diff --git a/camera/Android.bp b/camera/Android.bp
index b3f70f4..a3fd7f9 100644
--- a/camera/Android.bp
+++ b/camera/Android.bp
@@ -144,6 +144,7 @@
     srcs: [
         "aidl/android/hardware/CameraExtensionSessionStats.aidl",
         "aidl/android/hardware/ICameraService.aidl",
+        "aidl/android/hardware/CameraIdRemapping.aidl",
         "aidl/android/hardware/ICameraServiceListener.aidl",
         "aidl/android/hardware/ICameraServiceProxy.aidl",
         "aidl/android/hardware/camera2/ICameraDeviceCallbacks.aidl",
diff --git a/camera/aidl/android/hardware/CameraIdRemapping.aidl b/camera/aidl/android/hardware/CameraIdRemapping.aidl
new file mode 100644
index 0000000..e875c53
--- /dev/null
+++ b/camera/aidl/android/hardware/CameraIdRemapping.aidl
@@ -0,0 +1,50 @@
+/*
+ * 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.
+ */
+
+package android.hardware;
+
+/**
+ * Specifies a remapping of Camera Ids.
+ *
+ * Example: For a given package, a remapping of camera id0 to id1 specifies
+ * that any operation to perform on id0 should instead be performed on id1.
+ *
+ * @hide
+ */
+parcelable CameraIdRemapping {
+    /**
+     * Specifies remapping of Camera Ids per package.
+     */
+    parcelable PackageIdRemapping {
+        /** Package Name (e.g. com.android.xyz). */
+        String packageName;
+        /**
+         * Ordered list of Camera Ids to replace. Only Camera Ids present in this list will be
+         * affected.
+         */
+        List<String> cameraIdsToReplace;
+        /**
+         *  Ordered list of updated Camera Ids, where updatedCameraIds[i] corresponds to
+         *  the updated camera id for cameraIdsToReplace[i].
+         */
+        List<String> updatedCameraIds;
+    }
+
+    /**
+     * List of Camera Id remappings to perform.
+     */
+    List<PackageIdRemapping> packageIdRemappings;
+}
diff --git a/camera/aidl/android/hardware/ICameraService.aidl b/camera/aidl/android/hardware/ICameraService.aidl
index f8e1631..01b8ff8 100644
--- a/camera/aidl/android/hardware/ICameraService.aidl
+++ b/camera/aidl/android/hardware/ICameraService.aidl
@@ -29,6 +29,7 @@
 import android.hardware.camera2.impl.CameraMetadataNative;
 import android.hardware.ICameraServiceListener;
 import android.hardware.CameraInfo;
+import android.hardware.CameraIdRemapping;
 import android.hardware.CameraStatus;
 import android.hardware.CameraExtensionSessionStats;
 
@@ -131,6 +132,22 @@
             int targetSdkVersion);
 
     /**
+     * Remap Camera Ids in the CameraService.
+     *
+     * Once this is in effect, all binder calls in the ICameraService that
+     * use logicalCameraId should consult remapping state to arrive at the
+     * correct cameraId to perform the operation on.
+     *
+     * Note: Before the new cameraIdRemapping state is applied, the previous
+     * state is cleared.
+     *
+     * @param cameraIdRemapping the camera ids to remap. Sending an unpopulated
+     *        cameraIdRemapping object will result in clearing of any previous
+     *        cameraIdRemapping state in the camera service.
+     */
+    void remapCameraIds(in CameraIdRemapping cameraIdRemapping);
+
+    /**
      * Remove listener for changes to camera device and flashlight state.
      */
     void removeListener(ICameraServiceListener listener);
diff --git a/media/audioserver/main_audioserver.cpp b/media/audioserver/main_audioserver.cpp
index 1e3bfe0..c7a1bfd 100644
--- a/media/audioserver/main_audioserver.cpp
+++ b/media/audioserver/main_audioserver.cpp
@@ -184,7 +184,7 @@
         // attempting to call audio flinger on a null pointer could make the process crash
         // and attract attentions.
         std::vector<AudioMMapPolicyInfo> policyInfos;
-        status_t status = af->getMmapPolicyInfos(
+        status_t status = sp<IAudioFlinger>::cast(af)->getMmapPolicyInfos(
                 AudioMMapPolicyType::DEFAULT, &policyInfos);
         // Initialize aaudio service when querying mmap policy succeeds and
         // any of the policy supports MMAP.
diff --git a/media/codec2/sfplugin/CCodec.cpp b/media/codec2/sfplugin/CCodec.cpp
index 86fd8ab..a75ce70 100644
--- a/media/codec2/sfplugin/CCodec.cpp
+++ b/media/codec2/sfplugin/CCodec.cpp
@@ -46,6 +46,7 @@
 #include <media/stagefright/BufferProducerWrapper.h>
 #include <media/stagefright/MediaCodecConstants.h>
 #include <media/stagefright/PersistentSurface.h>
+#include <media/stagefright/RenderedFrameInfo.h>
 #include <utils/NativeHandle.h>
 
