()) {
                 if (status == NO_ERROR) {
                     mLock.unlock();
                     AudioSystem::stopOutput(track->portId());
@@ -2793,15 +2793,15 @@
         // set retry count for buffer fill
         if (track->isOffloaded()) {
             if (track->isStopping_1()) {
-                track->mRetryCount = kMaxTrackStopRetriesOffload;
+                track->retryCount() = kMaxTrackStopRetriesOffload;
             } else {
-                track->mRetryCount = kMaxTrackStartupRetriesOffload;
+                track->retryCount() = kMaxTrackStartupRetriesOffload;
             }
-            track->mFillingUpStatus = mStandby ? Track::FS_FILLING : Track::FS_FILLED;
+            track->fillingStatus() = mStandby ? IAfTrack::FS_FILLING : IAfTrack::FS_FILLED;
         } else {
-            track->mRetryCount = kMaxTrackStartupRetries;
-            track->mFillingUpStatus =
-                    track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING;
+            track->retryCount() = kMaxTrackStartupRetries;
+            track->fillingStatus() =
+                    track->sharedBuffer() != 0 ? IAfTrack::FS_FILLED : IAfTrack::FS_FILLING;
         }
 
         sp<IAfEffectChain> chain = getEffectChain_l(track->sessionId());
@@ -2841,7 +2841,7 @@
             }
         }
 
-        track->mResetDone = false;
+        track->setResetDone(false);
         track->resetPresentationComplete();
         mActiveTracks.add(track);
         if (chain != 0) {
@@ -2858,25 +2858,25 @@
     return status;
 }
 
-bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
+bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<IAfTrack>& track)
 {
     track->terminate();
     // active tracks are removed by threadLoop()
     bool trackActive = (mActiveTracks.indexOf(track) >= 0);
-    track->mState = TrackBase::STOPPED;
+    track->setState(IAfTrackBase::STOPPED);
     if (!trackActive) {
         removeTrack_l(track);
     } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) {
         if (track->isPausePending()) {
             track->pauseAck();
         }
-        track->mState = TrackBase::STOPPING_1;
+        track->setState(IAfTrackBase::STOPPING_1);
     }
 
     return trackActive;
 }
 
-void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
+void AudioFlinger::PlaybackThread::removeTrack_l(const sp<IAfTrack>& track)
 {
     track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
 
@@ -2890,12 +2890,12 @@
         mAudioTrackCallbacks.erase(track);
     }
     if (track->isFastTrack()) {
-        int index = track->mFastIndex;
+        int index = track->fastIndex();
         ALOG_ASSERT(0 < index && index < (int)FastMixerState::sMaxFastTracks);
         ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index)));
         mFastTrackAvailMask |= 1 << index;
         // redundant as track is about to be destroyed, for dumpsys only
-        track->mFastIndex = -1;
+        track->fastIndex() = -1;
     }
     sp<IAfEffectChain> chain = getEffectChain_l(track->sessionId());
     if (chain != 0) {
@@ -3223,7 +3223,7 @@
     }
     StreamOutHalInterface::SourceMetadata metadata;
     auto backInserter = std::back_inserter(metadata.tracks);
-    for (const sp<Track> &track : mActiveTracks) {
+    for (const sp<IAfTrack>& track : mActiveTracks) {
         // No track is invalid as this is called after prepareTrack_l in the same critical section
         track->copyMetadataTo(backInserter);
     }
@@ -3274,7 +3274,7 @@
         return getStrategyForStream(AUDIO_STREAM_MUSIC);
     }
     for (size_t i = 0; i < mTracks.size(); i++) {
-        sp<Track> track = mTracks[i];
+        sp<IAfTrack> track = mTracks[i];
         if (sessionId == track->sessionId() && !track->isInvalid()) {
             return getStrategyForStream(track->streamType());
         }
@@ -3325,7 +3325,7 @@
     Mutex::Autolock _l(mLock);
 
     for (size_t i = 0; i < mTracks.size(); ++i) {
-        sp<Track> track = mTracks[i];
+        sp<IAfTrack> track = mTracks[i];
         if (event->triggerSession() == track->sessionId()) {
             (void) track->setSyncEvent(event);
             return NO_ERROR;
@@ -3342,7 +3342,7 @@
 }
 
 void AudioFlinger::PlaybackThread::threadLoop_removeTracks(
-        [[maybe_unused]] const Vector< sp<Track> >& tracksToRemove)
+        [[maybe_unused]] const Vector<sp<IAfTrack>>& tracksToRemove)
 {
     // Miscellaneous track cleanup when removed from the active list,
     // called without Thread lock but synchronized with threadLoop processing.
@@ -3493,7 +3493,7 @@
     {
         Mutex::Autolock _l(mLock);
         for (size_t i = 0; i < mTracks.size(); i++) {
-            sp<Track> track = mTracks[i];
+            sp<IAfTrack> track = mTracks[i];
             track->invalidate();
         }
         // Clear ActiveTracks to update BatteryNotifier in case active tracks remain.
@@ -3548,7 +3548,7 @@
     bool trackMatch = false;
     size_t size = mTracks.size();
     for (size_t i = 0; i < size; i++) {
-        sp<Track> t = mTracks[i];
+        sp<IAfTrack> t = mTracks[i];
         if (t->streamType() == streamType && t->isExternalTrack()) {
             t->invalidate();
             trackMatch = true;
@@ -3572,7 +3572,7 @@
     bool trackMatch = false;
     const size_t size = mTracks.size();
     for (size_t i = 0; i < size; i++) {
-        sp<Track> t = mTracks[i];
+        sp<IAfTrack> t = mTracks[i];
         if (t->isExternalTrack() && portIds.find(t->portId()) != portIds.end()) {
             t->invalidate();
             portIds.erase(t->portId());
@@ -3586,7 +3586,7 @@
 }
 
 // getTrackById_l must be called with holding thread lock
-AudioFlinger::PlaybackThread::Track* AudioFlinger::PlaybackThread::getTrackById_l(
+IAfTrack* AudioFlinger::PlaybackThread::getTrackById_l(
         audio_port_handle_t trackPortId) {
     for (size_t i = 0; i < mTracks.size(); i++) {
         if (mTracks[i]->portId() == trackPortId) {
@@ -3680,7 +3680,7 @@
     if (!audio_is_global_session(session)) {
         // Attach all tracks with same session ID to this chain.
         for (size_t i = 0; i < mTracks.size(); ++i) {
-            sp<Track> track = mTracks[i];
+            sp<IAfTrack> track = mTracks[i];
             if (session == track->sessionId()) {
                 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p",
                         track.get(), buffer);
@@ -3690,7 +3690,7 @@
         }
 
         // indicate all active tracks in the chain
-        for (const sp<Track> &track : mActiveTracks) {
+        for (const sp<IAfTrack>& track : mActiveTracks) {
             if (session == track->sessionId()) {
                 ALOGV("addEffectChain_l() activating track %p on session %d",
                         track.get(), session);
@@ -3742,7 +3742,7 @@
         if (chain == mEffectChains[i]) {
             mEffectChains.removeAt(i);
             // detach all active tracks from the chain
-            for (const sp<Track> &track : mActiveTracks) {
+            for (const sp<IAfTrack>& track : mActiveTracks) {
                 if (session == track->sessionId()) {
                     ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
                             chain.get(), session);
@@ -3752,7 +3752,7 @@
 
             // detach all tracks with same session ID from this chain
             for (size_t j = 0; j < mTracks.size(); ++j) {
-                sp<Track> track = mTracks[j];
+                sp<IAfTrack> track = mTracks[j];
                 if (session == track->sessionId()) {
                     track->setMainBuffer(reinterpret_cast<float*>(mSinkBuffer));
                     chain->decTrackCnt();
@@ -3765,14 +3765,14 @@
 }
 
 status_t AudioFlinger::PlaybackThread::attachAuxEffect(
-        const sp<AudioFlinger::PlaybackThread::Track>& track, int EffectId)
+        const sp<IAfTrack>& track, int EffectId)
 {
     Mutex::Autolock _l(mLock);
     return attachAuxEffect_l(track, EffectId);
 }
 
 status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
-        const sp<AudioFlinger::PlaybackThread::Track>& track, int EffectId)
+        const sp<IAfTrack>& track, int EffectId)
 {
     status_t status = NO_ERROR;
 
@@ -3797,7 +3797,7 @@
 void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
 {
     for (size_t i = 0; i < mTracks.size(); ++i) {
-        sp<Track> track = mTracks[i];
+        sp<IAfTrack> track = mTracks[i];
         if (track->auxEffectId() == effectId) {
             attachAuxEffect_l(track, 0);
         }
@@ -3809,7 +3809,7 @@
 {
     aflog::setThreadWriter(mNBLogWriter.get());
 
-    Vector< sp<Track> > tracksToRemove;
+    Vector<sp<IAfTrack>> tracksToRemove;
 
     mStandbyTimeNs = systemTime();
     int64_t lastLoopCountWritten = -2; // never matches "previous" loop, when loopCount = 0.
@@ -3864,7 +3864,7 @@
         Vector<sp<IAfEffectChain>> effectChains;
         audio_session_t activeHapticSessionId = AUDIO_SESSION_NONE;
         bool isHapticSessionSpatialized = false;
-        std::vector<sp<Track>> activeTracks;
+        std::vector<sp<IAfTrack>> activeTracks;
 
         // If the device is AUDIO_DEVICE_OUT_BUS, check for downstream latency.
         //
@@ -4082,14 +4082,15 @@
 
                     // Tally underrun frames as we are inserting 0s here.
                     for (const auto& track : activeTracks) {
-                        if (track->mFillingUpStatus == Track::FS_ACTIVE
+                        if (track->fillingStatus() == IAfTrack::FS_ACTIVE
                                 && !track->isStopped()
                                 && !track->isPaused()
                                 && !track->isTerminated()) {
                             ALOGV("%s: track(%d) %s underrun due to thread sleep of %zu frames",
                                     __func__, track->id(), track->getTrackStateAsString(),
                                     mNormalFrameCount);
-                            track->mAudioTrackServerProxy->tallyUnderrunFrames(mNormalFrameCount);
+                            track->audioTrackServerProxy()->tallyUnderrunFrames(
+                                    mNormalFrameCount);
                         }
                     }
                 }
@@ -4564,10 +4565,10 @@
                     ? systemTime() : mLastIoBeginNs;
         }
 
-        for (const sp<Track> &t : mActiveTracks) {
+        for (const sp<IAfTrack>& t : mActiveTracks) {
             if (!t->isFastTrack()) {
                 t->updateTrackFrameInfo(
-                        t->mAudioTrackServerProxy->framesReleased(),
+                        t->audioTrackServerProxy()->framesReleased(),
                         mFramesWritten,
                         mSampleRate,
                         mTimestamp);
@@ -4595,7 +4596,7 @@
 }
 
 // removeTracks_l() must be called with ThreadBase::mLock held
-void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove)
+void AudioFlinger::PlaybackThread::removeTracks_l(const Vector<sp<IAfTrack>>& tracksToRemove)
 NO_THREAD_SAFETY_ANALYSIS  // release and re-acquire mLock
 {
     for (const auto& track : tracksToRemove) {
@@ -4822,13 +4823,13 @@
     return status;
 }
 
-void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track)
+void AudioFlinger::PlaybackThread::addPatchTrack(const sp<IAfPatchTrack>& track)
 {
     Mutex::Autolock _l(mLock);
     mTracks.add(track);
 }
 
-void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track)
+void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<IAfPatchTrack>& track)
 {
     Mutex::Autolock _l(mLock);
     destroyTrack_l(track);
@@ -5295,7 +5296,7 @@
 
 // prepareTracks_l() must be called with ThreadBase::mLock held
 AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
-        Vector< sp<Track> > *tracksToRemove)
+        Vector<sp<IAfTrack>>* tracksToRemove)
 {
     // clean up deleted track ids in AudioMixer before allocating new tracks
     (void)mTracks.processDeletedTrackIds([this](int trackId) {
@@ -5371,23 +5372,23 @@
         // tallyUnderrunFrames() is called to update the track counters
         // with the number of underrun frames for a particular mixer period.
         // We defer tallying until we know the final mixer status.
-        void tallyUnderrunFrames(const sp<Track>& track, size_t underrunFrames) {
+        void tallyUnderrunFrames(const sp<IAfTrack>& track, size_t underrunFrames) {
             mUnderrunFrames.emplace_back(track, underrunFrames);
         }
 
     private:
         const mixer_state * const mMixerStatus;
         ThreadMetrics * const mThreadMetrics;
-        std::vector<std::pair<sp<Track>, size_t>> mUnderrunFrames;
+        std::vector<std::pair<sp<IAfTrack>, size_t>> mUnderrunFrames;
     } deferredOperations(&mixerStatus, &mThreadMetrics);
     // implicit nested scope for variable capture
 
     bool noFastHapticTrack = true;
     for (size_t i=0 ; i<count ; i++) {
-        const sp<Track> t = mActiveTracks[i];
+        const sp<IAfTrack> t = mActiveTracks[i];
 
         // this const just means the local variable doesn't change
-        Track* const track = t.get();
+        IAfTrack* const track = t.get();
 
         // process fast tracks
         if (track->isFastTrack()) {
@@ -5405,7 +5406,7 @@
             // The converse, of removing an (active) track and then creating a new track
             // at the identical fast mixer slot within the same normal mix cycle,
             // is impossible because the slot isn't marked available until the end of each cycle.
-            int j = track->mFastIndex;
+            int j = track->fastIndex();
             ALOG_ASSERT(0 < j && j < (int)FastMixerState::sMaxFastTracks);
             ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j)));
             FastTrack *fastTrack = &state->mFastTracks[j];
@@ -5415,13 +5416,13 @@
             FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j];
             FastTrackUnderruns underruns = ftDump->mUnderruns;
             uint32_t recentFull = (underruns.mBitFields.mFull -
-                    track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK;
+                    track->fastTrackUnderruns().mBitFields.mFull) & UNDERRUN_MASK;
             uint32_t recentPartial = (underruns.mBitFields.mPartial -
-                    track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK;
+                    track->fastTrackUnderruns().mBitFields.mPartial) & UNDERRUN_MASK;
             uint32_t recentEmpty = (underruns.mBitFields.mEmpty -
-                    track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK;
+                    track->fastTrackUnderruns().mBitFields.mEmpty) & UNDERRUN_MASK;
             uint32_t recentUnderruns = recentPartial + recentEmpty;
-            track->mObservedUnderruns = underruns;
+            track->fastTrackUnderruns() = underruns;
             // don't count underruns that occur while stopping or pausing
             // or stopped which can occur when flush() is called while active
             size_t underrunFrames = 0;
@@ -5431,30 +5432,30 @@
                 underrunFrames = recentUnderruns * mFrameCount;
             }
             // Immediately account for FastTrack underruns.
-            track->mAudioTrackServerProxy->tallyUnderrunFrames(underrunFrames);
+            track->audioTrackServerProxy()->tallyUnderrunFrames(underrunFrames);
 
             // This is similar to the state machine for normal tracks,
             // with a few modifications for fast tracks.
             bool isActive = true;
-            switch (track->mState) {
-            case TrackBase::STOPPING_1:
+            switch (track->state()) {
+            case IAfTrackBase::STOPPING_1:
                 // track stays active in STOPPING_1 state until first underrun
                 if (recentUnderruns > 0 || track->isTerminated()) {
-                    track->mState = TrackBase::STOPPING_2;
+                    track->setState(IAfTrackBase::STOPPING_2);
                 }
                 break;
-            case TrackBase::PAUSING:
+            case IAfTrackBase::PAUSING:
                 // ramp down is not yet implemented
                 track->setPaused();
                 break;
-            case TrackBase::RESUMING:
+            case IAfTrackBase::RESUMING:
                 // ramp up is not yet implemented
-                track->mState = TrackBase::ACTIVE;
+                track->setState(IAfTrackBase::ACTIVE);
                 break;
-            case TrackBase::ACTIVE:
+            case IAfTrackBase::ACTIVE:
                 if (recentFull > 0 || recentPartial > 0) {
                     // track has provided at least some frames recently: reset retry count
-                    track->mRetryCount = kMaxTrackRetries;
+                    track->retryCount() = kMaxTrackRetries;
                 }
                 if (recentUnderruns == 0) {
                     // no recent underruns: stay active
@@ -5468,7 +5469,7 @@
                         break;
                     }
                     // there has recently been an "empty" underrun: decrement the retry counter
-                    if (--(track->mRetryCount) > 0) {
+                    if (--(track->retryCount()) > 0) {
                         break;
                     }
                     // indicate to client process that the track was disabled because of underrun;
@@ -5479,10 +5480,10 @@
                     break;
                 }
                 FALLTHROUGH_INTENDED;
-            case TrackBase::STOPPING_2:
-            case TrackBase::PAUSED:
-            case TrackBase::STOPPED:
-            case TrackBase::FLUSHED:   // flush() while active
+            case IAfTrackBase::STOPPING_2:
+            case IAfTrackBase::PAUSED:
+            case IAfTrackBase::STOPPED:
+            case IAfTrackBase::FLUSHED:   // flush() while active
                 // Check for presentation complete if track is inactive
                 // We have consumed all the buffers of this track.
                 // This would be incomplete if we auto-paused on underrun
@@ -5499,7 +5500,7 @@
                     }
                 }
                 if (track->isStopping_2()) {
-                    track->mState = TrackBase::STOPPED;
+                    track->setState(IAfTrackBase::STOPPED);
                 }
                 if (track->isStopped()) {
                     // Can't reset directly, as fast mixer is still polling this track
@@ -5509,20 +5510,20 @@
                 }
                 isActive = false;
                 break;
-            case TrackBase::IDLE:
+            case IAfTrackBase::IDLE:
             default:
-                LOG_ALWAYS_FATAL("unexpected track state %d", (int)track->mState);
+                LOG_ALWAYS_FATAL("unexpected track state %d", (int)track->state());
             }
 