 #include "C2OMXNode.h"
@@ -672,8 +673,7 @@
     }
 
     void onOutputFramesRendered(int64_t mediaTimeUs, nsecs_t renderTimeNs) override {
-        mCodec->mCallback->onOutputFramesRendered(
-                {RenderedFrameInfo(mediaTimeUs, renderTimeNs)});
+        mCodec->mCallback->onOutputFramesRendered({RenderedFrameInfo(mediaTimeUs, renderTimeNs)});
     }
 
     void onOutputBuffersChanged() override {
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index 7caaaaf..4bd12b8 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -646,7 +646,7 @@
     return result.value_or(0);
 }
 
-uint32_t AudioFlingerClientAdapter::getPrimaryOutputSamplingRate() {
+uint32_t AudioFlingerClientAdapter::getPrimaryOutputSamplingRate() const {
     auto result = [&]() -> ConversionResult<uint32_t> {
         int32_t aidlRet;
         RETURN_IF_ERROR(statusTFromBinderStatus(
@@ -657,7 +657,7 @@
     return result.value_or(0);
 }
 
-size_t AudioFlingerClientAdapter::getPrimaryOutputFrameCount() {
+size_t AudioFlingerClientAdapter::getPrimaryOutputFrameCount() const {
     auto result = [&]() -> ConversionResult<size_t> {
         int64_t aidlRet;
         RETURN_IF_ERROR(statusTFromBinderStatus(
@@ -672,7 +672,7 @@
     return statusTFromBinderStatus(mDelegate->setLowRamDevice(isLowRamDevice, totalMemory));
 }
 
-status_t AudioFlingerClientAdapter::getAudioPort(struct audio_port_v7* port) {
+status_t AudioFlingerClientAdapter::getAudioPort(struct audio_port_v7* port) const {
     media::AudioPortFw portAidl = VALUE_OR_RETURN_STATUS(
             legacy2aidl_audio_port_v7_AudioPortFw(*port));
     media::AudioPortFw aidlRet;
@@ -705,7 +705,7 @@
 }
 
 status_t AudioFlingerClientAdapter::listAudioPatches(unsigned int* num_patches,
-                                                     struct audio_patch* patches) {
+                                                     struct audio_patch* patches) const {
     std::vector<media::AudioPatchFw> aidlRet;
     int32_t maxPatches = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(*num_patches));
     RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
@@ -753,7 +753,8 @@
 }
 
 status_t
-AudioFlingerClientAdapter::getMicrophones(std::vector<media::MicrophoneInfoFw>* microphones) {
+AudioFlingerClientAdapter::getMicrophones(
+        std::vector<media::MicrophoneInfoFw>* microphones) const {
     std::vector<media::MicrophoneInfoFw> aidlRet;
     RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(mDelegate->getMicrophones(&aidlRet)));
     if (microphones != nullptr) {
@@ -788,7 +789,7 @@
     return statusTFromBinderStatus(mDelegate->getMmapPolicyInfos(policyType, policyInfos));
 }
 
-int32_t AudioFlingerClientAdapter::getAAudioMixerBurstCount() {
+int32_t AudioFlingerClientAdapter::getAAudioMixerBurstCount() const {
     auto result = [&]() -> ConversionResult<int32_t> {
         int32_t aidlRet;
         RETURN_IF_ERROR(statusTFromBinderStatus(mDelegate->getAAudioMixerBurstCount(&aidlRet)));
@@ -798,7 +799,7 @@
     return result.value_or(0);
 }
 
-int32_t AudioFlingerClientAdapter::getAAudioHardwareBurstMinUsec() {
+int32_t AudioFlingerClientAdapter::getAAudioHardwareBurstMinUsec() const {
     auto result = [&]() -> ConversionResult<int32_t> {
         int32_t aidlRet;
         RETURN_IF_ERROR(statusTFromBinderStatus(
@@ -829,7 +830,7 @@
 }
 
 status_t AudioFlingerClientAdapter::getSupportedLatencyModes(
-        audio_io_handle_t output, std::vector<audio_latency_mode_t>* modes) {
+        audio_io_handle_t output, std::vector<audio_latency_mode_t>* modes) const {
     if (modes == nullptr) {
         return BAD_VALUE;
     }
@@ -851,7 +852,7 @@
     return statusTFromBinderStatus(mDelegate->setBluetoothVariableLatencyEnabled(enabled));
 }
 
-status_t AudioFlingerClientAdapter::isBluetoothVariableLatencyEnabled(bool* enabled) {
+status_t AudioFlingerClientAdapter::isBluetoothVariableLatencyEnabled(bool* enabled) const {
     if (enabled == nullptr) {
         return BAD_VALUE;
     }
@@ -862,7 +863,7 @@
     return NO_ERROR;
 }
 
-status_t AudioFlingerClientAdapter::supportsBluetoothVariableLatency(bool* support) {
+status_t AudioFlingerClientAdapter::supportsBluetoothVariableLatency(bool* support) const {
     if (support == nullptr) {
         return BAD_VALUE;
     }
@@ -875,7 +876,7 @@
 
 status_t AudioFlingerClientAdapter::getSoundDoseInterface(
         const sp<media::ISoundDoseCallback> &callback,
-        sp<media::ISoundDose>* soundDose) {
+        sp<media::ISoundDose>* soundDose) const {
     return statusTFromBinderStatus(mDelegate->getSoundDoseInterface(callback, soundDose));
 }
 
diff --git a/media/libaudioclient/aidl/android/media/ISoundDose.aidl b/media/libaudioclient/aidl/android/media/ISoundDose.aidl
index 6cb22ef..d80b6bf 100644
--- a/media/libaudioclient/aidl/android/media/ISoundDose.aidl
+++ b/media/libaudioclient/aidl/android/media/ISoundDose.aidl
@@ -55,6 +55,30 @@
      */
     oneway void setCsdEnabled(boolean enabled);
 
+    /**
+     * Structure containing a device identifier by address and type together with
+     * the categorization whether it is a headphone or not.
+     */
+    @JavaDerive(toString = true)
+    parcelable AudioDeviceCategory {
+        @utf8InCpp String address;
+        int internalAudioType;
+        boolean csdCompatible;
+    }
+
+    /**
+     * Resets the list of stored device categories for the native layer. Should
+     * only be called once at boot time after parsing the existing AudioDeviceCategories.
+     */
+    oneway void initCachedAudioDeviceCategories(in AudioDeviceCategory[] audioDevices);
+
+    /**
+     * Sets whether a device for a given address and type is a headphone or not.
+     * This is used to determine whether we compute the CSD on the given device
+     * since we can not rely completely on the device annotations.
+     */
+    oneway void setAudioDeviceCategory(in AudioDeviceCategory audioDevice);
+
     /* -------------------------- Test API methods --------------------------
     /** Get the currently used RS2 upper bound. */
     float getOutputRs2UpperBound();
diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index 2e2ef65..3c96862 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -303,8 +303,8 @@
     // helpers for android.media.AudioManager.getProperty(), see description there for meaning
     // FIXME move these APIs to AudioPolicy to permit a more accurate implementation
     // that looks on primary device for a stream with fast flag, primary flag, or first one.
-    virtual uint32_t getPrimaryOutputSamplingRate() = 0;
-    virtual size_t getPrimaryOutputFrameCount() = 0;
+    virtual uint32_t getPrimaryOutputSamplingRate() const = 0;
+    virtual size_t getPrimaryOutputFrameCount() const = 0;
 
     // Intended for AudioService to inform AudioFlinger of device's low RAM attribute,
     // and should be called at most once.  For a definition of what "low RAM" means, see
@@ -313,7 +313,7 @@
     virtual status_t setLowRamDevice(bool isLowRamDevice, int64_t totalMemory) = 0;
 
     /* Get attributes for a given audio port */
-    virtual status_t getAudioPort(struct audio_port_v7 *port) = 0;
+    virtual status_t getAudioPort(struct audio_port_v7* port) const = 0;
 
     /* Create an audio patch between several source and sink ports */
     virtual status_t createAudioPatch(const struct audio_patch *patch,
@@ -324,7 +324,7 @@
 
     /* List existing audio patches */
     virtual status_t listAudioPatches(unsigned int *num_patches,
-                                      struct audio_patch *patches) = 0;
+                                      struct audio_patch* patches) const = 0;
     /* Set audio port configuration */
     virtual status_t setAudioPortConfig(const struct audio_port_config *config) = 0;
 
@@ -341,7 +341,7 @@
     virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const = 0;
 
     /* List available microphones and their characteristics */
-    virtual status_t getMicrophones(std::vector<media::MicrophoneInfoFw> *microphones) = 0;
+    virtual status_t getMicrophones(std::vector<media::MicrophoneInfoFw>* microphones) const = 0;
 
     virtual status_t setAudioHalPids(const std::vector<pid_t>& pids) = 0;
 
@@ -357,9 +357,9 @@
             media::audio::common::AudioMMapPolicyType policyType,
             std::vector<media::audio::common::AudioMMapPolicyInfo> *policyInfos) = 0;
 
-    virtual int32_t getAAudioMixerBurstCount() = 0;
+    virtual int32_t getAAudioMixerBurstCount() const = 0;
 
-    virtual int32_t getAAudioHardwareBurstMinUsec() = 0;
+    virtual int32_t getAAudioHardwareBurstMinUsec() const = 0;
 
     virtual status_t setDeviceConnectedState(const struct audio_port_v7 *port,
                                              media::DeviceConnectedState state) = 0;
@@ -370,18 +370,18 @@
             audio_io_handle_t output, audio_latency_mode_t mode) = 0;
 
     virtual status_t getSupportedLatencyModes(audio_io_handle_t output,
-            std::vector<audio_latency_mode_t>* modes) = 0;
+            std::vector<audio_latency_mode_t>* modes) const = 0;
 
     virtual status_t getSoundDoseInterface(const sp<media::ISoundDoseCallback>& callback,
-                                           sp<media::ISoundDose>* soundDose) = 0;
+                                           sp<media::ISoundDose>* soundDose) const = 0;
 
     virtual status_t invalidateTracks(const std::vector<audio_port_handle_t>& portIds) = 0;
 
     virtual status_t setBluetoothVariableLatencyEnabled(bool enabled) = 0;
 
-    virtual status_t isBluetoothVariableLatencyEnabled(bool* enabled) = 0;
+    virtual status_t isBluetoothVariableLatencyEnabled(bool* enabled) const = 0;
 