             if (isActive) {
                 // was it previously inactive?
                 if (!(state->mTrackMask & (1 << j))) {
-                    ExtendedAudioBufferProvider *eabp = track;
-                    VolumeProvider *vp = track;
+                    ExtendedAudioBufferProvider *eabp = track->asExtendedAudioBufferProvider();
+                    VolumeProvider *vp = track->asVolumeProvider();
                     fastTrack->mBufferProvider = eabp;
                     fastTrack->mVolumeProvider = vp;
-                    fastTrack->mChannelMask = track->mChannelMask;
-                    fastTrack->mFormat = track->mFormat;
+                    fastTrack->mChannelMask = track->channelMask();
+                    fastTrack->mFormat = track->format();
                     fastTrack->mHapticPlaybackEnabled = track->getHapticPlaybackEnabled();
                     fastTrack->mHapticIntensity = track->getHapticIntensity();
                     fastTrack->mHapticMaxAmplitude = track->getHapticMaxAmplitude();
@@ -5531,7 +5532,7 @@
                     didModify = true;
                     // no acknowledgement required for newly active tracks
                 }
-                sp<AudioTrackServerProxy> proxy = track->mAudioTrackServerProxy;
+                sp<AudioTrackServerProxy> proxy = track->audioTrackServerProxy();
                 float volume;
                 if (track->isPlaybackRestricted() || mStreamTypes[track->streamType()].mute) {
                     volume = 0.f;
@@ -5546,7 +5547,7 @@
                 const float vh = track->getVolumeHandler()->getVolume(
                     proxy->framesReleased()).first;
                 volume *= vh;
-                track->mCachedVolume = volume;
+                track->setCachedVolume(volume);
                 gain_minifloat_packed_t vlr = proxy->getVolumeLR();
                 float vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
                 float vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
@@ -5582,13 +5583,13 @@
                     // TODO Remove the ALOGW when this theory is confirmed.
                     ALOGW("fast track %d should have been active; "
                             "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d",
-                            j, (int)track->mState, state->mTrackMask, recentUnderruns,
+                            j, (int)track->state(), state->mTrackMask, recentUnderruns,
                             track->sharedBuffer() != 0);
                     // Since the FastMixer state already has the track inactive, do nothing here.
                 }
                 tracksToRemove->add(track);
                 // Avoids a misleading display in dumpsys
-                track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL;
+                track->fastTrackUnderruns().mBitFields.mMostRecent = UNDERRUN_FULL;
             }
             if (fastTrack->mHapticPlaybackEnabled != track->getHapticPlaybackEnabled()) {
                 fastTrack->mHapticPlaybackEnabled = track->getHapticPlaybackEnabled();
@@ -5610,14 +5611,14 @@
         if (!mAudioMixer->exists(trackId)) {
             status_t status = mAudioMixer->create(
                     trackId,
-                    track->mChannelMask,
-                    track->mFormat,
-                    track->mSessionId);
+                    track->channelMask(),
+                    track->format(),
+                    track->sessionId());
             if (status != OK) {
                 ALOGW("%s(): AudioMixer cannot create track(%d)"
                         " mask %#x, format %#x, sessionId %d",
                         __func__, trackId,
-                        track->mChannelMask, track->mFormat, track->mSessionId);
+                        track->channelMask(), track->format(), track->sessionId());
                 tracksToRemove->add(track);
                 track->invalidate(); // consider it dead.
                 continue;
@@ -5630,8 +5631,8 @@
         // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
         // during last round
         size_t desiredFrames;
-        const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate();
-        const AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate();
+        const uint32_t sampleRate = track->audioTrackServerProxy()->getSampleRate();
+        const AudioPlaybackRate playbackRate = track->audioTrackServerProxy()->getPlaybackRate();
 
         desiredFrames = sourceFramesNeededWithTimestretch(
                 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed);
@@ -5681,11 +5682,11 @@
 
 
             int param = AudioMixer::VOLUME;
-            if (track->mFillingUpStatus == Track::FS_FILLED) {
+            if (track->fillingStatus() == IAfTrack::FS_FILLED) {
                 // no ramp for the first volume setting
-                track->mFillingUpStatus = Track::FS_ACTIVE;
-                if (track->mState == TrackBase::RESUMING) {
-                    track->mState = TrackBase::ACTIVE;
+                track->fillingStatus() = IAfTrack::FS_ACTIVE;
+                if (track->state() == IAfTrackBase::RESUMING) {
+                    track->setState(IAfTrackBase::ACTIVE);
                     // If a new track is paused immediately after start, do not ramp on resume.
                     if (cblk->mServer != 0) {
                         param = AudioMixer::RAMP_VOLUME;
@@ -5706,9 +5707,9 @@
             // read original volumes with volume control
             float v = masterVolume * mStreamTypes[track->streamType()].volume;
             // Always fetch volumeshaper volume to ensure state is updated.
-            const sp<AudioTrackServerProxy> proxy = track->mAudioTrackServerProxy;
+            const sp<AudioTrackServerProxy> proxy = track->audioTrackServerProxy();
             const float vh = track->getVolumeHandler()->getVolume(
-                    track->mAudioTrackServerProxy->framesReleased()).first;
+                    track->audioTrackServerProxy()->framesReleased()).first;
 
             if (mStreamTypes[track->streamType()].mute || track->isPlaybackRestricted()) {
                 v = 0;
@@ -5770,18 +5771,18 @@
                 // Update remaining floating point volume levels
                 vlf = (float)vl / (1 << 24);
                 vrf = (float)vr / (1 << 24);
-                track->mHasVolumeController = true;
+                track->setHasVolumeController(true);
             } else {
                 // force no volume ramp when volume controller was just disabled or removed
                 // from effect chain to avoid volume spike
-                if (track->mHasVolumeController) {
+                if (track->hasVolumeController()) {
                     param = AudioMixer::VOLUME;
                 }
-                track->mHasVolumeController = false;
+                track->setHasVolumeController(false);
             }
 
             // XXX: these things DON'T need to be done each time
-            mAudioMixer->setBufferProvider(trackId, track);
+            mAudioMixer->setBufferProvider(trackId, track->asExtendedAudioBufferProvider());
             mAudioMixer->enable(trackId);
 
             mAudioMixer->setParameter(trackId, param, AudioMixer::VOLUME0, &vlf);
@@ -5889,13 +5890,14 @@
                 trackId,
                 AudioMixer::TRACK,
                 AudioMixer::HAPTIC_INTENSITY, (void *)(uintptr_t)track->getHapticIntensity());
+            const float hapticMaxAmplitude = track->getHapticMaxAmplitude();
             mAudioMixer->setParameter(
                 trackId,
                 AudioMixer::TRACK,
-                AudioMixer::HAPTIC_MAX_AMPLITUDE, (void *)(&(track->mHapticMaxAmplitude)));
+                AudioMixer::HAPTIC_MAX_AMPLITUDE, (void *)&hapticMaxAmplitude);
 
             // reset retry count
-            track->mRetryCount = kMaxTrackRetries;
+            track->retryCount() = kMaxTrackRetries;
 
             // If one track is ready, set the mixer ready if:
             //  - the mixer was not ready during previous round OR
@@ -5947,7 +5949,7 @@
             } else {
                 // No buffers for this track. Give it a few chances to
                 // fill a buffer, then remove it from active list.
-                if (--(track->mRetryCount) <= 0) {
+                if (--(track->retryCount()) <= 0) {
                     ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p",
                             trackId, this);
                     tracksToRemove->add(track);
@@ -6023,7 +6025,7 @@
         size_t i = __builtin_ctz(resetMask);
         ALOG_ASSERT(i < count);
         resetMask &= ~(1 << i);
-        sp<Track> track = mActiveTracks[i];
+        sp<IAfTrack> track = mActiveTracks[i];
         ALOG_ASSERT(track->isFastTrack() && track->isStopped());
         track->reset();
     }
@@ -6215,14 +6217,14 @@
                 const int trackId = track->id();
                 const status_t createStatus = mAudioMixer->create(
                         trackId,
-                        track->mChannelMask,
-                        track->mFormat,
-                        track->mSessionId);
+                        track->channelMask(),
+                        track->format(),
+                        track->sessionId());
                 ALOGW_IF(createStatus != NO_ERROR,
                         "%s(): AudioMixer cannot create track(%d)"
                         " mask %#x, format %#x, sessionId %d",
                         __func__,
-                        trackId, track->mChannelMask, track->mFormat, track->mSessionId);
+                        trackId, track->channelMask(), track->format(), track->sessionId());
             }
             sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
         }
@@ -6426,12 +6428,12 @@
     }
 }
 
-void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack)
+void AudioFlinger::DirectOutputThread::processVolume_l(IAfTrack *track, bool lastTrack)
 {
     float left, right;
 
     // Ensure volumeshaper state always advances even when muted.
-    const sp<AudioTrackServerProxy> proxy = track->mAudioTrackServerProxy;
+    const sp<AudioTrackServerProxy> proxy = track->audioTrackServerProxy();
 
     const size_t framesReleased = proxy->framesReleased();
     const int64_t frames = mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
@@ -6507,8 +6509,8 @@
 
 void AudioFlinger::DirectOutputThread::onAddNewTrack_l()
 {
-    sp<Track> previousTrack = mPreviousTrack.promote();
-    sp<Track> latestTrack = mActiveTracks.getLatest();
+    sp<IAfTrack> previousTrack = mPreviousTrack.promote();
+    sp<IAfTrack> latestTrack = mActiveTracks.getLatest();
 
     if (previousTrack != 0 && latestTrack != 0) {
         if (mType == DIRECT) {
@@ -6531,7 +6533,7 @@
 }
 
 AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l(
-    Vector< sp<Track> > *tracksToRemove
+    Vector<sp<IAfTrack>>* tracksToRemove
 )
 {
     size_t count = mActiveTracks.size();
@@ -6540,14 +6542,14 @@
     bool doHwResume = false;
 
     // find out which tracks need to be processed
-    for (const sp<Track> &t : mActiveTracks) {
+    for (const sp<IAfTrack>& t : mActiveTracks) {
         if (t->isInvalid()) {
             ALOGW("An invalidated track shouldn't be in active list");
             tracksToRemove->add(t);
             continue;
         }
 
-        Track* const track = t.get();
+        IAfTrack* const track = t.get();
 #ifdef VERY_VERY_VERBOSE_LOGGING
         audio_track_cblk_t* cblk = track->cblk();
 #endif
@@ -6555,7 +6557,7 @@
         // In theory an older track could underrun and restart after the new one starts
         // but as we only care about the transition phase between two tracks on a
         // direct output, it is not a problem to ignore the underrun case.
-        sp<Track> l = mActiveTracks.getLatest();
+        sp<IAfTrack> l = mActiveTracks.getLatest();
         bool last = l.get() == track;
 
         if (track->isPausePending()) {
@@ -6591,8 +6593,8 @@
         // for all its buffers to be filled before processing it.
         // Allow draining the buffer in case the client
         // app does not call stop() and relies on underrun to stop:
-        // hence the test on (track->mRetryCount > 1).
-        // If track->mRetryCount <= 1 then track is about to be disabled, paused, removed,
+        // hence the test on (track->retryCount() > 1).
+        // If track->retryCount() <= 1 then track is about to be disabled, paused, removed,
         // so we accept any nonzero amount of data delivered by the AudioTrack (which will
         // reset the retry counter).
         // Do not use a high threshold for compressed audio.
@@ -6604,7 +6606,7 @@
         const int32_t retryThreshold = targetRetryCount > 2 ? targetRetryCount - 1 : 1;
         uint32_t minFrames;
         if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()
-            && (track->mRetryCount > retryThreshold) && audio_has_proportional_frames(mFormat)) {
+            && (track->retryCount() > retryThreshold) && audio_has_proportional_frames(mFormat)) {
             minFrames = mNormalFrameCount;
         } else {
             minFrames = 1;
@@ -6622,8 +6624,8 @@
         {
             ALOGVV("track(%d) s=%08x [OK]", trackId, cblk->mServer);
 
-            if (track->mFillingUpStatus == Track::FS_FILLED) {
-                track->mFillingUpStatus = Track::FS_ACTIVE;
+            if (track->fillingStatus() == IAfTrack::FS_FILLED) {
+                track->fillingStatus() = IAfTrack::FS_ACTIVE;
                 if (last) {
                     // make sure processVolume_l() will apply new volume even if 0
                     mLeftVolFloat = mRightVolFloat = -1.0;
@@ -6636,7 +6638,7 @@
             // compute volume for this track
             processVolume_l(track, last);
             if (last) {
-                sp<Track> previousTrack = mPreviousTrack.promote();
+                sp<IAfTrack> previousTrack = mPreviousTrack.promote();
                 if (previousTrack != 0) {
                     if (track != previousTrack.get()) {
                         // Flush any data still being written from last track
@@ -6648,7 +6650,7 @@
                 mPreviousTrack = track;
 
                 // reset retry count
-                track->mRetryCount = targetRetryCount;
+                track->retryCount() = targetRetryCount;
                 mActiveTrack = t;
                 mixerStatus = MIXER_TRACKS_READY;
                 if (mHwPaused) {
@@ -6663,7 +6665,7 @@
                 mEffectChains[0]->clearInputBuffer();
             }
             if (track->isStopping_1()) {
-                track->mState = TrackBase::STOPPING_2;
+                track->setState(IAfTrackBase::STOPPING_2);
                 if (last && mHwPaused) {
                      doHwResume = true;
                      mHwPaused = false;
@@ -6681,7 +6683,7 @@
                         mOutput->presentationComplete();
                     }
                     if (track->isStopping_2()) {
-                        track->mState = TrackBase::STOPPED;
+                        track->setState(IAfTrackBase::STOPPED);
                     }
                     if (track->isStopped()) {
                         track->reset();
@@ -6694,9 +6696,9 @@
                 // Only consider last track started for mixer state control
                 bool isTimestampAdvancing = mIsTimestampAdvancing.check(mOutput);
                 if (!isTunerStream()  // tuner streams remain active in underrun
-                        && --(track->mRetryCount) <= 0) {
+                        && --(track->retryCount()) <= 0) {
                     if (isTimestampAdvancing) { // HAL is still playing audio, give us more time.
-                        track->mRetryCount = kMaxTrackRetriesOffload;
+                        track->retryCount() = kMaxTrackRetriesOffload;
                     } else {
                         ALOGV("BUFFER TIMEOUT: remove track(%d) from active list", trackId);
                         tracksToRemove->add(track);
@@ -6824,7 +6826,7 @@
     if (mTracks.size() > 0) {
         trackPaused = mTracks[mTracks.size() - 1]->isPaused();
         trackStopped = mTracks[mTracks.size() - 1]->isStopped() ||
-                           mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE;
+                           mTracks[mTracks.size() - 1]->state() == IAfTrackBase::IDLE;
     }
 
     return !mStandby && !(trackPaused || (mHwPaused && !trackStopped));
@@ -7086,7 +7088,7 @@
 }
 
 AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l(
-    Vector< sp<Track> > *tracksToRemove
+    Vector<sp<IAfTrack>>* tracksToRemove
 )
 {
     size_t count = mActiveTracks.size();
@@ -7098,8 +7100,8 @@
     ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count);
 
     // find out which tracks need to be processed
-    for (const sp<Track> &t : mActiveTracks) {
-        Track* const track = t.get();
+    for (const sp<IAfTrack>& t : mActiveTracks) {
+        IAfTrack* const track = t.get();
 #ifdef VERY_VERY_VERBOSE_LOGGING
         audio_track_cblk_t* cblk = track->cblk();
 #endif
@@ -7107,7 +7109,7 @@
         // In theory an older track could underrun and restart after the new one starts
         // but as we only care about the transition phase between two tracks on a
         // direct output, it is not a problem to ignore the underrun case.
-        sp<Track> l = mActiveTracks.getLatest();
+        sp<IAfTrack> l = mActiveTracks.getLatest();
         bool last = l.get() == track;
 
         if (track->isInvalid()) {
@@ -7116,7 +7118,7 @@
             continue;
         }
 
-        if (track->mState == TrackBase::IDLE) {
+        if (track->state() == IAfTrackBase::IDLE) {
             ALOGW("An idle track shouldn't be in active list");
             continue;
         }
@@ -7148,9 +7150,9 @@
             tracksToRemove->add(track);
         } else if (track->isFlushPending()) {
             if (track->isStopping_1()) {
-                track->mRetryCount = kMaxTrackStopRetriesOffload;
+                track->retryCount() = kMaxTrackStopRetriesOffload;
             } else {
-                track->mRetryCount = kMaxTrackRetriesOffload;
+                track->retryCount() = kMaxTrackRetriesOffload;
             }
             track->flushAck();
             if (last) {
@@ -7182,8 +7184,8 @@
         }  else if (track->framesReady() && track->isReady() &&
                 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) {
             ALOGVV("OffloadThread: track(%d) s=%08x [OK]", track->id(), cblk->mServer);
-            if (track->mFillingUpStatus == Track::FS_FILLED) {
-                track->mFillingUpStatus = Track::FS_ACTIVE;
+            if (track->fillingStatus() == IAfTrack::FS_FILLED) {
+                track->fillingStatus() = IAfTrack::FS_ACTIVE;
                 if (last) {
                     // make sure processVolume_l() will apply new volume even if 0
                     mLeftVolFloat = mRightVolFloat = -1.0;
@@ -7191,7 +7193,7 @@
             }
 
             if (last) {
-                sp<Track> previousTrack = mPreviousTrack.promote();
+                sp<IAfTrack> previousTrack = mPreviousTrack.promote();
                 if (previousTrack != 0) {
                     if (track != previousTrack.get()) {
                         // Flush any data still being written from last track
@@ -7217,9 +7219,9 @@
                 mPreviousTrack = track;
                 // reset retry count
                 if (track->isStopping_1()) {
-                    track->mRetryCount = kMaxTrackStopRetriesOffload;
+                    track->retryCount() = kMaxTrackStopRetriesOffload;
                 } else {
-                    track->mRetryCount = kMaxTrackRetriesOffload;
+                    track->retryCount() = kMaxTrackRetriesOffload;
                 }
                 mActiveTrack = t;
                 mixerStatus = MIXER_TRACKS_READY;
@@ -7227,7 +7229,7 @@
         } else {
             ALOGVV("OffloadThread: track(%d) s=%08x [NOT READY]", track->id(), cblk->mServer);
             if (track->isStopping_1()) {
-                if (--(track->mRetryCount) <= 0) {
+                if (--(track->retryCount()) <= 0) {
                     // Hardware buffer can hold a large amount of audio so we must
                     // wait for all current track's data to drain before we say
                     // that the track is stopped.
@@ -7235,7 +7237,8 @@
                         // Only start draining when all data in mixbuffer
                         // has been written
                         ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2");
-                        track->mState = TrackBase::STOPPING_2; // so presentation completes after
+                        track->setState(IAfTrackBase::STOPPING_2);
+                        // so presentation completes after
                         // drain do not drain if no data was ever sent to HAL (mStandby == true)
                         if (last && !mStandby) {
                             // do not modify drain sequence if we are already draining. This happens
@@ -7255,13 +7258,13 @@
                         }
                     }
                 } else if (last) {
-                    ALOGV("stopping1 underrun retries left %d", track->mRetryCount);
+                    ALOGV("stopping1 underrun retries left %d", track->retryCount());
                     mixerStatus = MIXER_TRACKS_ENABLED;
                 }
             } else if (track->isStopping_2()) {
                 // Drain has completed or we are in standby, signal presentation complete
                 if (!(mDrainSequence & 1) || !last || mStandby) {
-                    track->mState = TrackBase::STOPPED;
+                    track->setState(IAfTrackBase::STOPPED);
                     mOutput->presentationComplete();
                     track->presentationComplete(latency_l()); // always returns true
                     track->reset();
@@ -7281,9 +7284,9 @@
                 // fill a buffer, then remove it from active list.
                 bool isTimestampAdvancing = mIsTimestampAdvancing.check(mOutput);
                 if (!isTunerStream()  // tuner streams remain active in underrun
-                        && --(track->mRetryCount) <= 0) {
+                        && --(track->retryCount()) <= 0) {
                     if (isTimestampAdvancing) { // HAL is still playing audio, give us more time.
-                        track->mRetryCount = kMaxTrackRetriesOffload;
+                        track->retryCount() = kMaxTrackRetriesOffload;
                     } else {
                         ALOGV("OffloadThread: BUFFER TIMEOUT: remove track(%d) from active list",
                                 track->id());
@@ -7526,7 +7529,7 @@
     attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(
       IPCThreadState::self()->getCallingPid()));
     attributionSource.token = sp<BBinder>::make();
-    sp<OutputTrack> outputTrack = new OutputTrack(thread,
+    sp<IAfOutputTrack> outputTrack = IAfOutputTrack::create(thread,
                                             this,
                                             mSampleRate,
                                             mFormat,
@@ -7939,7 +7942,7 @@
     ALOGV("  preExit()");
     Mutex::Autolock _l(mLock);
     for (size_t i = 0; i < mTracks.size(); i++) {
-        sp<RecordTrack> track = mTracks[i];
+        sp<IAfRecordTrack> track = mTracks[i];
         track->invalidate();
     }
     mActiveTracks.clear();
@@ -7953,7 +7956,7 @@
     inputStandBy();
 
 reacquire_wakelock:
-    sp<RecordTrack> activeTrack;
+    sp<IAfRecordTrack> activeTrack;
     {
         Mutex::Autolock _l(mLock);
         acquireWakeLock_l();
@@ -7969,13 +7972,13 @@
         Vector<sp<IAfEffectChain>> effectChains;
 
         // activeTracks accumulates a copy of a subset of mActiveTracks
-        Vector< sp<RecordTrack> > activeTracks;
+        Vector<sp<IAfRecordTrack>> activeTracks;
 
         // reference to the (first and only) active fast track
-        sp<RecordTrack> fastTrack;
+        sp<IAfRecordTrack> fastTrack;
 
         // reference to a fast track which is about to be removed
-        sp<RecordTrack> fastTrackToRemove;
+        sp<IAfRecordTrack> fastTrackToRemove;
 
         bool silenceFastCapture = false;
 
@@ -8028,40 +8031,40 @@
                     continue;
                 }
 
-                TrackBase::track_state activeTrackState = activeTrack->mState;
+                IAfTrackBase::track_state activeTrackState = activeTrack->state();
                 switch (activeTrackState) {
 
-                case TrackBase::PAUSING:
+                case IAfTrackBase::PAUSING:
                     mActiveTracks.remove(activeTrack);
-                    activeTrack->mState = TrackBase::PAUSED;
+                    activeTrack->setState(IAfTrackBase::PAUSED);
                     doBroadcast = true;
                     size--;
                     continue;
 
-                case TrackBase::STARTING_1:
+                case IAfTrackBase::STARTING_1:
                     sleepUs = 10000;
                     i++;
                     allStopped = false;
                     continue;
 
-                case TrackBase::STARTING_2:
+                case IAfTrackBase::STARTING_2:
                     doBroadcast = true;
                     if (mStandby) {
                         mThreadMetrics.logBeginInterval();
                         mThreadSnapshot.onBegin();
                         mStandby = false;
                     }
-                    activeTrack->mState = TrackBase::ACTIVE;
+                    activeTrack->setState(IAfTrackBase::ACTIVE);
                     allStopped = false;
                     break;
 
-                case TrackBase::ACTIVE:
+                case IAfTrackBase::ACTIVE:
                     allStopped = false;
                     break;
 
-                case TrackBase::IDLE:    // cannot be on ActiveTracks if idle
-                case TrackBase::PAUSED:  // cannot be on ActiveTracks if paused
-                case TrackBase::STOPPED: // cannot be on ActiveTracks if destroyed/terminated
+                case IAfTrackBase::IDLE:    // cannot be on ActiveTracks if idle
+                case IAfTrackBase::PAUSED:  // cannot be on ActiveTracks if paused
+                case IAfTrackBase::STOPPED: // cannot be on ActiveTracks if destroyed/terminated
                 default:
                     LOG_ALWAYS_FATAL("%s: Unexpected active track state:%d, id:%d, tracks:%zu",
                             __func__, activeTrackState, activeTrack->id(), size);
@@ -8308,7 +8311,7 @@
 
         // From the timestamp, input read latency is negative output write latency.
         const audio_input_flags_t flags = mInput != NULL ? mInput->flags : AUDIO_INPUT_FLAG_NONE;
-        const double latencyMs = RecordTrack::checkServerLatencySupported(mFormat, flags)
+        const double latencyMs = IAfRecordTrack::checkServerLatencySupported(mFormat, flags)
                 ? - mTimestamp.getOutputServerLatencyMs(mSampleRate) : 0.;
         if (latencyMs != 0.) { // note 0. means timestamp is empty.
             mLatencyMs.add(latencyMs);
@@ -8365,16 +8368,16 @@
             // loop over getNextBuffer to handle circular sink
             for (;;) {
 
-                activeTrack->mSink.frameCount = ~0;
-                status_t status = activeTrack->getNextBuffer(&activeTrack->mSink);
-                size_t framesOut = activeTrack->mSink.frameCount;
+                activeTrack->sinkBuffer().frameCount = ~0;
+                status_t status = activeTrack->getNextBuffer(&activeTrack->sinkBuffer());
+                size_t framesOut = activeTrack->sinkBuffer().frameCount;
                 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0));
 
                 // check available frames and handle overrun conditions
                 // if the record track isn't draining fast enough.
                 bool hasOverrun;
                 size_t framesIn;
-                activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun);
+                activeTrack->resamplerBufferProvider()->sync(&framesIn, &hasOverrun);
                 if (hasOverrun) {
                     overrun = OVERRUN_TRUE;
                 }
@@ -8388,7 +8391,7 @@
                 // RecordBufferConverter.  TODO: remove when no longer needed.
                 framesOut = min(framesOut,
                         destinationFramesPossible(
-                                framesIn, mSampleRate, activeTrack->mSampleRate));
+                                framesIn, mSampleRate, activeTrack->sampleRate()));
 
                 if (activeTrack->isDirect()) {
                     // No RecordBufferConverter used for direct streams. Pass
@@ -8396,14 +8399,15 @@
                     AudioBufferProvider::Buffer buffer;
                     buffer.frameCount = framesOut;
                     const status_t getNextBufferStatus =
-                            activeTrack->mResamplerBufferProvider->getNextBuffer(&buffer);
+                            activeTrack->resamplerBufferProvider()->getNextBuffer(&buffer);
                     if (getNextBufferStatus == OK && buffer.frameCount != 0) {
                         ALOGV_IF(buffer.frameCount != framesOut,
                                 "%s() read less than expected (%zu vs %zu)",
                                 __func__, buffer.frameCount, framesOut);
                         framesOut = buffer.frameCount;
-                        memcpy(activeTrack->mSink.raw, buffer.raw, buffer.frameCount * mFrameSize);
-                        activeTrack->mResamplerBufferProvider->releaseBuffer(&buffer);
+                        memcpy(activeTrack->sinkBuffer().raw,
+                                buffer.raw, buffer.frameCount * mFrameSize);
+                        activeTrack->resamplerBufferProvider()->releaseBuffer(&buffer);
                     } else {
                         framesOut = 0;
                         ALOGE("%s() cannot fill request, status: %d, frameCount: %zu",
@@ -8412,9 +8416,9 @@
                 } else {
                     // process frames from the RecordThread buffer provider to the RecordTrack
                     // buffer
-                    framesOut = activeTrack->mRecordBufferConverter->convert(
-                            activeTrack->mSink.raw,
-                            activeTrack->mResamplerBufferProvider,
+                    framesOut = activeTrack->recordBufferConverter()->convert(
+                            activeTrack->sinkBuffer().raw,
+                            activeTrack->resamplerBufferProvider(),
                             framesOut);
                 }
 
@@ -8424,17 +8428,18 @@
 
                 // MediaSyncEvent handling: Synchronize AudioRecord to AudioTrack completion.
                 const ssize_t framesToDrop =
-                        activeTrack->mSynchronizedRecordState.updateRecordFrames(framesOut);
+                        activeTrack->synchronizedRecordState().updateRecordFrames(framesOut);
                 if (framesToDrop == 0) {
                     // no sync event, process normally, otherwise ignore.
                     if (framesOut > 0) {
-                        activeTrack->mSink.frameCount = framesOut;
+                        activeTrack->sinkBuffer().frameCount = framesOut;
                         // Sanitize before releasing if the track has no access to the source data
                         // An idle UID receives silence from non virtual devices until active
                         if (activeTrack->isSilenced()) {
-                            memset(activeTrack->mSink.raw, 0, framesOut * activeTrack->frameSize());
+                            memset(activeTrack->sinkBuffer().raw,
+                                    0, framesOut * activeTrack->frameSize());
                         }
-                        activeTrack->releaseBuffer(&activeTrack->mSink);
+                        activeTrack->releaseBuffer(&activeTrack->sinkBuffer());
                     }
                 }
                 if (framesOut == 0) {
@@ -8463,7 +8468,7 @@
 
             // update frame information and push timestamp out
             activeTrack->updateTrackFrameInfo(
-                    activeTrack->mServerProxy->framesReleased(),
+                    activeTrack->serverProxy()->framesReleased(),
                     mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER],
                     mSampleRate, mTimestamp);
         }
@@ -8496,7 +8501,7 @@
     {
         Mutex::Autolock _l(mLock);
         for (size_t i = 0; i < mTracks.size(); i++) {
-            sp<RecordTrack> track = mTracks[i];
+            sp<IAfRecordTrack> track = mTracks[i];
             track->invalidate();
         }
         mActiveTracks.clear();
@@ -8560,7 +8565,7 @@
 }
 
 // RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held
-sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l(
+sp<IAfRecordTrack> AudioFlinger::RecordThread::createRecordTrack_l(
         const sp<Client>& client,
         const audio_attributes_t& attr,
         uint32_t *pSampleRate,
@@ -8579,7 +8584,7 @@
 {
     size_t frameCount = *pFrameCount;
     size_t notificationFrameCount = *pNotificationFrameCount;
-    sp<RecordTrack> track;
+    sp<IAfRecordTrack> track;
     status_t lStatus;
     audio_input_flags_t inputFlags = mInput->flags;
     audio_input_flags_t requestedFlags = *flags;
@@ -8725,10 +8730,10 @@
             startFrames = mSharedAudioStartFrames;
         }
 
-        track = new RecordTrack(this, client, attr, sampleRate,
+        track = IAfRecordTrack::create(this, client, attr, sampleRate,
                       format, channelMask, frameCount,
                       nullptr /* buffer */, (size_t)0 /* bufferSize */, sessionId, creatorPid,
-                      attributionSource, *flags, TrackBase::TYPE_DEFAULT, portId,
+                      attributionSource, *flags, IAfTrackBase::TYPE_DEFAULT, portId,
                       startFrames);
 
         lStatus = track->initCheck();
@@ -8758,7 +8763,7 @@
     return track;
 }
 
-status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack,
+status_t AudioFlinger::RecordThread::start(IAfRecordTrack* recordTrack,
                                            AudioSystem::sync_event_t event,
                                            audio_session_t triggerSession)
 {
@@ -8769,7 +8774,7 @@
     if (event == AudioSystem::SYNC_EVENT_NONE) {
         recordTrack->clearSyncStartEvent();
     } else if (event != AudioSystem::SYNC_EVENT_SAME) {
-        recordTrack->mSynchronizedRecordState.startRecording(
+        recordTrack->synchronizedRecordState().startRecording(
                 mAudioFlinger->createSyncEvent(
                         event, triggerSession,
                         recordTrack->sessionId(), syncStartEventCallback, recordTrack));
@@ -8784,13 +8789,13 @@
             return DEAD_OBJECT;
         }
         if (mActiveTracks.indexOf(recordTrack) >= 0) {
-            if (recordTrack->mState == TrackBase::PAUSING) {
+            if (recordTrack->state() == IAfTrackBase::PAUSING) {
                 // We haven't stopped yet (moved to PAUSED and not in mActiveTracks)
                 // so no need to startInput().
                 ALOGV("active record track PAUSING -> ACTIVE");
-                recordTrack->mState = TrackBase::ACTIVE;
+                recordTrack->setState(IAfTrackBase::ACTIVE);
             } else {
-                ALOGV("active record track state %d", (int)recordTrack->mState);
+                ALOGV("active record track state %d", (int)recordTrack->state());
             }
             return status;
         }
@@ -8798,7 +8803,7 @@
         // TODO consider other ways of handling this, such as changing the state to :STARTING and
         //      adding the track to mActiveTracks after returning from AudioSystem::startInput(),
         //      or using a separate command thread
-        recordTrack->mState = TrackBase::STARTING_1;
+        recordTrack->setState(IAfTrackBase::STARTING_1);
         mActiveTracks.add(recordTrack);
         if (recordTrack->isExternalTrack()) {
             mLock.unlock();
@@ -8806,16 +8811,16 @@
             mLock.lock();
             if (recordTrack->isInvalid()) {
                 recordTrack->clearSyncStartEvent();
-                if (status == NO_ERROR && recordTrack->mState == TrackBase::STARTING_1) {
-                    recordTrack->mState = TrackBase::STARTING_2;
+                if (status == NO_ERROR && recordTrack->state() == IAfTrackBase::STARTING_1) {
+                    recordTrack->setState(IAfTrackBase::STARTING_2);
                     // STARTING_2 forces destroy to call stopInput.
                 }
                 ALOGW("%s track %d: invalidated after startInput", __func__, recordTrack->portId());
                 return DEAD_OBJECT;
             }
-            if (recordTrack->mState != TrackBase::STARTING_1) {
+            if (recordTrack->state() != IAfTrackBase::STARTING_1) {
                 ALOGW("%s(%d): unsynchronized mState:%d change",
-                    __func__, recordTrack->id(), (int)recordTrack->mState);
+                    __func__, recordTrack->id(), (int)recordTrack->state());
                 // Someone else has changed state, let them take over,
                 // leave mState in the new state.
                 recordTrack->clearSyncStartEvent();
@@ -8842,12 +8847,12 @@
         // was initialized to some value closer to the thread's mRsmpInFront, then the track could
         // see previously buffered data before it called start(), but with greater risk of overrun.
 
-        recordTrack->mResamplerBufferProvider->reset();
+        recordTrack->resamplerBufferProvider()->reset();
         if (!recordTrack->isDirect()) {
             // clear any converter state as new data will be discontinuous
-            recordTrack->mRecordBufferConverter->reset();
+            recordTrack->recordBufferConverter()->reset();
         }
-        recordTrack->mState = TrackBase::STARTING_2;
+        recordTrack->setState(IAfTrackBase::STARTING_2);
         // signal thread to start
         mWaitWorkCV.broadcast();
         return status;
@@ -8868,31 +8873,31 @@
     }
 }
 
-bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
+bool AudioFlinger::RecordThread::stop(IAfRecordTrack* recordTrack) {
     ALOGV("RecordThread::stop");
     AutoMutex _l(mLock);
     // if we're invalid, we can't be on the ActiveTracks.
-    if (mActiveTracks.indexOf(recordTrack) < 0 || recordTrack->mState == TrackBase::PAUSING) {
+    if (mActiveTracks.indexOf(recordTrack) < 0 || recordTrack->state() == IAfTrackBase::PAUSING) {
         return false;
     }
     // note that threadLoop may still be processing the track at this point [without lock]
-    recordTrack->mState = TrackBase::PAUSING;
+    recordTrack->setState(IAfTrackBase::PAUSING);
 
     // NOTE: Waiting here is important to keep stop synchronous.
     // This is needed for proper patchRecord peer release.
-    while (recordTrack->mState == TrackBase::PAUSING && !recordTrack->isInvalid()) {
+    while (recordTrack->state() == IAfTrackBase::PAUSING && !recordTrack->isInvalid()) {
         mWaitWorkCV.broadcast(); // signal thread to stop
         mStartStopCond.wait(mLock);
     }
 
-    if (recordTrack->mState == TrackBase::PAUSED) { // successful stop
+    if (recordTrack->state() == IAfTrackBase::PAUSED) { // successful stop
         ALOGV("Record stopped OK");
         return true;
     }
 
     // don't handle anything - we've been invalidated or restarted and in a different state
     ALOGW_IF("%s(%d): unsynchronized stop, state: %d",
-            __func__, recordTrack->id(), recordTrack->mState);
+            __func__, recordTrack->id(), recordTrack->state());
     return false;
 }
 
@@ -8916,7 +8921,7 @@
     Mutex::Autolock _l(mLock);
 
     for (size_t i = 0; i < mTracks.size(); i++) {
-        sp<RecordTrack> track = mTracks[i];
+        sp<IAfRecordTrack> track = mTracks[i];
         if (eventSession == track->sessionId()) {
             (void) track->setSyncEvent(event);
             ret = NO_ERROR;
@@ -9021,7 +9026,7 @@
     }
     StreamInHalInterface::SinkMetadata metadata;
     auto backInserter = std::back_inserter(metadata.tracks);
-    for (const sp<RecordTrack> &track : mActiveTracks) {
+    for (const sp<IAfRecordTrack>& track : mActiveTracks) {
         track->copyMetadataTo(backInserter);
     }
     mInput->stream->updateSinkMetadata(metadata);
@@ -9031,10 +9036,10 @@
 }
 
 // destroyTrack_l() must be called with ThreadBase::mLock held
-void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track)
+void AudioFlinger::RecordThread::destroyTrack_l(const sp<IAfRecordTrack>& track)
 {
     track->terminate();
-    track->mState = TrackBase::STOPPED;
+    track->setState(IAfTrackBase::STOPPED);
 
     // active tracks are removed by threadLoop()
     if (mActiveTracks.indexOf(track) < 0) {
@@ -9042,7 +9047,7 @@
     }
 }
 
-void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track)
+void AudioFlinger::RecordThread::removeTrack_l(const sp<IAfRecordTrack>& track)
 {
     String8 result;
     track->appendDump(result, false /* active */);
@@ -9097,7 +9102,7 @@
         result.append(prefix);
         mTracks[0]->appendDumpHeader(result);
         for (size_t i = 0; i < numtracks ; ++i) {
-            sp<RecordTrack> track = mTracks[i];
+            sp<IAfRecordTrack> track = mTracks[i];
             if (track != 0) {
                 bool active = mActiveTracks.indexOf(track) >= 0;
                 if (active) {
@@ -9117,7 +9122,7 @@
         result.append(prefix);
         mActiveTracks[0]->appendDumpHeader(result);
         for (size_t i = 0; i < numactive; ++i) {
-            sp<RecordTrack> track = mActiveTracks[i];
+            sp<IAfRecordTrack> track = mActiveTracks[i];
             if (mTracks.indexOf(track) < 0) {
                 result.append(prefix);
                 track->appendDump(result, true /* active */);
@@ -9132,17 +9137,17 @@
 {
     Mutex::Autolock _l(mLock);
     for (size_t i = 0; i < mTracks.size() ; i++) {
-        sp<RecordTrack> track = mTracks[i];
+        sp<IAfRecordTrack> track = mTracks[i];
         if (track != 0 && track->portId() == portId) {
             track->setSilenced(silenced);
         }
     }
 }
 