-    virtual status_t supportsBluetoothVariableLatency(bool* support) = 0;
+    virtual status_t supportsBluetoothVariableLatency(bool* support) const = 0;
 
     virtual status_t getAudioPolicyConfig(media::AudioPolicyConfig* output) = 0;
 };
@@ -459,22 +459,22 @@
                             audio_session_t sessionId,
                             bool suspended) override;
     audio_module_handle_t loadHwModule(const char* name) override;
-    uint32_t getPrimaryOutputSamplingRate() override;
-    size_t getPrimaryOutputFrameCount() override;
+    uint32_t getPrimaryOutputSamplingRate() const override;
+    size_t getPrimaryOutputFrameCount() const override;
     status_t setLowRamDevice(bool isLowRamDevice, int64_t totalMemory) override;
-    status_t getAudioPort(struct audio_port_v7* port) override;
+    status_t getAudioPort(struct audio_port_v7* port) const override;
     status_t createAudioPatch(const struct audio_patch* patch,
                               audio_patch_handle_t* handle) override;
     status_t releaseAudioPatch(audio_patch_handle_t handle) override;
     status_t listAudioPatches(unsigned int* num_patches,
-                              struct audio_patch* patches) override;
+                              struct audio_patch* patches) const override;
     status_t setAudioPortConfig(const struct audio_port_config* config) override;
     audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId) override;
     status_t systemReady() override;
     status_t audioPolicyReady() override;
 
     size_t frameCountHAL(audio_io_handle_t ioHandle) const override;
-    status_t getMicrophones(std::vector<media::MicrophoneInfoFw>* microphones) override;
+    status_t getMicrophones(std::vector<media::MicrophoneInfoFw>* microphones) const override;
     status_t setAudioHalPids(const std::vector<pid_t>& pids) override;
     status_t setVibratorInfos(const std::vector<media::AudioVibratorInfo>& vibratorInfos) override;
     status_t updateSecondaryOutputs(
@@ -482,20 +482,20 @@
     status_t getMmapPolicyInfos(
             media::audio::common::AudioMMapPolicyType policyType,
             std::vector<media::audio::common::AudioMMapPolicyInfo> *policyInfos) override;
-    int32_t getAAudioMixerBurstCount() override;
-    int32_t getAAudioHardwareBurstMinUsec() override;
+    int32_t getAAudioMixerBurstCount() const override;
+    int32_t getAAudioHardwareBurstMinUsec() const override;
     status_t setDeviceConnectedState(const struct audio_port_v7 *port,
                                      media::DeviceConnectedState state) override;
     status_t setSimulateDeviceConnections(bool enabled) override;
     status_t setRequestedLatencyMode(audio_io_handle_t output,
             audio_latency_mode_t mode) override;
     status_t getSupportedLatencyModes(
-            audio_io_handle_t output, std::vector<audio_latency_mode_t>* modes) override;
+            audio_io_handle_t output, std::vector<audio_latency_mode_t>* modes) const override;
     status_t setBluetoothVariableLatencyEnabled(bool enabled) override;
-    status_t isBluetoothVariableLatencyEnabled(bool* enabled) override;
-    status_t supportsBluetoothVariableLatency(bool* support) override;
+    status_t isBluetoothVariableLatencyEnabled(bool* enabled) const override;
+    status_t supportsBluetoothVariableLatency(bool* support) const override;
     status_t getSoundDoseInterface(const sp<media::ISoundDoseCallback>& callback,
-                                   sp<media::ISoundDose>* soundDose) override;
+                                   sp<media::ISoundDose>* soundDose) const override;
     status_t invalidateTracks(const std::vector<audio_port_handle_t>& portIds) override;
     status_t getAudioPolicyConfig(media::AudioPolicyConfig* output) override;
 
diff --git a/media/libeffects/visualizer/aidl/VisualizerContext.cpp b/media/libeffects/visualizer/aidl/VisualizerContext.cpp
index 5d0d08d..a1726ad 100644
--- a/media/libeffects/visualizer/aidl/VisualizerContext.cpp
+++ b/media/libeffects/visualizer/aidl/VisualizerContext.cpp
@@ -223,8 +223,7 @@
         deltaSamples = kMaxCaptureBufSize;
     }
 
-    int32_t capturePoint;
-    //capturePoint = (int32_t)mCaptureIdx - deltaSamples;
+    int32_t capturePoint, captureSamples = mCaptureSamples;
     __builtin_sub_overflow((int32_t) mCaptureIdx, deltaSamples, &capturePoint);
     // a negative capturePoint means we wrap the buffer.
     if (capturePoint < 0) {
@@ -232,13 +231,14 @@
         if (size > mCaptureSamples) {
             size = mCaptureSamples;
         }
+        // first part of two stages copy, capture to the end of buffer and reset the size/point
         result.insert(result.end(), &mCaptureBuf[kMaxCaptureBufSize + capturePoint],
                         &mCaptureBuf[kMaxCaptureBufSize + capturePoint + size]);
-        mCaptureSamples -= size;
+        captureSamples -= size;
         capturePoint = 0;
     }
     result.insert(result.end(), &mCaptureBuf[capturePoint],
-                    &mCaptureBuf[capturePoint + mCaptureSamples]);
+                  &mCaptureBuf[capturePoint + captureSamples]);
     mLastCaptureIdx = mCaptureIdx;
     return result;
 }
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index a91b24a..2223f24 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -43,6 +43,7 @@
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/OMXClient.h>
 #include <media/stagefright/PersistentSurface.h>
+#include <media/stagefright/RenderedFrameInfo.h>
 #include <media/stagefright/SurfaceUtils.h>
 #include <media/hardware/HardwareAPI.h>
 #include <media/MediaBufferHolder.h>
@@ -64,11 +65,14 @@
 #include "include/SharedMemoryBuffer.h"
 #include <media/stagefright/omx/OMXUtils.h>
 