-void AudioFlinger::RecordThread::ResamplerBufferProvider::reset()
+void ResamplerBufferProvider::reset()
 {
-    sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
-    RecordThread *recordThread = (RecordThread *) threadBase.get();
+    const auto threadBase = sp<AudioFlinger::ThreadBase>::cast(mRecordTrack->thread().promote());
+    auto* const recordThread = static_cast<AudioFlinger::RecordThread *>(threadBase.get());
     mRsmpInUnrel = 0;
     const int32_t rear = recordThread->mRsmpInRear;
     ssize_t deltaFrames = 0;
@@ -9162,11 +9167,11 @@
     mRsmpInFront = audio_utils::safe_sub_overflow(rear, static_cast<int32_t>(deltaFrames));
 }
 
-void AudioFlinger::RecordThread::ResamplerBufferProvider::sync(
+void ResamplerBufferProvider::sync(
         size_t *framesAvailable, bool *hasOverrun)
 {
-    sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
-    RecordThread *recordThread = (RecordThread *) threadBase.get();
+    const auto threadBase = sp<AudioFlinger::ThreadBase>::cast(mRecordTrack->thread().promote());
+    auto* const recordThread = static_cast<AudioFlinger::RecordThread *>(threadBase.get());
     const int32_t rear = recordThread->mRsmpInRear;
     const int32_t front = mRsmpInFront;
     const ssize_t filled = audio_utils::safe_sub_overflow(rear, front);
@@ -9196,16 +9201,16 @@
 }
 
 // AudioBufferProvider interface
-status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
+status_t ResamplerBufferProvider::getNextBuffer(
         AudioBufferProvider::Buffer* buffer)
 {
-    sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
+    const auto threadBase = sp<AudioFlinger::ThreadBase>::cast(mRecordTrack->thread().promote());
     if (threadBase == 0) {
         buffer->frameCount = 0;
         buffer->raw = NULL;
         return NOT_ENOUGH_DATA;
     }
-    RecordThread *recordThread = (RecordThread *) threadBase.get();
+    auto* const recordThread = static_cast<AudioFlinger::RecordThread *>(threadBase.get());
     int32_t rear = recordThread->mRsmpInRear;
     int32_t front = mRsmpInFront;
     ssize_t filled = audio_utils::safe_sub_overflow(rear, front);
@@ -9239,7 +9244,7 @@
 }
 
 // AudioBufferProvider interface
-void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer(
+void ResamplerBufferProvider::releaseBuffer(
         AudioBufferProvider::Buffer* buffer)
 {
     int32_t stepCount = static_cast<int32_t>(buffer->frameCount);
@@ -9453,7 +9458,7 @@
     KeyedVector<audio_session_t, bool> ids;
     Mutex::Autolock _l(mLock);
     for (size_t j = 0; j < mTracks.size(); ++j) {
-        sp<RecordThread::RecordTrack> track = mTracks[j];
+        sp<IAfRecordTrack> track = mTracks[j];
         audio_session_t sessionId = track->sessionId();
         if (ids.indexOfKey(sessionId) < 0) {
             ids.add(sessionId, true);
@@ -9604,7 +9609,7 @@
     int32_t oldestFront = mRsmpInRear;
     int32_t maxFilled = 0;
     for (size_t i = 0; i < mTracks.size(); i++) {
-        int32_t front = mTracks[i]->mResamplerBufferProvider->getFront();
+        int32_t front = mTracks[i]->resamplerBufferProvider()->getFront();
         int32_t filled;
         (void)__builtin_sub_overflow(mRsmpInRear, front, &filled);
         if (filled > maxFilled) {
@@ -9624,9 +9629,9 @@
         return;
     }
     for (size_t i = 0; i < mTracks.size(); i++) {
-        int32_t front = mTracks[i]->mResamplerBufferProvider->getFront();
+        int32_t front = mTracks[i]->resamplerBufferProvider()->getFront();
         front = audio_utils::safe_sub_overflow(front, offset);
-        mTracks[i]->mResamplerBufferProvider->setFront(front);
+        mTracks[i]->resamplerBufferProvider()->setFront(front);
     }
 }
 
@@ -9723,7 +9728,7 @@
     mRsmpInBuffer = rsmpInBuffer;
 }
 
-void AudioFlinger::RecordThread::addPatchTrack(const sp<PatchRecord>& record)
+void AudioFlinger::RecordThread::addPatchTrack(const sp<IAfPatchRecord>& record)
 {
     Mutex::Autolock _l(mLock);
     mTracks.add(record);
@@ -9732,7 +9737,7 @@
     }
 }
 
-void AudioFlinger::RecordThread::deletePatchTrack(const sp<PatchRecord>& record)
+void AudioFlinger::RecordThread::deletePatchTrack(const sp<IAfPatchRecord>& record)
 {
     Mutex::Autolock _l(mLock);
     if (mSource == record->getSource()) {
@@ -9832,14 +9837,14 @@
 
 void AudioFlinger::MmapThread::disconnect()
 {
-    ActiveTracks<MmapTrack> activeTracks;
+    ActiveTracks<IAfMmapTrack> activeTracks;
     {
         Mutex::Autolock _l(mLock);
-        for (const sp<MmapTrack> &t : mActiveTracks) {
+        for (const sp<IAfMmapTrack>& t : mActiveTracks) {
             activeTracks.add(t);
         }
     }
-    for (const sp<MmapTrack> &t : activeTracks) {
+    for (const sp<IAfMmapTrack>& t : activeTracks) {
         stop(t->portId());
     }
     // This will decrement references and may cause the destruction of this thread.
@@ -10004,7 +10009,8 @@
     }
 
     // Given that MmapThread::mAttr is mutable, should a MmapTrack have attributes ?
-    sp<MmapTrack> track = new MmapTrack(this, attr == nullptr ? mAttr : *attr, mSampleRate, mFormat,
+    sp<IAfMmapTrack> track = IAfMmapTrack::create(
+            this, attr == nullptr ? mAttr : *attr, mSampleRate, mFormat,
                                         mChannelMask, mSessionId, isOutput(),
                                         client.attributionSource,
                                         IPCThreadState::self()->getCallingPid(), portId);
@@ -10016,7 +10022,7 @@
         // force volume update when a new track is added
         mHalVolFloat = -1.0f;
     } else if (!track->isSilenced_l()) {
-        for (const sp<MmapTrack> &t : mActiveTracks) {
+        for (const sp<IAfMmapTrack>& t : mActiveTracks) {
             if (t->isSilenced_l()
                     && t->uid() != static_cast<uid_t>(client.attributionSource.uid)) {
                 t->invalidate();
@@ -10061,8 +10067,8 @@
 
     Mutex::Autolock _l(mLock);
 
-    sp<MmapTrack> track;
-    for (const sp<MmapTrack> &t : mActiveTracks) {
+    sp<IAfMmapTrack> track;
+    for (const sp<IAfMmapTrack>& t : mActiveTracks) {
         if (handle == t->portId()) {
             track = t;
             break;
@@ -10419,7 +10425,7 @@
     ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
     // Attach all tracks with same session ID to this chain.
     // indicate all active tracks in the chain
-    for (const sp<MmapTrack> &track : mActiveTracks) {
+    for (const sp<IAfMmapTrack>& track : mActiveTracks) {
         if (session == track->sessionId()) {
             chain->incTrackCnt();
             chain->incActiveTrackCnt();
@@ -10447,7 +10453,7 @@
             mEffectChains.removeAt(i);
             // detach all active tracks from the chain
             // detach all tracks with same session ID from this chain
-            for (const sp<MmapTrack> &track : mActiveTracks) {
+            for (const sp<IAfMmapTrack>& track : mActiveTracks) {
                 if (session == track->sessionId()) {
                     chain->decActiveTrackCnt();
                     chain->decTrackCnt();
@@ -10519,7 +10525,7 @@
 NO_THREAD_SAFETY_ANALYSIS  // release and re-acquire mLock
 {
     sp<MmapStreamCallback> callback;
-    for (const sp<MmapTrack> &track : mActiveTracks) {
+    for (const sp<IAfMmapTrack>& track : mActiveTracks) {
         if (track->isInvalid()) {
             callback = mCallback.promote();
             if (callback == nullptr &&  mNoCallbackWarningCount < kMaxNoCallbackWarnings) {
@@ -10556,7 +10562,7 @@
         result.append(prefix);
         mActiveTracks[0]->appendDumpHeader(result);
         for (size_t i = 0; i < numtracks ; ++i) {
-            sp<MmapTrack> track = mActiveTracks[i];
+            sp<IAfMmapTrack> track = mActiveTracks[i];
             result.append(prefix);
             track->appendDump(result, true /* active */);
         }
@@ -10663,7 +10669,7 @@
 {
     Mutex::Autolock _l(mLock);
     if (streamType == mStreamType) {
-        for (const sp<MmapTrack> &track : mActiveTracks) {
+        for (const sp<IAfMmapTrack>& track : mActiveTracks) {
             track->invalidate();
         }
         broadcast_l();
@@ -10674,7 +10680,7 @@
 {
     Mutex::Autolock _l(mLock);
     bool trackMatch = false;
-    for (const sp<MmapTrack> &track : mActiveTracks) {
+    for (const sp<IAfMmapTrack>& track : mActiveTracks) {
         if (portIds.find(track->portId()) != portIds.end()) {
             track->invalidate();
             trackMatch = true;
@@ -10731,7 +10737,7 @@
                 }
             }
         }
-        for (const sp<MmapTrack> &track : mActiveTracks) {
+        for (const sp<IAfMmapTrack>& track : mActiveTracks) {
             track->setMetadataHasChanged();
             track->processMuteEvent_l(mAudioFlinger->getOrCreateAudioManager(),
                 /*muteState=*/{mMasterMute,
@@ -10751,7 +10757,7 @@
         return {}; // nothing to do
     }
     StreamOutHalInterface::SourceMetadata metadata;
-    for (const sp<MmapTrack> &track : mActiveTracks) {
+    for (const sp<IAfMmapTrack>& track : mActiveTracks) {
         // No track is invalid as this is called after prepareTrack_l in the same critical section
         playback_track_metadata_v7_t trackMetadata;
         trackMetadata.base = {
@@ -10916,7 +10922,7 @@
         return {}; // nothing to do
     }
     StreamInHalInterface::SinkMetadata metadata;
-    for (const sp<MmapTrack> &track : mActiveTracks) {
+    for (const sp<IAfMmapTrack>& track : mActiveTracks) {
         // No track is invalid as this is called after prepareTrack_l in the same critical section
         record_track_metadata_v7_t trackMetadata;
         trackMetadata.base = {
@@ -10970,7 +10976,7 @@
         : MixerThread(audioflinger, output, id, systemReady, BIT_PERFECT) {}
 
 AudioFlinger::PlaybackThread::mixer_state AudioFlinger::BitPerfectThread::prepareTracks_l(
-        Vector<sp<Track>> *tracksToRemove) {
+        Vector<sp<IAfTrack>>* tracksToRemove) {
     mixer_state result = MixerThread::prepareTracks_l(tracksToRemove);
     // If there is only one active track and it is bit-perfect, enable tee buffer.
     float volumeLeft = 1.0f;
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 8b420c0..51aeb7e 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -21,10 +21,11 @@
 
 public: // TODO(b/288339104) extract out of AudioFlinger class
 class ThreadBase : public Thread {
+    // TODO(b/288339104) remove friends
+    friend class RecordTrack;
+    friend class Track;
+    friend class TrackBase;
 public:
-
-#include "TrackBase.h"
-
     enum type_t {
         MIXER,              // Thread class is MixerThread
         DIRECT,             // Thread class is DirectOutputThread
@@ -488,7 +489,7 @@
                         result = EFFECT_SESSION;
                     }
                     for (size_t i = 0; i < tracks.size(); ++i) {
-                        const sp<TrackBase>& track = tracks[i];
+                        const sp<IAfTrackBase>& track = tracks[i];
                         if (sessionId == track->sessionId()
                                 && !track->isInvalid()       // not yet removed from tracks.
                                 && !track->isTerminated()) {
@@ -568,7 +569,7 @@
                 void invalidateTracksForAudioSession_l(audio_session_t sessionId,
                                                        const T& tracks) const {
                     for (size_t i = 0; i < tracks.size(); ++i) {
-                        const sp<TrackBase>& track = tracks[i];
+                        const sp<IAfTrackBase>& track = tracks[i];
                         if (sessionId == track->sessionId()) {
                             track->invalidate();
                         }
@@ -861,10 +862,11 @@
 // --- PlaybackThread ---
 class PlaybackThread : public ThreadBase, public StreamOutHalInterfaceCallback,
                        public VolumeInterface, public StreamOutHalInterfaceEventCallback {
+    // TODO(b/288339104) remove friends
+    friend class OutputTrack;
+    friend class Track;
 public:
 
-#include "PlaybackTracks.h"
-
     enum mixer_state {
         MIXER_IDLE,             // no active tracks
         MIXER_TRACKS_ENABLED,   // at least one active track, but no track has any data ready
@@ -913,14 +915,14 @@
     virtual     void        threadLoop_drain();
     virtual     void        threadLoop_standby();
     virtual     void        threadLoop_exit();
-    virtual     void        threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove);
+    virtual     void        threadLoop_removeTracks(const Vector<sp<IAfTrack>>& tracksToRemove);
 
                 // prepareTracks_l reads and writes mActiveTracks, and returns
                 // the pending set of tracks to remove via Vector 'tracksToRemove'.  The caller
                 // is responsible for clearing or destroying this Vector later on, when it
                 // is safe to do so. That will drop the final ref count and destroy the tracks.
-    virtual     mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0;
-                void        removeTracks_l(const Vector< sp<Track> >& tracksToRemove);
+    virtual     mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove) = 0;
+                void        removeTracks_l(const Vector<sp<IAfTrack>>& tracksToRemove);
                 status_t    handleVoipVolume_l(float *volume);
 
     // StreamOutHalInterfaceCallback implementation
@@ -976,7 +978,7 @@
 
                 void        setVolumeForOutput_l(float left, float right) const override;
 
-                sp<Track>   createTrack_l(
+                sp<IAfTrack>   createTrack_l(
                                 const sp<Client>& client,
                                 audio_stream_type_t streamType,
                                 const audio_attributes_t& attr,
@@ -1021,14 +1023,14 @@
                                             audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
                 status_t    getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
                 // Consider also removing and passing an explicit mMainBuffer initialization
-                // parameter to AF::PlaybackThread::Track::Track().
+                // parameter to AF::IAfTrack::Track().
                 float *sinkBuffer() const {
                     return reinterpret_cast<float *>(mSinkBuffer); };
 
     virtual     void detachAuxEffect_l(int effectId);
-                status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track>& track,
+                status_t attachAuxEffect(const sp<IAfTrack>& track,
                         int EffectId);
-                status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track>& track,
+                status_t attachAuxEffect_l(const sp<IAfTrack>& track,
                         int EffectId);
 
                 virtual status_t addEffectChain_l(const sp<IAfEffectChain>& chain);
@@ -1058,8 +1060,8 @@
 
                 status_t    getTimestamp_l(AudioTimestamp& timestamp);
 
-                void        addPatchTrack(const sp<PatchTrack>& track);
-                void        deletePatchTrack(const sp<PatchTrack>& track);
+                void        addPatchTrack(const sp<IAfPatchTrack>& track);
+                void        deletePatchTrack(const sp<IAfPatchTrack>& track);
 
     virtual     void        toAudioPortConfig(struct audio_port_config *config);
 
@@ -1096,7 +1098,7 @@
                     mDownStreamPatch = *patch;
                 }
 
-                PlaybackThread::Track* getTrackById_l(audio_port_handle_t trackId);
+                IAfTrack* getTrackById_l(audio_port_handle_t trackId);
 
                 bool hasMixer() const {
                     return mType == MIXER || mType == DUPLICATING || mType == SPATIALIZER;
@@ -1259,7 +1261,7 @@
                 }
 
 protected:
-    ActiveTracks<Track>     mActiveTracks;
+    ActiveTracks<IAfTrack> mActiveTracks;
 
     // Time to sleep between cycles when:
     virtual uint32_t        activeSleepTimeUs() const;      // mixer state MIXER_TRACKS_ENABLED
@@ -1304,9 +1306,9 @@
 
     DISALLOW_COPY_AND_ASSIGN(PlaybackThread);
 
-    status_t    addTrack_l(const sp<Track>& track);
-    bool        destroyTrack_l(const sp<Track>& track);
-    void        removeTrack_l(const sp<Track>& track);
+    status_t    addTrack_l(const sp<IAfTrack>& track);
+    bool        destroyTrack_l(const sp<IAfTrack>& track);
+    void        removeTrack_l(const sp<IAfTrack>& track);
 
     void        readOutputParameters_l();
     MetadataUpdate          updateMetadata_l() final;
@@ -1364,7 +1366,7 @@
         SortedVector<sp<T>> mTracks; // wrapped SortedVector.
     };
 
-    Tracks<Track>                   mTracks;
+    Tracks<IAfTrack>                   mTracks;
 
     stream_type_t                   mStreamTypes[AUDIO_STREAM_CNT];
     AudioStreamOut                  *mOutput;
@@ -1425,7 +1427,7 @@
 
     Mutex                                    mAudioTrackCbLock;
     // Record of IAudioTrackCallback
-    std::map<sp<Track>, sp<media::IAudioTrackCallback>> mAudioTrackCallbacks;
+    std::map<sp<IAfTrack>, sp<media::IAudioTrackCallback>> mAudioTrackCallbacks;
 
 private:
     // The HAL output sink is treated as non-blocking, but current implementation is blocking
@@ -1533,7 +1535,7 @@
                                     audio_channel_mask_t channelMask, audio_format_t format,
                                     audio_session_t sessionId, uid_t uid) const override;
 protected:
-    virtual     mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
+    virtual     mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove);
     virtual     uint32_t    idleSleepTimeUs() const;
     virtual     uint32_t    suspendSleepTimeUs() const;
     virtual     void        cacheParameters_l();
@@ -1663,7 +1665,7 @@
                 void        dumpInternals_l(int fd, const Vector<String16>& args) override;
 
     // threadLoop snippets
-    virtual     mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
+    virtual     mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove);
     virtual     void        threadLoop_mix();
     virtual     void        threadLoop_sleepTime();
     virtual     void        threadLoop_exit();
@@ -1679,13 +1681,13 @@
     DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
                        audio_io_handle_t id, ThreadBase::type_t type, bool systemReady,
                        const audio_offload_info_t& offloadInfo);
-    void processVolume_l(Track *track, bool lastTrack);
+    void processVolume_l(IAfTrack *track, bool lastTrack);
     bool isTunerStream() const { return (mOffloadInfo.content_id > 0); }
 
     // prepareTracks_l() tells threadLoop_mix() the name of the single active track
-    sp<Track>               mActiveTrack;
+    sp<IAfTrack>               mActiveTrack;
 
-    wp<Track>               mPreviousTrack;         // used to detect track switch
+    wp<IAfTrack>               mPreviousTrack;         // used to detect track switch
 
     // This must be initialized for initial condition of mMasterBalance = 0 (disabled).
     float                   mMasterBalanceLeft = 1.f;
@@ -1725,7 +1727,7 @@
 
 protected:
     // threadLoop snippets
-    virtual     mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
+    virtual     mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove);
     virtual     void        threadLoop_exit();
 
     virtual     bool        waitingAsyncCallback();
@@ -1812,8 +1814,8 @@
 private:
 
                 uint32_t    mWaitTimeMs;
-    SortedVector < sp<OutputTrack> >  outputTracks;
-    SortedVector < sp<OutputTrack> >  mOutputTracks;
+    SortedVector <sp<IAfOutputTrack>>  outputTracks;
+    SortedVector <sp<IAfOutputTrack>>  mOutputTracks;
 public:
     virtual     bool        hasFastMixer() const { return false; }
                 status_t    threadloop_getHalTimestamp_l(
@@ -1864,55 +1866,12 @@
 // record thread
 class RecordThread : public ThreadBase
 {
+    // TODO(b/288339104) remove friends
+    friend class PassthruPatchRecord;
+    friend class RecordTrack;
+    friend class ResamplerBufferProvider;
 public:
 
-    class RecordTrack;
-
-    /* The ResamplerBufferProvider is used to retrieve recorded input data from the
-     * RecordThread.  It maintains local state on the relative position of the read
-     * position of the RecordTrack compared with the RecordThread.
-     */
-    class ResamplerBufferProvider : public AudioBufferProvider
-    {
-    public:
-        explicit ResamplerBufferProvider(RecordTrack* recordTrack) :
-            mRecordTrack(recordTrack),
-            mRsmpInUnrel(0), mRsmpInFront(0) { }
-        virtual ~ResamplerBufferProvider() { }
-
-        // called to set the ResamplerBufferProvider to head of the RecordThread data buffer,
-        // skipping any previous data read from the hal.
-        virtual void reset();
-
-        /* Synchronizes RecordTrack position with the RecordThread.
-         * Calculates available frames and handle overruns if the RecordThread
-         * has advanced faster than the ResamplerBufferProvider has retrieved data.
-         * TODO: why not do this for every getNextBuffer?
-         *
-         * Parameters
-         * framesAvailable:  pointer to optional output size_t to store record track
-         *                   frames available.
-         *      hasOverrun:  pointer to optional boolean, returns true if track has overrun.
-         */
-
-        virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL);
-
-        // AudioBufferProvider interface
-        virtual status_t    getNextBuffer(AudioBufferProvider::Buffer* buffer);
-        virtual void        releaseBuffer(AudioBufferProvider::Buffer* buffer);
-
-                int32_t     getFront() const { return mRsmpInFront; }
-                void        setFront(int32_t front) { mRsmpInFront = front; }
-    private:
-        RecordTrack * const mRecordTrack;
-        size_t              mRsmpInUnrel;   // unreleased frames remaining from
-                                            // most recent getNextBuffer
-                                            // for debug only
-        int32_t             mRsmpInFront;   // next available frame
-                                            // rolling counter that is never cleared
-    };
-
-#include "RecordTracks.h"
 
             RecordThread(const sp<AudioFlinger>& audioFlinger,
                     AudioStreamIn *input,
@@ -1922,8 +1881,8 @@
             virtual     ~RecordThread();
 
     // no addTrack_l ?
-    void        destroyTrack_l(const sp<RecordTrack>& track);
-    void        removeTrack_l(const sp<RecordTrack>& track);
+    void        destroyTrack_l(const sp<IAfRecordTrack>& track);
+    void        removeTrack_l(const sp<IAfRecordTrack>& track);
 
     // Thread virtuals
     virtual bool        threadLoop();
@@ -1938,7 +1897,7 @@
 
     virtual sp<IMemory> pipeMemory() const { return mPipeMemory; }
 
-            sp<AudioFlinger::RecordThread::RecordTrack>  createRecordTrack_l(
+            sp<IAfRecordTrack> createRecordTrack_l(
                     const sp<Client>& client,
                     const audio_attributes_t& attr,
                     uint32_t *pSampleRate,
@@ -1955,13 +1914,13 @@
                     audio_port_handle_t portId,
                     int32_t maxSharedAudioHistoryMs);
 
-            status_t    start(RecordTrack* recordTrack,
+            status_t start(IAfRecordTrack* recordTrack,
                               AudioSystem::sync_event_t event,
                               audio_session_t triggerSession);
 
             // ask the thread to stop the specified track, and
             // return true if the caller should then do it's part of the stopping process
-            bool        stop(RecordTrack* recordTrack);
+            bool stop(IAfRecordTrack* recordTrack);
 
             AudioStreamIn* clearInput();
             virtual sp<StreamHalInterface> stream() const;
@@ -1979,8 +1938,8 @@
             void        updateOutDevices(const DeviceDescriptorBaseVector& outDevices) override;
             void        resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs) override;
 
-            void        addPatchTrack(const sp<PatchRecord>& record);
-            void        deletePatchTrack(const sp<PatchRecord>& record);
+            void        addPatchTrack(const sp<IAfPatchRecord>& record);
+            void        deletePatchTrack(const sp<IAfPatchRecord>& record);
 
             void        readInputParameters_l();
     virtual uint32_t    getInputFramesLost();
@@ -2066,10 +2025,10 @@
 
             AudioStreamIn                       *mInput;
             Source                              *mSource;
-            SortedVector < sp<RecordTrack> >    mTracks;
+            SortedVector <sp<IAfRecordTrack>>    mTracks;
             // mActiveTracks has dual roles:  it indicates the current active track(s), and
             // is used together with mStartStopCond to indicate start()/stop() progress
-            ActiveTracks<RecordTrack>           mActiveTracks;
+            ActiveTracks<IAfRecordTrack>           mActiveTracks;
 
             Condition                           mStartStopCond;
 
@@ -2139,9 +2098,6 @@
 class MmapThread : public ThreadBase
 {
  public:
-
-#include "MmapTracks.h"
-
     MmapThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
                AudioHwDevice *hwDev, const sp<StreamHalInterface>& stream, bool systemReady,
                bool isOut);
@@ -2258,7 +2214,7 @@
                 sp<StreamHalInterface>  mHalStream;
                 sp<DeviceHalInterface>  mHalDevice;
                 AudioHwDevice* const    mAudioHwDev;
-                ActiveTracks<MmapTrack> mActiveTracks;
+                ActiveTracks<IAfMmapTrack> mActiveTracks;
                 float                   mHalVolFloat;
                 std::map<audio_port_handle_t, bool> mClientSilencedStates;
 
@@ -2363,7 +2319,7 @@
                      audio_io_handle_t id, bool systemReady);
 
 protected:
-    mixer_state prepareTracks_l(Vector<sp<Track>> *tracksToRemove) override;
+    mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove) override;
     void threadLoop_mix() override;
 
 private:
@@ -2372,4 +2328,4 @@
     float mVolumeRight = 0.f;
 };
 
-private:
+private:
\ No newline at end of file
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 7cd78c3..8f31468 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -15,14 +15,14 @@
 ** limitations under the License.
 */
 
-#ifndef INCLUDING_FROM_AUDIOFLINGER_H
-    #error This header file should only be included from AudioFlinger.h
-#endif
+#pragma once
+
+namespace android {
 
 // base for record and playback
 class TrackBase : public ExtendedAudioBufferProvider, public virtual IAfTrackBase {
 public:
-                        TrackBase(ThreadBase *thread,
+                        TrackBase(AudioFlinger::ThreadBase* thread,
                                 const sp<Client>& client,
                                 const audio_attributes_t& mAttr,
                                 uint32_t sampleRate,
@@ -322,7 +322,7 @@
                                     // true for Track, false for RecordTrack,
                                     // this could be a track type if needed later
 
-    const wp<ThreadBase> mThread;
+    const wp<AudioFlinger::ThreadBase> mThread;
     const alloc_type     mAllocType;
     /*const*/ sp<Client> mClient;   // see explanation at ~TrackBase() why not const
     sp<IMemory>         mCblkMemory;
@@ -391,8 +391,8 @@
 class PatchTrackBase : public PatchProxyBufferProvider, public virtual IAfPatchTrackBase
 {
 public:
-    using Timeout = std::optional<std::chrono::nanoseconds>;
-                        PatchTrackBase(const sp<ClientProxy>& proxy, const ThreadBase& thread,
+                        PatchTrackBase(const sp<ClientProxy>& proxy,
+                                       const AudioFlinger::ThreadBase& thread,
                                        const Timeout& timeout);
             void setPeerTimeout(std::chrono::nanoseconds timeout) final;
             void setPeerProxy(const sp<IAfPatchTrackBase>& proxy, bool holdReference) final {
@@ -417,5 +417,6 @@
     sp<RefBase>                 mPeerReferenceHold;   // keeps mPeerProxy alive during access.
     PatchProxyBufferProvider*   mPeerProxy = nullptr;
     struct timespec             mPeerTimeout{};
-
 };
+
+} // namespace android
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index fca9745..7bdd374 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -30,6 +30,10 @@
 #include <private/media/AudioTrackShared.h>
 
 #include "AudioFlinger.h"
+#include "TrackBase.h"
+#include "PlaybackTracks.h"
+#include "RecordTracks.h"
+#include "MmapTracks.h"
 
 #include <media/nbaio/Pipe.h>
 #include <media/nbaio/PipeReader.h>
@@ -76,8 +80,8 @@
 static volatile int32_t nextTrackId = 55;
 
 // TrackBase constructor must be called with AudioFlinger::mLock held
-AudioFlinger::ThreadBase::TrackBase::TrackBase(
-            ThreadBase *thread,
+TrackBase::TrackBase(
+            AudioFlinger::ThreadBase *thread,
             const sp<Client>& client,
             const audio_attributes_t& attr,
             uint32_t sampleRate,
@@ -253,7 +257,7 @@
    return attributionSource;
 }
 
-status_t AudioFlinger::ThreadBase::TrackBase::initCheck() const
+status_t TrackBase::initCheck() const
 {
     status_t status;
     if (mType == TYPE_OUTPUT || mType == TYPE_PATCH) {
@@ -264,7 +268,7 @@
     return status;
 }
 
-AudioFlinger::ThreadBase::TrackBase::~TrackBase()
+TrackBase::~TrackBase()
 {
     // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference
     mServerProxy.clear();
@@ -289,7 +293,7 @@
 // AudioBufferProvider interface
 // getNextBuffer() = 0;
 // This implementation of releaseBuffer() is used by Track and RecordTrack
-void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+void TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
 {
 #ifdef TEE_SINK
     mTee.write(buffer->raw, buffer->frameCount);
@@ -303,16 +307,15 @@
     mServerProxy->releaseBuffer(&buf);
 }
 
-status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(
+status_t TrackBase::setSyncEvent(
         const sp<audioflinger::SyncEvent>& event)
 {
     mSyncEvents.emplace_back(event);
     return NO_ERROR;
 }
 
-AudioFlinger::ThreadBase::PatchTrackBase::PatchTrackBase(const sp<ClientProxy>& proxy,
-                                                         const ThreadBase& thread,
-                                                         const Timeout& timeout)
+PatchTrackBase::PatchTrackBase(const sp<ClientProxy>& proxy,
+        const AudioFlinger::ThreadBase& thread, const Timeout& timeout)
     : mProxy(proxy)
 {
     if (timeout) {
@@ -325,7 +328,7 @@
     }
 }
 
-void AudioFlinger::ThreadBase::PatchTrackBase::setPeerTimeout(std::chrono::nanoseconds timeout) {
+void PatchTrackBase::setPeerTimeout(std::chrono::nanoseconds timeout) {
     mPeerTimeout.tv_sec = timeout.count() / std::nano::den;
     mPeerTimeout.tv_nsec = timeout.count() % std::nano::den;
 }
@@ -555,8 +558,7 @@
 // -------------------------------
 
 // static
-sp<AudioFlinger::PlaybackThread::OpPlayAudioMonitor>
-AudioFlinger::PlaybackThread::OpPlayAudioMonitor::createIfNeeded(
+sp<OpPlayAudioMonitor> OpPlayAudioMonitor::createIfNeeded(
             AudioFlinger::ThreadBase* thread,
             const AttributionSourceState& attributionSource, const audio_attributes_t& attr, int id,
             audio_stream_type_t streamType)
@@ -587,7 +589,7 @@
     return sp<OpPlayAudioMonitor>::make(thread, attributionSource, attr.usage, id, uid);
 }
 
-AudioFlinger::PlaybackThread::OpPlayAudioMonitor::OpPlayAudioMonitor(
+OpPlayAudioMonitor::OpPlayAudioMonitor(
         AudioFlinger::ThreadBase* thread,
         const AttributionSourceState& attributionSource,
         audio_usage_t usage, int id, uid_t uid)
@@ -600,7 +602,7 @@
       mPackageName(VALUE_OR_FATAL(aidl2legacy_string_view_String16(
                   attributionSource.packageName.value_or("")))) {}
 
-AudioFlinger::PlaybackThread::OpPlayAudioMonitor::~OpPlayAudioMonitor()
+OpPlayAudioMonitor::~OpPlayAudioMonitor()
 {
     if (mOpCallback != 0) {
         mAppOpsManager.stopWatchingMode(mOpCallback);
@@ -608,7 +610,7 @@
     mOpCallback.clear();
 }
 
-void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::onFirstRef()
+void OpPlayAudioMonitor::onFirstRef()
 {
     checkPlayAudioForUsage();
     if (mAttributionSource.packageName.has_value()) {
@@ -618,14 +620,14 @@
     }
 }
 
-bool AudioFlinger::PlaybackThread::OpPlayAudioMonitor::hasOpPlayAudio() const {
+bool OpPlayAudioMonitor::hasOpPlayAudio() const {
     return mHasOpPlayAudio.load();
 }
 
 // Note this method is never called (and never to be) for audio server / patch record track
 // - not called from constructor due to check on UID,
 // - not called from PlayAudioOpCallback because the callback is not installed in this case
-void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::checkPlayAudioForUsage()
+void OpPlayAudioMonitor::checkPlayAudioForUsage()
 {
     const bool hasAppOps = mAttributionSource.packageName.has_value()
         && mAppOpsManager.checkAudioOpNoThrow(
@@ -644,11 +646,11 @@
     }
 }
 
-AudioFlinger::PlaybackThread::OpPlayAudioMonitor::PlayAudioOpCallback::PlayAudioOpCallback(
+OpPlayAudioMonitor::PlayAudioOpCallback::PlayAudioOpCallback(
         const wp<OpPlayAudioMonitor>& monitor) : mMonitor(monitor)
 { }
 
-void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::PlayAudioOpCallback::opChanged(int32_t op,
+void OpPlayAudioMonitor::PlayAudioOpCallback::opChanged(int32_t op,
             const String16& packageName) {
     // we only have uid, so we need to check all package names anyway
     UNUSED(packageName);
@@ -662,7 +664,7 @@
 }
 
 // static
-void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::getPackagesForUid(
+void OpPlayAudioMonitor::getPackagesForUid(
     uid_t uid, Vector<String16>& packages)
 {
     PermissionController permissionController;
@@ -673,9 +675,57 @@
 #undef LOG_TAG
 #define LOG_TAG "AF::Track"
 
+/* static */
+sp<IAfTrack> IAfTrack::create( // TODO(b/288339104) void*
+        void * /* AudioFlinger::PlaybackThread */ thread,
+        const sp<Client>& client,
+        audio_stream_type_t streamType,
+        const audio_attributes_t& attr,
+        uint32_t sampleRate,
+        audio_format_t format,
+        audio_channel_mask_t channelMask,
+        size_t frameCount,
+        void *buffer,
+        size_t bufferSize,
+        const sp<IMemory>& sharedBuffer,
+        audio_session_t sessionId,
+        pid_t creatorPid,
+        const AttributionSourceState& attributionSource,
+        audio_output_flags_t flags,
+        track_type type,
+        audio_port_handle_t portId,
+        /** default behaviour is to start when there are as many frames
+          * ready as possible (aka. Buffer is full). */
+        size_t frameCountToBeReady,
+        float speed,
+        bool isSpatialized,
+        bool isBitPerfect) {
+    return sp<Track>::make(reinterpret_cast<AudioFlinger::PlaybackThread*>(thread),
+            client,
+            streamType,
+            attr,
+            sampleRate,
+            format,
+            channelMask,
+            frameCount,
+            buffer,
+            bufferSize,
+            sharedBuffer,
+            sessionId,
+            creatorPid,
+            attributionSource,
+            flags,
+            type,
+            portId,
+            frameCountToBeReady,
+            speed,
+            isSpatialized,
+            isBitPerfect);
+}
+
 // Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
-AudioFlinger::PlaybackThread::Track::Track(
-            PlaybackThread *thread,
+Track::Track(
+            AudioFlinger::PlaybackThread *thread,
             const sp<Client>& client,
             audio_stream_type_t streamType,
             const audio_attributes_t& attr,
@@ -709,7 +759,7 @@
                   type,
                   portId,
                   std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_TRACK) + std::to_string(portId)),
-    mFillingUpStatus(FS_INVALID),
+    mFillingStatus(FS_INVALID),
     // mRetryCount initialized later when needed
     mSharedBuffer(sharedBuffer),
     mStreamType(streamType),
@@ -799,7 +849,7 @@
     mTrackMetrics.logConstructor(creatorPid, uid, id(), traits, streamType);
 }
 
-AudioFlinger::PlaybackThread::Track::~Track()
+Track::~Track()
 {
     ALOGV("%s(%d)", __func__, mId);
 
@@ -812,7 +862,7 @@
     }
 }
 
-status_t AudioFlinger::PlaybackThread::Track::initCheck() const
+status_t Track::initCheck() const
 {
     status_t status = TrackBase::initCheck();
     if (status == NO_ERROR && mCblk == nullptr) {
@@ -821,7 +871,7 @@
     return status;
 }
 
-void AudioFlinger::PlaybackThread::Track::destroy()
+void Track::destroy()
 {
     // NOTE: destroyTrack_l() can remove a strong reference to this Track
     // by removing it from mTracks vector, so there is a risk that this Tracks's
@@ -834,10 +884,10 @@
     sp<Track> keep(this);
     { // scope for mLock
         bool wasActive = false;
-        sp<ThreadBase> thread = mThread.promote();
+        sp<AudioFlinger::ThreadBase> thread = mThread.promote();
         if (thread != 0) {
             Mutex::Autolock _l(thread->mLock);
-            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+            auto* const playbackThread = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
             wasActive = playbackThread->destroyTrack_l(this);
             forEachTeePatchTrack_l([](const auto& patchTrack) { patchTrack->destroy(); });
         }
@@ -847,7 +897,7 @@
     }
 }
 