+#include <server_configurable_flags/get_flags.h>
+
 namespace android {
 
 typedef hardware::media::omx::V1_0::IGraphicBufferSource HGraphicBufferSource;
 
 using hardware::media::omx::V1_0::Status;
+using server_configurable_flags::GetServerConfigurableFlag;
 
 enum {
     kMaxIndicesToCheck = 32, // used when enumerating supported formats and profiles
@@ -81,6 +85,11 @@
 
 }
 
+static bool areRenderMetricsEnabled() {
+    std::string v = GetServerConfigurableFlag("media_native", "render_metrics_enabled", "false");
+    return v == "true";
+}
+
 // OMX errors are directly mapped into status_t range if
 // there is no corresponding MediaError status code.
 // Use the statusFromOMXError(int32_t omxError) function.
@@ -563,6 +572,9 @@
 ACodec::ACodec()
     : mSampleRate(0),
       mNodeGeneration(0),
+      mAreRenderMetricsEnabled(areRenderMetricsEnabled()),
+      mIsWindowToDisplay(false),
+      mHasPresentFenceTimes(false),
       mUsingNativeWindow(false),
       mNativeWindowUsageBits(0),
       mLastNativeWindowDataSpace(HAL_DATASPACE_UNKNOWN),
@@ -634,7 +646,8 @@
     if (!mBufferChannel) {
         mBufferChannel = std::make_shared<ACodecBufferChannel>(
                 new AMessage(kWhatInputBufferFilled, this),
-                new AMessage(kWhatOutputBufferDrained, this));
+                new AMessage(kWhatOutputBufferDrained, this),
+                new AMessage(kWhatPollForRenderedBuffers, this));
     }
     return mBufferChannel;
 }
@@ -744,6 +757,7 @@
     // if we have not yet started the codec, we can simply set the native window
     if (mBuffers[kPortIndexInput].size() == 0) {
         mNativeWindow = surface;
+        initializeFrameTracking();
         return OK;
     }
 
@@ -852,6 +866,7 @@
 
     mNativeWindow = nativeWindow;
     mNativeWindowUsageBits = usageBits;
+    initializeFrameTracking();
     return OK;
 }
 
@@ -962,7 +977,6 @@
                 BufferInfo info;
                 info.mStatus = BufferInfo::OWNED_BY_US;
                 info.mFenceFd = -1;
-                info.mRenderInfo = NULL;
                 info.mGraphicBuffer = NULL;
                 info.mNewGraphicBuffer = false;
 
@@ -1230,6 +1244,7 @@
 
     *bufferCount = def.nBufferCountActual;
     *bufferSize =  def.nBufferSize;
+    initializeFrameTracking();
     return err;
 }
 
@@ -1268,7 +1283,6 @@
         info.mStatus = BufferInfo::OWNED_BY_US;
         info.mFenceFd = fenceFd;
         info.mIsReadFence = false;
-        info.mRenderInfo = NULL;
         info.mGraphicBuffer = graphicBuffer;
         info.mNewGraphicBuffer = false;
         info.mDequeuedAt = mDequeueCounter;
@@ -1345,7 +1359,6 @@
         BufferInfo info;
         info.mStatus = BufferInfo::OWNED_BY_NATIVE_WINDOW;
         info.mFenceFd = -1;
-        info.mRenderInfo = NULL;
         info.mGraphicBuffer = NULL;
         info.mNewGraphicBuffer = false;
         info.mDequeuedAt = mDequeueCounter;
@@ -1441,42 +1454,6 @@
     return err;
 }
 
-void ACodec::updateRenderInfoForDequeuedBuffer(
-        ANativeWindowBuffer *buf, int fenceFd, BufferInfo *info) {
-
-    info->mRenderInfo =
-        mRenderTracker.updateInfoForDequeuedBuffer(
-                buf, fenceFd, info - &mBuffers[kPortIndexOutput][0]);
-
-    // check for any fences already signaled
-    notifyOfRenderedFrames(false /* dropIncomplete */, info->mRenderInfo);
-}
-
-void ACodec::onFrameRendered(int64_t mediaTimeUs, nsecs_t systemNano) {
-    if (mRenderTracker.onFrameRendered(mediaTimeUs, systemNano) != OK) {
-        mRenderTracker.dumpRenderQueue();
-    }
-}
-
-void ACodec::notifyOfRenderedFrames(bool dropIncomplete, FrameRenderTracker::Info *until) {
-    std::list<FrameRenderTracker::Info> done =
-        mRenderTracker.checkFencesAndGetRenderedFrames(until, dropIncomplete);
-
-    // unlink untracked frames
-    for (std::list<FrameRenderTracker::Info>::const_iterator it = done.cbegin();
-            it != done.cend(); ++it) {
-        ssize_t index = it->getIndex();
-        if (index >= 0 && (size_t)index < mBuffers[kPortIndexOutput].size()) {
-            mBuffers[kPortIndexOutput][index].mRenderInfo = NULL;
-        } else if (index >= 0) {
-            // THIS SHOULD NEVER HAPPEN
-            ALOGE("invalid index %zd in %zu", index, mBuffers[kPortIndexOutput].size());
-        }
-    }
-
-    mCallback->onOutputFramesRendered(done);
-}
-
 void ACodec::onFirstTunnelFrameReady() {
     mCallback->onFirstTunnelFrameReady();
 }
@@ -1531,7 +1508,6 @@
 