-void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) const
+void Track::appendDumpHeader(String8& result) const
 {
     result.appendFormat("Type     Id Active Client Session Port Id S  Flags "
                         "  Format Chn mask  SRate "
@@ -858,7 +908,7 @@
                         isServerLatencySupported() ? "   Latency" : "");
 }
 
-void AudioFlinger::PlaybackThread::Track::appendDump(String8& result, bool active) const
+void Track::appendDump(String8& result, bool active) const
 {
     char trackType;
     switch (mType) {
@@ -900,7 +950,7 @@
     }
 
     char fillingStatus;
-    switch (mFillingUpStatus) {
+    switch (mFillingStatus) {
     case FS_INVALID:
         fillingStatus = 'I';
         break;
@@ -986,12 +1036,12 @@
     result.append("\n");
 }
 
-uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const {
+uint32_t Track::sampleRate() const {
     return mAudioTrackServerProxy->getSampleRate();
 }
 
 // AudioBufferProvider interface
-status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+status_t Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
 {
     ServerProxy::Buffer buf;
     size_t desiredFrames = buffer->frameCount;
@@ -1009,14 +1059,14 @@
     return status;
 }
 
-void AudioFlinger::PlaybackThread::Track::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+void Track::releaseBuffer(AudioBufferProvider::Buffer* buffer)
 {
     interceptBuffer(*buffer);
     TrackBase::releaseBuffer(buffer);
 }
 
 // TODO: compensate for time shift between HW modules.
-void AudioFlinger::PlaybackThread::Track::interceptBuffer(
+void Track::interceptBuffer(
         const AudioBufferProvider::Buffer& sourceBuffer) {
     auto start = std::chrono::steady_clock::now();
     const size_t frameCount = sourceBuffer.frameCount;
@@ -1026,12 +1076,12 @@
         // does not allow 0 frame size request contrary to getNextBuffer
     }
     for (auto& teePatch : mTeePatches) {
-        RecordThread::PatchRecord* patchRecord = teePatch.patchRecord.get();
+        IAfPatchRecord* patchRecord = teePatch.patchRecord.get();
         const size_t framesWritten = patchRecord->writeFrames(
                 sourceBuffer.i8, frameCount, mFrameSize);
         const size_t framesLeft = frameCount - framesWritten;
         ALOGW_IF(framesLeft != 0, "%s(%d) PatchRecord %d can not provide big enough "
-                 "buffer %zu/%zu, dropping %zu frames", __func__, mId, patchRecord->mId,
+                 "buffer %zu/%zu, dropping %zu frames", __func__, mId, patchRecord->id(),
                  framesWritten, frameCount, framesLeft);
     }
     auto spent = ceil<std::chrono::microseconds>(std::chrono::steady_clock::now() - start);
@@ -1047,7 +1097,7 @@
 // from a different thread than the one calling Proxy->obtainBuffer() and
 // Proxy->releaseBuffer(). Also note there is no mutual exclusion in the
 // AudioTrackServerProxy so be especially careful calling with FastTracks.
-size_t AudioFlinger::PlaybackThread::Track::framesReady() const {
+size_t Track::framesReady() const {
     if (mSharedBuffer != 0 && (isStopped() || isStopping())) {
         // Static tracks return zero frames immediately upon stopping (for FastTracks).
         // The remainder of the buffer is not drained.
@@ -1056,12 +1106,12 @@
     return mAudioTrackServerProxy->framesReady();
 }
 
-int64_t AudioFlinger::PlaybackThread::Track::framesReleased() const
+int64_t Track::framesReleased() const
 {
     return mAudioTrackServerProxy->framesReleased();
 }
 
-void AudioFlinger::PlaybackThread::Track::onTimestamp(const ExtendedTimestamp &timestamp)
+void Track::onTimestamp(const ExtendedTimestamp &timestamp)
 {
     // This call comes from a FastTrack and should be kept lockless.
     // The server side frames are already translated to client frames.
@@ -1078,14 +1128,14 @@
 }
 
 // Don't call for fast tracks; the framesReady() could result in priority inversion
-bool AudioFlinger::PlaybackThread::Track::isReady() const {
-    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
+bool Track::isReady() const {
+    if (mFillingStatus != FS_FILLING || isStopped() || isPausing()) {
         return true;
     }
 
     if (isStopping()) {
         if (framesReady() > 0) {
-            mFillingUpStatus = FS_FILLED;
+            mFillingStatus = FS_FILLED;
         }
         return true;
     }
@@ -1099,21 +1149,21 @@
     if (framesReady() >= framesToBeReady || (mCblk->mFlags & CBLK_FORCEREADY)) {
         ALOGV("%s(%d): consider track ready with %zu/%zu, target was %zu)",
               __func__, mId, framesReady(), bufferSizeInFrames, framesToBeReady);
-        mFillingUpStatus = FS_FILLED;
+        mFillingStatus = FS_FILLED;
         android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
         return true;
     }
     return false;
 }
 
-status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused,
+status_t Track::start(AudioSystem::sync_event_t event __unused,
                                                     audio_session_t triggerSession __unused)
 {
     status_t status = NO_ERROR;
     ALOGV("%s(%d): calling pid %d session %d",
             __func__, mId, IPCThreadState::self()->getCallingPid(), mSessionId);
 
-    sp<ThreadBase> thread = mThread.promote();
+    sp<AudioFlinger::ThreadBase> thread = mThread.promote();
     if (thread != 0) {
         if (isOffloaded()) {
             Mutex::Autolock _laf(thread->mAudioFlinger->mLock);
@@ -1157,7 +1207,7 @@
                     __func__, mId, (int)mThreadIoHandle);
         }
 
-        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+        auto* const playbackThread = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
 
         // states to reset position info for pcm tracks
         if (audio_is_linear_pcm(mFormat)
@@ -1244,16 +1294,16 @@
     return status;
 }
 
-void AudioFlinger::PlaybackThread::Track::stop()
+void Track::stop()
 {
     ALOGV("%s(%d): calling pid %d", __func__, mId, IPCThreadState::self()->getCallingPid());
-    sp<ThreadBase> thread = mThread.promote();
+    sp<AudioFlinger::ThreadBase> thread = mThread.promote();
     if (thread != 0) {
         Mutex::Autolock _l(thread->mLock);
         track_state state = mState;
         if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) {
             // If the track is not active (PAUSED and buffers full), flush buffers
-            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+            auto* const playbackThread = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
             if (playbackThread->mActiveTracks.indexOf(this) < 0) {
                 reset();
                 mState = STOPPED;
@@ -1266,7 +1316,7 @@
                 // move to STOPPING_2 when drain completes and then STOPPED
                 mState = STOPPING_1;
                 if (isOffloaded()) {
-                    mRetryCount = PlaybackThread::kMaxTrackStopRetriesOffload;
+                    mRetryCount = AudioFlinger::PlaybackThread::kMaxTrackStopRetriesOffload;
                 }
             }
             playbackThread->broadcast_l();
@@ -1277,13 +1327,13 @@
     }
 }
 
-void AudioFlinger::PlaybackThread::Track::pause()
+void Track::pause()
 {
     ALOGV("%s(%d): calling pid %d", __func__, mId, IPCThreadState::self()->getCallingPid());
-    sp<ThreadBase> thread = mThread.promote();
+    sp<AudioFlinger::ThreadBase> thread = mThread.promote();
     if (thread != 0) {
         Mutex::Autolock _l(thread->mLock);
-        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+        auto* const playbackThread = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
         switch (mState) {
         case STOPPING_1:
         case STOPPING_2:
@@ -1314,13 +1364,13 @@
     }
 }
 
-void AudioFlinger::PlaybackThread::Track::flush()
+void Track::flush()
 {
     ALOGV("%s(%d)", __func__, mId);
-    sp<ThreadBase> thread = mThread.promote();
+    sp<AudioFlinger::ThreadBase> thread = mThread.promote();
     if (thread != 0) {
         Mutex::Autolock _l(thread->mLock);
-        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+        auto* const playbackThread = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
 
         // Flush the ring buffer now if the track is not active in the PlaybackThread.
         // Otherwise the flush would not be done until the track is resumed.
@@ -1379,7 +1429,7 @@
 }
 
 // must be called with thread lock held
-void AudioFlinger::PlaybackThread::Track::flushAck()
+void Track::flushAck()
 {
     if (!isOffloaded() && !isDirect()) {
         return;
@@ -1392,12 +1442,12 @@
     mFlushHwPending = false;
 }
 
-void AudioFlinger::PlaybackThread::Track::pauseAck()
+void Track::pauseAck()
 {
     mPauseHwPending = false;
 }
 
-void AudioFlinger::PlaybackThread::Track::reset()
+void Track::reset()
 {
     // Do not reset twice to avoid discarding data written just after a flush and before
     // the audioflinger thread detects the track is stopped.
@@ -1405,7 +1455,7 @@
         // Force underrun condition to avoid false underrun callback until first data is
         // written to buffer
         android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
-        mFillingUpStatus = FS_FILLING;
+        mFillingStatus = FS_FILLING;
         mResetDone = true;
         if (mState == FLUSHED) {
             mState = IDLE;
@@ -1413,34 +1463,35 @@
     }
 }
 
-status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs)
+status_t Track::setParameters(const String8& keyValuePairs)
 {
-    sp<ThreadBase> thread = mThread.promote();
+    sp<AudioFlinger::ThreadBase> thread = mThread.promote();
     if (thread == 0) {
         ALOGE("%s(%d): thread is dead", __func__, mId);
         return FAILED_TRANSACTION;
-    } else if ((thread->type() == ThreadBase::DIRECT) ||
-                    (thread->type() == ThreadBase::OFFLOAD)) {
+    } else if ((thread->type() == AudioFlinger::ThreadBase::DIRECT) ||
+                    (thread->type() == AudioFlinger::ThreadBase::OFFLOAD)) {
         return thread->setParameters(keyValuePairs);
     } else {
         return PERMISSION_DENIED;
     }
 }
 
-status_t AudioFlinger::PlaybackThread::Track::selectPresentation(int presentationId,
+status_t Track::selectPresentation(int presentationId,
         int programId) {
-    sp<ThreadBase> thread = mThread.promote();
+    sp<AudioFlinger::ThreadBase> thread = mThread.promote();
     if (thread == 0) {
         ALOGE("thread is dead");
         return FAILED_TRANSACTION;
-    } else if ((thread->type() == ThreadBase::DIRECT) || (thread->type() == ThreadBase::OFFLOAD)) {
-        DirectOutputThread *directOutputThread = static_cast<DirectOutputThread*>(thread.get());
+    } else if ((thread->type() == AudioFlinger::ThreadBase::DIRECT)
+            || (thread->type() == AudioFlinger::ThreadBase::OFFLOAD)) {
+        auto directOutputThread = static_cast<AudioFlinger::DirectOutputThread*>(thread.get());
         return directOutputThread->selectPresentation(presentationId, programId);
     }
     return INVALID_OPERATION;
 }
 
-VolumeShaper::Status AudioFlinger::PlaybackThread::Track::applyVolumeShaper(
+VolumeShaper::Status Track::applyVolumeShaper(
         const sp<VolumeShaper::Configuration>& configuration,
         const sp<VolumeShaper::Operation>& operation)
 {
@@ -1448,7 +1499,7 @@
 
     if (isOffloadedOrDirect()) {
         // Signal thread to fetch new volume.
-        sp<ThreadBase> thread = mThread.promote();
+        sp<AudioFlinger::ThreadBase> thread = mThread.promote();
         if (thread != 0) {
             Mutex::Autolock _l(thread->mLock);
             thread->broadcast_l();
@@ -1457,7 +1508,7 @@
     return status;
 }
 
-sp<VolumeShaper::State> AudioFlinger::PlaybackThread::Track::getVolumeShaperState(int id) const
+sp<VolumeShaper::State> Track::getVolumeShaperState(int id) const
 {
     // Note: We don't check if Thread exists.
 
@@ -1465,7 +1516,7 @@
     return mVolumeHandler->getVolumeShaperState(id);
 }
 
-void AudioFlinger::PlaybackThread::Track::setFinalVolume(float volumeLeft, float volumeRight)
+void Track::setFinalVolume(float volumeLeft, float volumeRight)
 {
     mFinalVolumeLeft = volumeLeft;
     mFinalVolumeRight = volumeRight;
@@ -1481,7 +1532,7 @@
     }
 }
 
-void AudioFlinger::PlaybackThread::Track::copyMetadataTo(MetadataInserter& backInserter) const
+void Track::copyMetadataTo(MetadataInserter& backInserter) const
 {
     // Do not forward metadata for PatchTrack with unspecified stream type
     if (mStreamType == AUDIO_STREAM_PATCH) {
@@ -1553,7 +1604,7 @@
     *backInserter++ = metadata;
 }
 
-void AudioFlinger::PlaybackThread::Track::updateTeePatches_l() {
+void Track::updateTeePatches_l() {
     if (mTeePatchesToUpdate.has_value()) {
         forEachTeePatchTrack_l([](const auto& patchTrack) { patchTrack->destroy(); });
         mTeePatches = mTeePatchesToUpdate.value();
@@ -1565,14 +1616,14 @@
     }
 }
 
-void AudioFlinger::PlaybackThread::Track::setTeePatchesToUpdate_l(TeePatches teePatchesToUpdate) {
+void Track::setTeePatchesToUpdate_l(AudioFlinger::TeePatches teePatchesToUpdate) {
     ALOGW_IF(mTeePatchesToUpdate.has_value(),
              "%s, existing tee patches to update will be ignored", __func__);
     mTeePatchesToUpdate = std::move(teePatchesToUpdate);
 }
 
 // must be called with player thread lock held
-void AudioFlinger::PlaybackThread::Track::processMuteEvent_l(const sp<
+void Track::processMuteEvent_l(const sp<
     IAudioManager>& audioManager, mute_state_t muteState)
 {
     if (mMuteState == muteState) {
@@ -1604,30 +1655,31 @@
     }
 }
 
-status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp)
+status_t Track::getTimestamp(AudioTimestamp& timestamp)
 {
     if (!isOffloaded() && !isDirect()) {
         return INVALID_OPERATION; // normal tracks handled through SSQ
     }
-    sp<ThreadBase> thread = mThread.promote();
+    sp<AudioFlinger::ThreadBase> thread = mThread.promote();
     if (thread == 0) {
         return INVALID_OPERATION;
     }
 
     Mutex::Autolock _l(thread->mLock);
-    PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+    auto* const playbackThread = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
     return playbackThread->getTimestamp_l(timestamp);
 }
 
-status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
+status_t Track::attachAuxEffect(int EffectId)
 {
-    sp<ThreadBase> thread = mThread.promote();
+    sp<AudioFlinger::ThreadBase> thread = mThread.promote();
     if (thread == nullptr) {
         return DEAD_OBJECT;
     }
 
-    sp<PlaybackThread> dstThread = (PlaybackThread *)thread.get();
-    sp<PlaybackThread> srcThread; // srcThread is initialized by call to moveAuxEffectToIo()
+    auto dstThread = sp<AudioFlinger::PlaybackThread>::cast(thread);
+    // srcThread is initialized by call to moveAuxEffectToIo()
+    sp<AudioFlinger::PlaybackThread> srcThread;
     sp<AudioFlinger> af = mClient->audioFlinger();
     status_t status = af->moveAuxEffectToIo(EffectId, dstThread, &srcThread);
 
@@ -1644,14 +1696,14 @@
     return status;
 }
 
-void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
+void Track::setAuxBuffer(int EffectId, int32_t *buffer)
 {
     mAuxEffectId = EffectId;
     mAuxBuffer = buffer;
 }
 
 // presentationComplete verified by frames, used by Mixed tracks.
-bool AudioFlinger::PlaybackThread::Track::presentationComplete(
+bool Track::presentationComplete(
         int64_t framesWritten, size_t audioHalFrames)
 {
     // TODO: improve this based on FrameMap if it exists, to ensure full drain.
@@ -1694,7 +1746,7 @@
 }
 
 // presentationComplete checked by time, used by DirectTracks.
-bool AudioFlinger::PlaybackThread::Track::presentationComplete(uint32_t latencyMs)
+bool Track::presentationComplete(uint32_t latencyMs)
 {
     // For Offloaded or Direct tracks.
 
@@ -1726,14 +1778,14 @@
     return false;
 }
 
-void AudioFlinger::PlaybackThread::Track::notifyPresentationComplete()
+void Track::notifyPresentationComplete()
 {
     // This only triggers once. TODO: should we enforce this?
     triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
     mAudioTrackServerProxy->setStreamEndDone();
 }
 
-void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
+void Track::triggerEvents(AudioSystem::sync_event_t type)
 {
     for (auto it = mSyncEvents.begin(); it != mSyncEvents.end();) {
         if ((*it)->type() == type) {
@@ -1748,7 +1800,7 @@
 
 // implement VolumeBufferProvider interface
 
-gain_minifloat_packed_t AudioFlinger::PlaybackThread::Track::getVolumeLR() const
+gain_minifloat_packed_t 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));
@@ -1773,7 +1825,7 @@
     return vlr;
 }
 
-status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(
+status_t Track::setSyncEvent(
         const sp<audioflinger::SyncEvent>& event)
 {
     if (isTerminated() || mState == PAUSED ||
@@ -1789,19 +1841,19 @@
     return NO_ERROR;
 }
 
-void AudioFlinger::PlaybackThread::Track::invalidate()
+void Track::invalidate()
 {
     TrackBase::invalidate();
     signalClientFlag(CBLK_INVALID);
 }
 
-void AudioFlinger::PlaybackThread::Track::disable()
+void Track::disable()
 {
     // TODO(b/142394888): the filling status should also be reset to filling
     signalClientFlag(CBLK_DISABLED);
 }
 
-void AudioFlinger::PlaybackThread::Track::signalClientFlag(int32_t flag)
+void Track::signalClientFlag(int32_t flag)
 {
     // FIXME should use proxy, and needs work
     audio_track_cblk_t* cblk = mCblk;
@@ -1811,23 +1863,23 @@
     (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX);
 }
 
-void AudioFlinger::PlaybackThread::Track::signal()
+void Track::signal()
 {
-    sp<ThreadBase> thread = mThread.promote();
+    sp<AudioFlinger::ThreadBase> thread = mThread.promote();
     if (thread != 0) {
-        PlaybackThread *t = (PlaybackThread *)thread.get();
+        auto* const t = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
         Mutex::Autolock _l(t->mLock);
         t->broadcast_l();
     }
 }
 
-status_t AudioFlinger::PlaybackThread::Track::getDualMonoMode(audio_dual_mono_mode_t* mode) const
+status_t Track::getDualMonoMode(audio_dual_mono_mode_t* mode) const
 {
     status_t status = INVALID_OPERATION;
     if (isOffloadedOrDirect()) {
-        sp<ThreadBase> thread = mThread.promote();
+        sp<AudioFlinger::ThreadBase> thread = mThread.promote();
         if (thread != nullptr) {
-            PlaybackThread *t = (PlaybackThread *)thread.get();
+            auto* const t = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
             Mutex::Autolock _l(t->mLock);
             status = t->mOutput->stream->getDualMonoMode(mode);
             ALOGD_IF((status == NO_ERROR) && (mDualMonoMode != *mode),
@@ -1837,13 +1889,13 @@
     return status;
 }
 