                 info->mStatus = BufferInfo::OWNED_BY_US;
                 info->setWriteFence(fenceFd, "dequeueBufferFromNativeWindow");
-                updateRenderInfoForDequeuedBuffer(buf, fenceFd, info);
                 return info;
             }
         }
@@ -1576,18 +1552,96 @@
     oldest->mNewGraphicBuffer = true;
     oldest->mStatus = BufferInfo::OWNED_BY_US;
     oldest->setWriteFence(fenceFd, "dequeueBufferFromNativeWindow for oldest");
-    mRenderTracker.untrackFrame(oldest->mRenderInfo);
-    oldest->mRenderInfo = NULL;
 
     ALOGV("replaced oldest buffer #%u with age %u, graphicBuffer %p",
             (unsigned)(oldest - &mBuffers[kPortIndexOutput][0]),
             mDequeueCounter - oldest->mDequeuedAt,
             oldest->mGraphicBuffer->handle);
-
-    updateRenderInfoForDequeuedBuffer(buf, fenceFd, oldest);
     return oldest;
 }
 
+void ACodec::initializeFrameTracking() {
+    mTrackedFrames.clear();
+
+    int isWindowToDisplay = 0;
+    mNativeWindow->query(mNativeWindow.get(), NATIVE_WINDOW_QUEUES_TO_WINDOW_COMPOSER,
+            &isWindowToDisplay);
+    mIsWindowToDisplay = isWindowToDisplay == 1;
+    // No frame tracking is needed if we're not sending frames to the display
+    if (!mIsWindowToDisplay) {
+        // Return early so we don't call into SurfaceFlinger (requiring permissions)
+        return;
+    }
+
+    int hasPresentFenceTimes = 0;
+    mNativeWindow->query(mNativeWindow.get(), NATIVE_WINDOW_FRAME_TIMESTAMPS_SUPPORTS_PRESENT,
+            &hasPresentFenceTimes);
+    mHasPresentFenceTimes = hasPresentFenceTimes == 1;
+    if (!mHasPresentFenceTimes) {
+        ALOGI("Using latch times for frame rendered signals - present fences not supported");
+    }
+
+    status_t err = native_window_enable_frame_timestamps(mNativeWindow.get(), true);
+    if (err) {
+        ALOGE("Failed to enable frame timestamps (%d)", err);
+    }
+}
+
+void ACodec::trackReleasedFrame(int64_t frameId, int64_t mediaTimeUs, int64_t desiredRenderTimeNs) {
+    // If the render time is earlier than now, then we're suggesting it should be rendered ASAP,
+    // so track the frame as if the desired render time is now.
+    int64_t nowNs = systemTime(SYSTEM_TIME_MONOTONIC);
+    if (desiredRenderTimeNs < nowNs) {
+        desiredRenderTimeNs = nowNs;
+    }
+    // We've just queued a frame to the surface, so keep track of it and later check to see if it is
+    // actually rendered.
+    TrackedFrame frame;
+    frame.id = frameId;
+    frame.mediaTimeUs = mediaTimeUs;
+    frame.desiredRenderTimeNs = desiredRenderTimeNs;
+    mTrackedFrames.push_back(frame);
+}
+
+void ACodec::pollForRenderedFrames() {
+    std::list<RenderedFrameInfo> renderedFrameInfos;
+    // Scan all frames and check to see if the frames that SHOULD have been rendered by now, have,
+    // in fact, been rendered.
+    int64_t nowNs = systemTime(SYSTEM_TIME_MONOTONIC);
+    while (!mTrackedFrames.empty()) {
+        TrackedFrame & frame = mTrackedFrames.front();
+        // Frames that should have been rendered at least 100ms in the past are checked
+        if (frame.desiredRenderTimeNs > nowNs - 100*1000*1000LL) {
+            break;
+        }
+
+        status_t err;
+        nsecs_t latchOrPresentTimeNs = NATIVE_WINDOW_TIMESTAMP_INVALID;
+        err = native_window_get_frame_timestamps(mNativeWindow.get(), frame.id,
+                /* outRequestedPresentTime */ nullptr, /* outAcquireTime */ nullptr,
+                mHasPresentFenceTimes ? nullptr : &latchOrPresentTimeNs, // latch time
+                /* outFirstRefreshStartTime */ nullptr, /* outLastRefreshStartTime */ nullptr,
+                /* outGpuCompositionDoneTime */ nullptr,
+                mHasPresentFenceTimes ? &latchOrPresentTimeNs : nullptr, // display present time,
+                /* outDequeueReadyTime */ nullptr, /* outReleaseTime */ nullptr);
+        if (err) {
+            ALOGE("Failed to get frame timestamps for %lld: %d", (long long) frame.id, err);
+        }
+        // If we don't have a render time by now, then consider the frame as dropped
+        if (latchOrPresentTimeNs != NATIVE_WINDOW_TIMESTAMP_PENDING &&
+            latchOrPresentTimeNs != NATIVE_WINDOW_TIMESTAMP_INVALID) {
+            renderedFrameInfos.push_back(RenderedFrameInfo(frame.mediaTimeUs,
+                                                           latchOrPresentTimeNs));
+        }
+
+        mTrackedFrames.pop_front();
+    }
+
+    if (!renderedFrameInfos.empty()) {
+        mCallback->onOutputFramesRendered(renderedFrameInfos);
+    }
+}
+
 status_t ACodec::freeBuffersOnPort(OMX_U32 portIndex) {
     if (portIndex == kPortIndexInput) {
         mBufferChannel->setInputBufferArray({});
@@ -1663,11 +1717,6 @@
         ::close(info->mFenceFd);
     }
 
-    if (portIndex == kPortIndexOutput) {
-        mRenderTracker.untrackFrame(info->mRenderInfo, i);
-        info->mRenderInfo = NULL;
-    }
-
     // remove buffer even if mOMXNode->freeBuffer fails
     mBuffers[portIndex].erase(mBuffers[portIndex].begin() + i);
     return err;
@@ -6032,22 +6081,10 @@
     sp<RefBase> obj;
     CHECK(msg->findObject("messages", &obj));
     sp<MessageList> msgList = static_cast<MessageList *>(obj.get());
-
-    bool receivedRenderedEvents = false;
     for (std::list<sp<AMessage>>::const_iterator it = msgList->getList().cbegin();
           it != msgList->getList().cend(); ++it) {
         (*it)->setWhat(ACodec::kWhatOMXMessageItem);
         mCodec->handleMessage(*it);
-        int32_t type;
-        CHECK((*it)->findInt32("type", &type));
-        if (type == omx_message::FRAME_RENDERED) {
-            receivedRenderedEvents = true;
-        }
-    }
-
-    if (receivedRenderedEvents) {
-        // NOTE: all buffers are rendered in this case
-        mCodec->notifyOfRenderedFrames();
     }
     return true;
 }
@@ -6609,15 +6646,6 @@
     info->mDequeuedAt = ++mCodec->mDequeueCounter;
     info->mStatus = BufferInfo::OWNED_BY_US;
 
-    if (info->mRenderInfo != NULL) {
-        // The fence for an emptied buffer must have signaled, but there still could be queued
-        // or out-of-order dequeued buffers in the render queue prior to this buffer. Drop these,
-        // as we will soon requeue this buffer to the surface. While in theory we could still keep
-        // track of buffers that are requeued to the surface, it is better to add support to the
-        // buffer-queue to notify us of released buffers and their fences (in the future).
-        mCodec->notifyOfRenderedFrames(true /* dropIncomplete */);
-    }
-
     // byte buffers cannot take fences, so wait for any fence now
     if (mCodec->mNativeWindow == NULL) {
         (void)mCodec->waitForFence(fenceFd, "onOMXFillBufferDone");
@@ -6824,14 +6852,6 @@
             mCodec->mLastHdr10PlusBuffer = hdr10PlusInfo;
         }
 
-        // save buffers sent to the surface so we can get render time when they return
-        int64_t mediaTimeUs = -1;
-        buffer->meta()->findInt64("timeUs", &mediaTimeUs);
-        if (mediaTimeUs >= 0) {
-            mCodec->mRenderTracker.onFrameQueued(
-                    mediaTimeUs, info->mGraphicBuffer, new Fence(::dup(info->mFenceFd)));
-        }
-
         int64_t timestampNs = 0;
         if (!msg->findInt64("timestampNs", &timestampNs)) {
             // use media timestamp if client did not request a specific render timestamp
@@ -6845,11 +6865,25 @@
         err = native_window_set_buffers_timestamp(mCodec->mNativeWindow.get(), timestampNs);
         ALOGW_IF(err != NO_ERROR, "failed to set buffer timestamp: %d", err);
 
+        uint64_t frameId;
+        err = native_window_get_next_frame_id(mCodec->mNativeWindow.get(), &frameId);
+
         info->checkReadFence("onOutputBufferDrained before queueBuffer");
         err = mCodec->mNativeWindow->queueBuffer(
                     mCodec->mNativeWindow.get(), info->mGraphicBuffer.get(), info->mFenceFd);
-        // TODO(b/266211548): Poll the native window for rendered buffers, since when queueing
-        // buffers, the frame event history delta is retrieved.
+
+        int64_t mediaTimeUs = -1;
+        buffer->meta()->findInt64("timeUs", &mediaTimeUs);
+        if (mCodec->mAreRenderMetricsEnabled && mCodec->mIsWindowToDisplay) {
+            mCodec->trackReleasedFrame(frameId, mediaTimeUs, timestampNs);
+            mCodec->pollForRenderedFrames();
+        } else {
+            // When the surface is an intermediate surface, onFrameRendered is triggered immediately
+            // when the frame is queued to the non-display surface
+            mCodec->mCallback->onOutputFramesRendered({RenderedFrameInfo(mediaTimeUs,
+                                                                         timestampNs)});
+        }
+
         info->mFenceFd = -1;
         if (err == OK) {
             info->mStatus = BufferInfo::OWNED_BY_NATIVE_WINDOW;
@@ -7076,7 +7110,6 @@
     ++mCodec->mNodeGeneration;
 