-status_t AudioFlinger::PlaybackThread::Track::setDualMonoMode(audio_dual_mono_mode_t mode)
+status_t Track::setDualMonoMode(audio_dual_mono_mode_t mode)
 {
     status_t status = INVALID_OPERATION;
     if (isOffloadedOrDirect()) {
-        sp<ThreadBase> thread = mThread.promote();
+        sp<AudioFlinger::ThreadBase> thread = mThread.promote();
         if (thread != nullptr) {
-            auto t = static_cast<PlaybackThread *>(thread.get());
+            auto* const t = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
             Mutex::Autolock lock(t->mLock);
             status = t->mOutput->stream->setDualMonoMode(mode);
             if (status == NO_ERROR) {
@@ -1854,13 +1906,13 @@
     return status;
 }
 
-status_t AudioFlinger::PlaybackThread::Track::getAudioDescriptionMixLevel(float* leveldB) const
+status_t Track::getAudioDescriptionMixLevel(float* leveldB) const
 {
     status_t status = INVALID_OPERATION;
     if (isOffloadedOrDirect()) {
-        sp<ThreadBase> thread = mThread.promote();
+        sp<AudioFlinger::ThreadBase> thread = mThread.promote();
         if (thread != nullptr) {
-            auto t = static_cast<PlaybackThread *>(thread.get());
+            auto* const t = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
             Mutex::Autolock lock(t->mLock);
             status = t->mOutput->stream->getAudioDescriptionMixLevel(leveldB);
             ALOGD_IF((status == NO_ERROR) && (mAudioDescriptionMixLevel != *leveldB),
@@ -1870,13 +1922,13 @@
     return status;
 }
 
-status_t AudioFlinger::PlaybackThread::Track::setAudioDescriptionMixLevel(float leveldB)
+status_t Track::setAudioDescriptionMixLevel(float leveldB)
 {
     status_t status = INVALID_OPERATION;
     if (isOffloadedOrDirect()) {
-        sp<ThreadBase> thread = mThread.promote();
+        sp<AudioFlinger::ThreadBase> thread = mThread.promote();
         if (thread != nullptr) {
-            auto t = static_cast<PlaybackThread *>(thread.get());
+            auto* const t = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
             Mutex::Autolock lock(t->mLock);
             status = t->mOutput->stream->setAudioDescriptionMixLevel(leveldB);
             if (status == NO_ERROR) {
@@ -1887,14 +1939,14 @@
     return status;
 }
 
-status_t AudioFlinger::PlaybackThread::Track::getPlaybackRateParameters(
+status_t Track::getPlaybackRateParameters(
         audio_playback_rate_t* playbackRate) const
 {
     status_t status = INVALID_OPERATION;
     if (isOffloadedOrDirect()) {
-        sp<ThreadBase> thread = mThread.promote();
+        sp<AudioFlinger::ThreadBase> thread = mThread.promote();
         if (thread != nullptr) {
-            auto t = static_cast<PlaybackThread *>(thread.get());
+            auto* const t = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
             Mutex::Autolock lock(t->mLock);
             status = t->mOutput->stream->getPlaybackRateParameters(playbackRate);
             ALOGD_IF((status == NO_ERROR) &&
@@ -1905,14 +1957,14 @@
     return status;
 }
 
-status_t AudioFlinger::PlaybackThread::Track::setPlaybackRateParameters(
+status_t Track::setPlaybackRateParameters(
         const audio_playback_rate_t& playbackRate)
 {
     status_t status = INVALID_OPERATION;
     if (isOffloadedOrDirect()) {
-        sp<ThreadBase> thread = mThread.promote();
+        sp<AudioFlinger::ThreadBase> thread = mThread.promote();
         if (thread != nullptr) {
-            auto t = static_cast<PlaybackThread *>(thread.get());
+            auto* const t = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
             Mutex::Autolock lock(t->mLock);
             status = t->mOutput->stream->setPlaybackRateParameters(playbackRate);
             if (status == NO_ERROR) {
@@ -1924,7 +1976,7 @@
 }
 
 //To be called with thread lock held
-bool AudioFlinger::PlaybackThread::Track::isResumePending() {
+bool Track::isResumePending() const {
     if (mState == RESUMING) {
         return true;
     }
@@ -1938,7 +1990,7 @@
 }
 
 //To be called with thread lock held
-void AudioFlinger::PlaybackThread::Track::resumeAck() {
+void Track::resumeAck() {
     if (mState == RESUMING) {
         mState = ACTIVE;
     }
@@ -1952,7 +2004,7 @@
 }
 
 //To be called with thread lock held
-void AudioFlinger::PlaybackThread::Track::updateTrackFrameInfo(
+void Track::updateTrackFrameInfo(
         int64_t trackFramesReleased, int64_t sinkFramesWritten,
         uint32_t halSampleRate, const ExtendedTimestamp &timeStamp) {
    // Make the kernel frametime available.
@@ -2032,12 +2084,12 @@
     }
 }
 
-bool AudioFlinger::PlaybackThread::Track::AudioVibrationController::setMute(bool muted) {
-    sp<ThreadBase> thread = mTrack->mThread.promote();
+bool Track::AudioVibrationController::setMute(bool muted) {
+    sp<AudioFlinger::ThreadBase> thread = mTrack->mThread.promote();
     if (thread != 0) {
         // Lock for updating mHapticPlaybackEnabled.
         Mutex::Autolock _l(thread->mLock);
-        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+        auto* const playbackThread = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
         if ((mTrack->channelMask() & AUDIO_CHANNEL_HAPTIC_ALL) != AUDIO_CHANNEL_NONE
                 && playbackThread->mHapticChannelCount > 0) {
             ALOGD("%s, haptic playback was %s for track %d",
@@ -2049,13 +2101,13 @@
     return false;
 }
 
-binder::Status AudioFlinger::PlaybackThread::Track::AudioVibrationController::mute(
+binder::Status Track::AudioVibrationController::mute(
         /*out*/ bool *ret) {
     *ret = setMute(true);
     return binder::Status::ok();
 }
 
-binder::Status AudioFlinger::PlaybackThread::Track::AudioVibrationController::unmute(
+binder::Status Track::AudioVibrationController::unmute(
         /*out*/ bool *ret) {
     *ret = setMute(false);
     return binder::Status::ok();
@@ -2065,9 +2117,28 @@
 #undef LOG_TAG
 #define LOG_TAG "AF::OutputTrack"
 
-AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
-            PlaybackThread *playbackThread,
-            DuplicatingThread *sourceThread,
+/* static */
+sp<IAfOutputTrack> IAfOutputTrack::create(  // TODO(b/288339104) void*
+        void* /* AudioFlinger::PlaybackThread */ playbackThread,
+        void* /* AudioFlinger::DuplicatingThread */ sourceThread,
+        uint32_t sampleRate,
+        audio_format_t format,
+        audio_channel_mask_t channelMask,
+        size_t frameCount,
+        const AttributionSourceState& attributionSource) {
+    return sp<OutputTrack>::make(
+            reinterpret_cast<AudioFlinger::PlaybackThread*>(playbackThread),
+            reinterpret_cast<AudioFlinger::DuplicatingThread*>(sourceThread),
+            sampleRate,
+            format,
+            channelMask,
+            frameCount,
+            attributionSource);
+}
+
+OutputTrack::OutputTrack(
+            AudioFlinger::PlaybackThread *playbackThread,
+            AudioFlinger::DuplicatingThread *sourceThread,
             uint32_t sampleRate,
             audio_format_t format,
             audio_channel_mask_t channelMask,
@@ -2102,13 +2173,13 @@
     }
 }
 
-AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
+OutputTrack::~OutputTrack()
 {
     clearBufferQueue();
     // superclass destructor will now delete the server proxy and shared memory both refer to
 }
 
-status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
+status_t OutputTrack::start(AudioSystem::sync_event_t event,
                                                           audio_session_t triggerSession)
 {
     status_t status = Track::start(event, triggerSession);
@@ -2121,7 +2192,7 @@
     return status;
 }
 
-void AudioFlinger::PlaybackThread::OutputTrack::stop()
+void OutputTrack::stop()
 {
     Track::stop();
     clearBufferQueue();
@@ -2129,10 +2200,10 @@
     mActive = false;
 }
 
-ssize_t AudioFlinger::PlaybackThread::OutputTrack::write(void* data, uint32_t frames)
+ssize_t OutputTrack::write(void* data, uint32_t frames)
 {
     if (!mActive && frames != 0) {
-        sp<ThreadBase> thread = mThread.promote();
+        sp<AudioFlinger::ThreadBase> thread = mThread.promote();
         if (thread != nullptr && thread->standby()) {
             // preload one silent buffer to trigger mixer on start()
             ClientProxy::Buffer buf { .mFrameCount = mClientProxy->getStartThresholdInFrames() };
@@ -2151,7 +2222,7 @@
             // If another OutputTrack has already started it can underrun but this is OK
             // as only silence has been played so far and the retry count is very high on
             // OutputTrack.
-            auto pt = static_cast<PlaybackThread *>(thread.get());
+            auto* const pt = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
             if (!pt->waitForHalStart()) {
                 ALOGW("%s(%d): timeout waiting for thread to exit standby", __func__, mId);
                 stop();
@@ -2240,7 +2311,7 @@
 
     // If we could not write all frames, allocate a buffer and queue it for next time.
     if (inBuffer.frameCount) {
-        sp<ThreadBase> thread = mThread.promote();
+        sp<AudioFlinger::ThreadBase> thread = mThread.promote();
         if (thread != 0 && !thread->standby()) {
             queueBuffer(inBuffer);
         }
@@ -2255,7 +2326,7 @@
     return frames - inBuffer.frameCount;  // number of frames consumed.
 }
 
-void AudioFlinger::PlaybackThread::OutputTrack::queueBuffer(Buffer& inBuffer) {
+void OutputTrack::queueBuffer(Buffer& inBuffer) {
 
     if (mBufferQueue.size() < kMaxOverFlowBuffers) {
         Buffer *pInBuffer = new Buffer;
@@ -2278,13 +2349,13 @@
     }
 }
 
-void AudioFlinger::PlaybackThread::OutputTrack::copyMetadataTo(MetadataInserter& backInserter) const
+void OutputTrack::copyMetadataTo(MetadataInserter& backInserter) const
 {
     std::lock_guard<std::mutex> lock(mTrackMetadatasMutex);
     backInserter = std::copy(mTrackMetadatas.begin(), mTrackMetadatas.end(), backInserter);
 }
 
-void AudioFlinger::PlaybackThread::OutputTrack::setMetadatas(const SourceMetadatas& metadatas) {
+void OutputTrack::setMetadatas(const SourceMetadatas& metadatas) {
     {
         std::lock_guard<std::mutex> lock(mTrackMetadatasMutex);
         mTrackMetadatas = metadatas;
@@ -2293,7 +2364,7 @@
     setMetadataHasChanged();
 }
 
-status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
+status_t OutputTrack::obtainBuffer(
         AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
 {
     ClientProxy::Buffer buf;
@@ -2307,7 +2378,7 @@
     return status;
 }
 
-void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
+void OutputTrack::clearBufferQueue()
 {
     size_t size = mBufferQueue.size();
 
@@ -2319,7 +2390,7 @@
     mBufferQueue.clear();
 }
 
-void AudioFlinger::PlaybackThread::OutputTrack::restartIfDisabled()
+void OutputTrack::restartIfDisabled()
 {
     int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
     if (mActive && (flags & CBLK_DISABLED)) {
@@ -2331,7 +2402,38 @@
 #undef LOG_TAG
 #define LOG_TAG "AF::PatchTrack"
 
-AudioFlinger::PlaybackThread::PatchTrack::PatchTrack(PlaybackThread *playbackThread,
+/* static */
+sp<IAfPatchTrack> IAfPatchTrack::create(
+        void* /* PlaybackThread */ playbackThread, // TODO(b/288339104)
+        audio_stream_type_t streamType,
+        uint32_t sampleRate,
+        audio_channel_mask_t channelMask,
+        audio_format_t format,
+        size_t frameCount,
+        void* buffer,
+        size_t bufferSize,
+        audio_output_flags_t flags,
+        const Timeout& timeout,
+        size_t frameCountToBeReady /** Default behaviour is to start
+                                         *  as soon as possible to have
+                                         *  the lowest possible latency
+                                         *  even if it might glitch. */)
+{
+    return sp<PatchTrack>::make(
+            reinterpret_cast<AudioFlinger::PlaybackThread*>(playbackThread),
+            streamType,
+            sampleRate,
+            channelMask,
+            format,
+            frameCount,
+            buffer,
+            bufferSize,
+            flags,
+            timeout,
+            frameCountToBeReady);
+}
+
+PatchTrack::PatchTrack(AudioFlinger::PlaybackThread *playbackThread,
                                                      audio_stream_type_t streamType,
                                                      uint32_t sampleRate,
                                                      audio_channel_mask_t channelMask,
@@ -2358,12 +2460,12 @@
                                       (int)(mPeerTimeout.tv_nsec / 1000000));
 }
 
-AudioFlinger::PlaybackThread::PatchTrack::~PatchTrack()
+PatchTrack::~PatchTrack()
 {
     ALOGV("%s(%d)", __func__, mId);
 }
 
-size_t AudioFlinger::PlaybackThread::PatchTrack::framesReady() const
+size_t PatchTrack::framesReady() const
 {
     if (mPeerProxy && mPeerProxy->producesBufferOnDemand()) {
         return std::numeric_limits<size_t>::max();
@@ -2372,7 +2474,7 @@
     }
 }
 
-status_t AudioFlinger::PlaybackThread::PatchTrack::start(AudioSystem::sync_event_t event,
+status_t PatchTrack::start(AudioSystem::sync_event_t event,
                                                          audio_session_t triggerSession)
 {
     status_t status = Track::start(event, triggerSession);
@@ -2384,7 +2486,7 @@
 }
 
 // AudioBufferProvider interface
-status_t AudioFlinger::PlaybackThread::PatchTrack::getNextBuffer(
+status_t PatchTrack::getNextBuffer(
         AudioBufferProvider::Buffer* buffer)
 {
     ALOG_ASSERT(mPeerProxy != 0, "%s(%d): called without peer proxy", __func__, mId);
@@ -2410,7 +2512,7 @@
     return status;
 }
 
-void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+void PatchTrack::releaseBuffer(AudioBufferProvider::Buffer* buffer)
 {
     ALOG_ASSERT(mPeerProxy != 0, "%s(%d): called without peer proxy", __func__, mId);
     Proxy::Buffer buf;
@@ -2420,7 +2522,7 @@
     TrackBase::releaseBuffer(buffer); // Note: this is the base class.
 }
 
-status_t AudioFlinger::PlaybackThread::PatchTrack::obtainBuffer(Proxy::Buffer* buffer,
+status_t PatchTrack::obtainBuffer(Proxy::Buffer* buffer,
                                                                 const struct timespec *timeOut)
 {
     status_t status = NO_ERROR;
@@ -2437,7 +2539,7 @@
     return status;
 }
 
-void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(Proxy::Buffer* buffer)
+void PatchTrack::releaseBuffer(Proxy::Buffer* buffer)
 {
     mProxy->releaseBuffer(buffer);
     restartIfDisabled();
@@ -2446,23 +2548,23 @@
     // If not, prevent an underrun from occurring by moving the track into FS_FILLING;
     // this logic avoids glitches when suspending A2DP with AudioPlaybackCapture.
     // TODO: perhaps underrun avoidance could be a track property checked in isReady() instead.
-    if (mFillingUpStatus == FS_ACTIVE
+    if (mFillingStatus == FS_ACTIVE
             && audio_is_linear_pcm(mFormat)
             && !isOffloadedOrDirect()) {
-        if (sp<ThreadBase> thread = mThread.promote();
+        if (sp<AudioFlinger::ThreadBase> thread = mThread.promote();
             thread != 0) {
-            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+            auto* const playbackThread = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
             const size_t frameCount = playbackThread->frameCount() * sampleRate()
                     / playbackThread->sampleRate();
             if (framesReady() < frameCount) {
                 ALOGD("%s(%d) Not enough data, wait for buffer to fill", __func__, mId);
-                mFillingUpStatus = FS_FILLING;
+                mFillingStatus = FS_FILLING;
             }
         }
     }
 }
 
-void AudioFlinger::PlaybackThread::PatchTrack::restartIfDisabled()
+void PatchTrack::restartIfDisabled()
 {
     if (android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags) & CBLK_DISABLED) {
         ALOGW("%s(%d): disabled due to previous underrun, restarting", __func__, mId);
@@ -2565,9 +2667,47 @@
 #undef LOG_TAG
 #define LOG_TAG "AF::RecordTrack"
 
+
+/* static */ // TODO(b/288339104)
+sp<IAfRecordTrack> IAfRecordTrack::create(void* /*AudioFlinger::RecordThread */ thread,
+        const sp<Client>& client,
+        const audio_attributes_t& attr,
+        uint32_t sampleRate,
+        audio_format_t format,
+        audio_channel_mask_t channelMask,
+        size_t frameCount,
+        void* buffer,
+        size_t bufferSize,
+        audio_session_t sessionId,
+        pid_t creatorPid,
+        const AttributionSourceState& attributionSource,
+        audio_input_flags_t flags,
+        track_type type,
+        audio_port_handle_t portId,
+        int32_t startFrames)
+{
+    return sp<RecordTrack>::make(
+        reinterpret_cast<AudioFlinger::RecordThread*>(thread),
+        client,
+        attr,
+        sampleRate,
+        format,
+        channelMask,
+        frameCount,
+        buffer,
+        bufferSize,
+        sessionId,
+        creatorPid,
+        attributionSource,
+        flags,
+        type,
+        portId,
+        startFrames);
+}
+
 // RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
-AudioFlinger::RecordThread::RecordTrack::RecordTrack(
-            RecordThread *thread,
+RecordTrack::RecordTrack(
+            AudioFlinger::RecordThread* thread,
             const sp<Client>& client,
             const audio_attributes_t& attr,
             uint32_t sampleRate,
@@ -2642,14 +2782,14 @@
     mTrackMetrics.logConstructor(creatorPid, uid(), id());
 }
 
-AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
+RecordTrack::~RecordTrack()
 {
     ALOGV("%s()", __func__);
     delete mRecordBufferConverter;
     delete mResamplerBufferProvider;
 }
 
-status_t AudioFlinger::RecordThread::RecordTrack::initCheck() const
+status_t RecordTrack::initCheck() const
 {
     status_t status = TrackBase::initCheck();
     if (status == NO_ERROR && mServerProxy == 0) {
@@ -2659,7 +2799,7 @@
 }
 
 // AudioBufferProvider interface
-status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+status_t RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
 {
     ServerProxy::Buffer buf;
     buf.mFrameCount = buffer->frameCount;
@@ -2673,12 +2813,12 @@
     return status;
 }
 
-status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
+status_t RecordTrack::start(AudioSystem::sync_event_t event,
                                                         audio_session_t triggerSession)
 {
-    sp<ThreadBase> thread = mThread.promote();
+    sp<AudioFlinger::ThreadBase> thread = mThread.promote();
     if (thread != 0) {
-        RecordThread *recordThread = (RecordThread *)thread.get();
+        auto* const recordThread = static_cast<AudioFlinger::RecordThread*>(thread.get());
         return recordThread->start(this, event, triggerSession);
     } else {
         ALOGW("%s track %d: thread was destroyed", __func__, portId());
@@ -2686,27 +2826,27 @@
     }
 }
 
-void AudioFlinger::RecordThread::RecordTrack::stop()
+void RecordTrack::stop()
 {
-    sp<ThreadBase> thread = mThread.promote();
+    sp<AudioFlinger::ThreadBase> thread = mThread.promote();
     if (thread != 0) {
-        RecordThread *recordThread = (RecordThread *)thread.get();
+        auto* const recordThread = static_cast<AudioFlinger::RecordThread*>(thread.get());
         if (recordThread->stop(this) && isExternalTrack()) {
             AudioSystem::stopInput(mPortId);
         }
     }
 }
 
-void AudioFlinger::RecordThread::RecordTrack::destroy()
+void RecordTrack::destroy()
 {
-    // see comments at AudioFlinger::PlaybackThread::Track::destroy()
+    // see comments at Track::destroy()
     sp<RecordTrack> keep(this);
     {
         track_state priorState = mState;
-        sp<ThreadBase> thread = mThread.promote();
+        sp<AudioFlinger::ThreadBase> thread = mThread.promote();
         if (thread != 0) {
             Mutex::Autolock _l(thread->mLock);
-            RecordThread *recordThread = (RecordThread *) thread.get();
+            auto* const recordThread = static_cast<AudioFlinger::RecordThread*>(thread.get());
             priorState = mState;
             if (!mSharedAudioPackageName.empty()) {
                 recordThread->resetAudioHistory_l();
@@ -2737,7 +2877,7 @@
     }
 }
 
-void AudioFlinger::RecordThread::RecordTrack::invalidate()
+void RecordTrack::invalidate()
 {
     TrackBase::invalidate();
     // FIXME should use proxy, and needs work
@@ -2749,7 +2889,7 @@
 }
 
 
-void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) const
+void RecordTrack::appendDumpHeader(String8& result) const
 {
     result.appendFormat("Active     Id Client Session Port Id  S  Flags  "
                         " Format Chn mask  SRate Source  "
@@ -2757,7 +2897,7 @@
                         isServerLatencySupported() ? "   Latency" : "");
 }
 
-void AudioFlinger::RecordThread::RecordTrack::appendDump(String8& result, bool active) const
+void RecordTrack::appendDump(String8& result, bool active) const
 {
     result.appendFormat("%c%5s %6d %6u %7u %7u  %2s 0x%03X "
             "%08X %08X %6u %6X "
@@ -2796,11 +2936,11 @@
 }
 
 // This is invoked by SyncEvent callback.
-void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(
+void RecordTrack::handleSyncStartEvent(
         const sp<audioflinger::SyncEvent>& event)
 {
     size_t framesToDrop = 0;
-    sp<ThreadBase> threadBase = mThread.promote();
+    sp<AudioFlinger::ThreadBase> threadBase = mThread.promote();
     if (threadBase != 0) {
         // TODO: use actual buffer filling status instead of 2 buffers when info is available
         // from audio HAL
@@ -2810,12 +2950,12 @@
     mSynchronizedRecordState.onPlaybackFinished(event, framesToDrop);
 }
 
-void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent()
+void RecordTrack::clearSyncStartEvent()
 {
     mSynchronizedRecordState.clear();
 }
 
-void AudioFlinger::RecordThread::RecordTrack::updateTrackFrameInfo(
+void RecordTrack::updateTrackFrameInfo(
         int64_t trackFramesReleased, int64_t sourceFramesRead,
         uint32_t halSampleRate, const ExtendedTimestamp &timestamp)
 {
@@ -2855,40 +2995,40 @@
     mServerLatencyMs.store(latencyMs);
 }
 
-status_t AudioFlinger::RecordThread::RecordTrack::getActiveMicrophones(
+status_t RecordTrack::getActiveMicrophones(
         std::vector<media::MicrophoneInfoFw>* activeMicrophones) const
 {
-    sp<ThreadBase> thread = mThread.promote();
+    sp<AudioFlinger::ThreadBase> thread = mThread.promote();
     if (thread != 0) {
-        RecordThread *recordThread = (RecordThread *)thread.get();
+        auto* const recordThread = static_cast<AudioFlinger::RecordThread*>(thread.get());
         return recordThread->getActiveMicrophones(activeMicrophones);
     } else {
         return BAD_VALUE;
     }
 }
 
-status_t AudioFlinger::RecordThread::RecordTrack::setPreferredMicrophoneDirection(
+status_t RecordTrack::setPreferredMicrophoneDirection(
         audio_microphone_direction_t direction) {
-    sp<ThreadBase> thread = mThread.promote();
+    sp<AudioFlinger::ThreadBase> thread = mThread.promote();
     if (thread != 0) {
-        RecordThread *recordThread = (RecordThread *)thread.get();
+        auto* const recordThread = static_cast<AudioFlinger::RecordThread*>(thread.get());
         return recordThread->setPreferredMicrophoneDirection(direction);
     } else {
         return BAD_VALUE;
     }
 }
 
-status_t AudioFlinger::RecordThread::RecordTrack::setPreferredMicrophoneFieldDimension(float zoom) {
-    sp<ThreadBase> thread = mThread.promote();
+status_t RecordTrack::setPreferredMicrophoneFieldDimension(float zoom) {
+    sp<AudioFlinger::ThreadBase> thread = mThread.promote();
     if (thread != 0) {
-        RecordThread *recordThread = (RecordThread *)thread.get();
+        auto* const recordThread = static_cast<AudioFlinger::RecordThread*>(thread.get());
         return recordThread->setPreferredMicrophoneFieldDimension(zoom);
     } else {
         return BAD_VALUE;
     }
 }
 
-status_t AudioFlinger::RecordThread::RecordTrack::shareAudioHistory(
+status_t RecordTrack::shareAudioHistory(
         const std::string& sharedAudioPackageName, int64_t sharedAudioStartMs) {
 
     const uid_t callingUid = IPCThreadState::self()->getCallingUid();
@@ -2905,9 +3045,9 @@
         return PERMISSION_DENIED;
     }
 
-    sp<ThreadBase> thread = mThread.promote();
+    sp<AudioFlinger::ThreadBase> thread = mThread.promote();
     if (thread != 0) {
-        RecordThread *recordThread = (RecordThread *)thread.get();
+        auto* const recordThread = static_cast<AudioFlinger::RecordThread*>(thread.get());
         status_t status = recordThread->shareAudioHistory(
                 sharedAudioPackageName, mSessionId, sharedAudioStartMs);
         if (status == NO_ERROR) {
@@ -2919,7 +3059,7 @@
     }
 }
 
-void AudioFlinger::RecordThread::RecordTrack::copyMetadataTo(MetadataInserter& backInserter) const
+void RecordTrack::copyMetadataTo(MetadataInserter& backInserter) const
 {
 
     // Do not forward PatchRecord metadata with unspecified audio source
@@ -2943,7 +3083,33 @@
 #undef LOG_TAG
 #define LOG_TAG "AF::PatchRecord"
 
-AudioFlinger::RecordThread::PatchRecord::PatchRecord(RecordThread *recordThread,
+/* static */
+sp<IAfPatchRecord> IAfPatchRecord::create(
+        void* /* RecordThread */ recordThread, // TODO(b/288339104)
+        uint32_t sampleRate,
+        audio_channel_mask_t channelMask,
+        audio_format_t format,
+        size_t frameCount,
+        void *buffer,
+        size_t bufferSize,
+        audio_input_flags_t flags,
+        const Timeout& timeout,
+        audio_source_t source)
+{
+    return sp<PatchRecord>::make(
+            reinterpret_cast<AudioFlinger::RecordThread*>(recordThread),
+            sampleRate,
+            channelMask,
+            format,
+            frameCount,
+            buffer,
+            bufferSize,
+            flags,
+            timeout,
+            source);
+}
+
+PatchRecord::PatchRecord(AudioFlinger::RecordThread *recordThread,
                                                      uint32_t sampleRate,
                                                      audio_channel_mask_t channelMask,
                                                      audio_format_t format,
@@ -2968,7 +3134,7 @@
                                       (int)(mPeerTimeout.tv_nsec / 1000000));
 }
 
-AudioFlinger::RecordThread::PatchRecord::~PatchRecord()
+PatchRecord::~PatchRecord()
 {
     ALOGV("%s(%d)", __func__, mId);
 }
@@ -2992,7 +3158,7 @@
 }
 
 // static
-size_t AudioFlinger::RecordThread::PatchRecord::writeFrames(
+size_t PatchRecord::writeFrames(
         AudioBufferProvider* dest, const void* src, size_t frameCount, size_t frameSize)
 {
     size_t framesWritten = writeFramesHelper(dest, src, frameCount, frameSize);
@@ -3007,7 +3173,7 @@
 }
 
 // AudioBufferProvider interface
-status_t AudioFlinger::RecordThread::PatchRecord::getNextBuffer(
+status_t PatchRecord::getNextBuffer(
                                                   AudioBufferProvider::Buffer* buffer)
 {
     ALOG_ASSERT(mPeerProxy != 0, "%s(%d): called without peer proxy", __func__, mId);
@@ -3029,7 +3195,7 @@
     return status;
 }
 
-void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+void PatchRecord::releaseBuffer(AudioBufferProvider::Buffer* buffer)
 {
     ALOG_ASSERT(mPeerProxy != 0, "%s(%d): called without peer proxy", __func__, mId);
     Proxy::Buffer buf;
@@ -3039,13 +3205,13 @@
     TrackBase::releaseBuffer(buffer);
 }
 
-status_t AudioFlinger::RecordThread::PatchRecord::obtainBuffer(Proxy::Buffer* buffer,
+status_t PatchRecord::obtainBuffer(Proxy::Buffer* buffer,
                                                                const struct timespec *timeOut)
 {
     return mProxy->obtainBuffer(buffer, timeOut);
 }
 
-void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(Proxy::Buffer* buffer)
+void PatchRecord::releaseBuffer(Proxy::Buffer* buffer)
 {
     mProxy->releaseBuffer(buffer);
 }
@@ -3060,8 +3226,28 @@
     return {ptr, free};
 }
 
-AudioFlinger::RecordThread::PassthruPatchRecord::PassthruPatchRecord(
-        RecordThread *recordThread,
+/* static */
+sp<IAfPatchRecord> IAfPatchRecord::createPassThru(
+        void* /* RecordThread */ recordThread, // TODO(b/288339104)
+        uint32_t sampleRate,
+        audio_channel_mask_t channelMask,
+        audio_format_t format,
+        size_t frameCount,
+        audio_input_flags_t flags,
+        audio_source_t source)
+{
+    return sp<PassthruPatchRecord>::make(
+            reinterpret_cast<AudioFlinger::RecordThread*>(recordThread),
+            sampleRate,
+            channelMask,
+            format,
+            frameCount,
+            flags,
+            source);
+}
+
+PassthruPatchRecord::PassthruPatchRecord(
+        AudioFlinger::RecordThread* recordThread,
         uint32_t sampleRate,
         audio_channel_mask_t channelMask,
         audio_format_t format,
@@ -3077,18 +3263,18 @@
     memset(mStubBuffer.get(), 0, mFrameCount * mFrameSize);
 }
 
-sp<StreamInHalInterface> AudioFlinger::RecordThread::PassthruPatchRecord::obtainStream(
-        sp<ThreadBase>* thread)
+sp<StreamInHalInterface> PassthruPatchRecord::obtainStream(
+        sp<AudioFlinger::ThreadBase>* thread)
 {
     *thread = mThread.promote();
     if (!*thread) return nullptr;
-    RecordThread *recordThread = static_cast<RecordThread*>((*thread).get());
+    auto* const recordThread = static_cast<AudioFlinger::RecordThread*>((*thread).get());
     Mutex::Autolock _l(recordThread->mLock);
     return recordThread->mInput ? recordThread->mInput->stream : nullptr;
 }
 
 // PatchProxyBufferProvider methods are called on DirectOutputThread
-status_t AudioFlinger::RecordThread::PassthruPatchRecord::obtainBuffer(
+status_t PassthruPatchRecord::obtainBuffer(
         Proxy::Buffer* buffer, const struct timespec* timeOut)
 {
     if (mUnconsumedFrames) {
@@ -3106,7 +3292,7 @@
     const size_t framesToRead = std::min(buffer->mFrameCount, mFrameCount);
     buffer->mFrameCount = 0;
     buffer->mRaw = nullptr;
-    sp<ThreadBase> thread;
+    sp<AudioFlinger::ThreadBase> thread;
     sp<StreamInHalInterface> stream = obtainStream(&thread);
     if (!stream) return NO_INIT;  // If there is no stream, RecordThread is not reading.
 
@@ -3154,7 +3340,7 @@
     return result;
 }
 
-void AudioFlinger::RecordThread::PassthruPatchRecord::releaseBuffer(Proxy::Buffer* buffer)
+void PassthruPatchRecord::releaseBuffer(Proxy::Buffer* buffer)
 {
     if (buffer->mFrameCount <= mUnconsumedFrames) {
         mUnconsumedFrames -= buffer->mFrameCount;
@@ -3171,7 +3357,7 @@
 // and 'releaseBuffer' are stubbed out and ignore their input.
 // It's not possible to retrieve actual data here w/o blocking 'obtainBuffer'
 // until we copy it.
-status_t AudioFlinger::RecordThread::PassthruPatchRecord::read(
+status_t PassthruPatchRecord::read(
         void* buffer, size_t bytes, size_t* read)
 {
     bytes = std::min(bytes, mFrameCount * mFrameSize);
@@ -3190,15 +3376,15 @@
     return 0;
 }
 
-status_t AudioFlinger::RecordThread::PassthruPatchRecord::getCapturePosition(
+status_t PassthruPatchRecord::getCapturePosition(
         int64_t* frames, int64_t* time)
 {
-    sp<ThreadBase> thread;
+    sp<AudioFlinger::ThreadBase> thread;
     sp<StreamInHalInterface> stream = obtainStream(&thread);
     return stream ? stream->getCapturePosition(frames, time) : NO_INIT;
 }
 
-status_t AudioFlinger::RecordThread::PassthruPatchRecord::standby()
+status_t PassthruPatchRecord::standby()
 {
     // RecordThread issues 'standby' command in two major cases:
     // 1. Error on read--this case is handled in 'obtainBuffer'.
@@ -3210,7 +3396,7 @@
 }
 
 // As the buffer gets filled in obtainBuffer, here we only simulate data consumption.
-status_t AudioFlinger::RecordThread::PassthruPatchRecord::getNextBuffer(
+status_t PassthruPatchRecord::getNextBuffer(
         AudioBufferProvider::Buffer* buffer)
 {
     buffer->frameCount = mLastReadFrames;
@@ -3218,7 +3404,7 @@
     return NO_ERROR;
 }
 
-void AudioFlinger::RecordThread::PassthruPatchRecord::releaseBuffer(
+void PassthruPatchRecord::releaseBuffer(
         AudioBufferProvider::Buffer* buffer)
 {
     buffer->frameCount = 0;
@@ -3229,7 +3415,32 @@
 #undef LOG_TAG
 #define LOG_TAG "AF::MmapTrack"
 
-AudioFlinger::MmapThread::MmapTrack::MmapTrack(ThreadBase *thread,
+/* static */
+sp<IAfMmapTrack> IAfMmapTrack::create(void* /* AudioFlinger::ThreadBase */ thread,
+          const audio_attributes_t& attr,
+          uint32_t sampleRate,
+          audio_format_t format,
+          audio_channel_mask_t channelMask,
+          audio_session_t sessionId,
+          bool isOut,
+          const android::content::AttributionSourceState& attributionSource,
+          pid_t creatorPid,
+          audio_port_handle_t portId)
+{
+    return sp<MmapTrack>::make(
+            reinterpret_cast<AudioFlinger::ThreadBase*>(thread),
+            attr,
+            sampleRate,
+            format,
+            channelMask,
+            sessionId,
+            isOut,
+            attributionSource,
+            creatorPid,
+            portId);
+}
+
+MmapTrack::MmapTrack(AudioFlinger::ThreadBase* thread,
         const audio_attributes_t& attr,
         uint32_t sampleRate,
         audio_format_t format,
@@ -3255,27 +3466,27 @@
     mTrackMetrics.logConstructor(creatorPid, uid(), id());
 }
 
-AudioFlinger::MmapThread::MmapTrack::~MmapTrack()
+MmapTrack::~MmapTrack()
 {
 }
 
-status_t AudioFlinger::MmapThread::MmapTrack::initCheck() const
+status_t MmapTrack::initCheck() const
 {
     return NO_ERROR;
 }
 
-status_t AudioFlinger::MmapThread::MmapTrack::start(AudioSystem::sync_event_t event __unused,
+status_t MmapTrack::start(AudioSystem::sync_event_t event __unused,
                                                     audio_session_t triggerSession __unused)
 {
     return NO_ERROR;
 }
 
-void AudioFlinger::MmapThread::MmapTrack::stop()
+void MmapTrack::stop()
 {
 }
 
 // AudioBufferProvider interface
-status_t AudioFlinger::MmapThread::MmapTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+status_t MmapTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
 {
     buffer->frameCount = 0;
     buffer->raw = nullptr;
@@ -3283,21 +3494,20 @@
 }
 
 // ExtendedAudioBufferProvider interface
-size_t AudioFlinger::MmapThread::MmapTrack::framesReady() const {
+size_t MmapTrack::framesReady() const {
     return 0;
 }
 
-int64_t AudioFlinger::MmapThread::MmapTrack::framesReleased() const
+int64_t MmapTrack::framesReleased() const
 {
     return 0;
 }
 
-void AudioFlinger::MmapThread::MmapTrack::onTimestamp(const ExtendedTimestamp &timestamp __unused)
+void MmapTrack::onTimestamp(const ExtendedTimestamp& timestamp __unused)
 {
 }
 
-void AudioFlinger::MmapThread::MmapTrack::processMuteEvent_l(const sp<
-    IAudioManager>& audioManager, mute_state_t muteState)
+void MmapTrack::processMuteEvent_l(const sp<IAudioManager>& audioManager, mute_state_t muteState)
 {
     if (mMuteState == muteState) {
         // mute state did not change, do nothing
@@ -3328,13 +3538,13 @@
     }
 }
 
-void AudioFlinger::MmapThread::MmapTrack::appendDumpHeader(String8& result) const
+void 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) const
+void MmapTrack::appendDump(String8& result, bool active __unused) const
 {
     result.appendFormat("%6u %7u %7u %08X %08X %6u 0x%03X ",
             mPid,