     mCodec->mComponentName = componentName;
-    mCodec->mRenderTracker.setComponentName(componentName);
     mCodec->mFlags = 0;
 
     if (componentName.endsWith(".secure")) {
@@ -7713,7 +7746,6 @@
 
 void ACodec::ExecutingState::stateEntered() {
     ALOGV("[%s] Now Executing", mCodec->mComponentName.c_str());
-    mCodec->mRenderTracker.clear(systemTime(CLOCK_MONOTONIC));
     mCodec->processDeferredMessages();
 }
 
@@ -7824,7 +7856,15 @@
                     mCodec->signalSubmitOutputMetadataBufferIfEOS_workaround();
                 }
             }
-            return true;
+            handled = true;
+            break;
+        }
+
+        case kWhatPollForRenderedBuffers:
+        {
+            mCodec->pollForRenderedFrames();
+            handled = true;
+            break;
         }
 
         default:
@@ -8520,7 +8560,7 @@
 }
 
 bool ACodec::ExecutingState::onOMXFrameRendered(int64_t mediaTimeUs, nsecs_t systemNano) {
-    mCodec->onFrameRendered(mediaTimeUs, systemNano);
+    mCodec->mCallback->onOutputFramesRendered({RenderedFrameInfo(mediaTimeUs, systemNano)});
     return true;
 }
 
@@ -8694,7 +8734,7 @@
 
 bool ACodec::OutputPortSettingsChangedState::onOMXFrameRendered(
         int64_t mediaTimeUs, nsecs_t systemNano) {
-    mCodec->onFrameRendered(mediaTimeUs, systemNano);
+    mCodec->mCallback->onOutputFramesRendered({RenderedFrameInfo(mediaTimeUs, systemNano)});
     return true;
 }
 
@@ -8725,10 +8765,6 @@
                             OMX_CommandPortEnable, kPortIndexOutput);
                 }
 
-                // Clear the RenderQueue in which queued GraphicBuffers hold the
-                // actual buffer references in order to free them early.
-                mCodec->mRenderTracker.clear(systemTime(CLOCK_MONOTONIC));
-
                 if (err == OK) {
                     err = mCodec->allocateBuffersOnPort(kPortIndexOutput);
                     ALOGE_IF(err != OK, "Failed to allocate output port buffers after port "
@@ -9112,8 +9148,6 @@
         // the native window for rendering. Let's get those back as well.
         mCodec->waitUntilAllPossibleNativeWindowBuffersAreReturnedToUs();
 
-        mCodec->mRenderTracker.clear(systemTime(CLOCK_MONOTONIC));
-
         mCodec->mCallback->onFlushCompleted();
 
         mCodec->mPortEOS[kPortIndexInput] =
diff --git a/media/libstagefright/ACodecBufferChannel.cpp b/media/libstagefright/ACodecBufferChannel.cpp
index 8f2bed2..ad42813 100644
--- a/media/libstagefright/ACodecBufferChannel.cpp
+++ b/media/libstagefright/ACodecBufferChannel.cpp
@@ -32,6 +32,7 @@
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/AMessage.h>
 #include <media/stagefright/foundation/AUtils.h>
+#include <media/stagefright/ACodec.h>
 #include <media/stagefright/MediaCodec.h>
 #include <media/MediaCodecBuffer.h>
 #include <system/window.h>
@@ -87,9 +88,11 @@
 }
 
 ACodecBufferChannel::ACodecBufferChannel(
-        const sp<AMessage> &inputBufferFilled, const sp<AMessage> &outputBufferDrained)
+        const sp<AMessage> &inputBufferFilled, const sp<AMessage> &outputBufferDrained,
+        const sp<AMessage> &pollForRenderedBuffers)
     : mInputBufferFilled(inputBufferFilled),
       mOutputBufferDrained(outputBufferDrained),
+      mPollForRenderedBuffers(pollForRenderedBuffers),
       mHeapSeqNum(-1) {
 }
 
@@ -488,7 +491,7 @@
 }
 
 void ACodecBufferChannel::pollForRenderedBuffers() {
-    // TODO(b/266211548): Poll the native window for rendered buffers.
+    mPollForRenderedBuffers->post();
 }
 
 status_t ACodecBufferChannel::discardBuffer(const sp<MediaCodecBuffer> &buffer) {
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index 91286b9..ea24126 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -79,6 +79,7 @@
 #include <media/stagefright/MediaErrors.h>
 #include <media/stagefright/OMXClient.h>
 #include <media/stagefright/PersistentSurface.h>
+#include <media/stagefright/RenderedFrameInfo.h>
 #include <media/stagefright/SurfaceUtils.h>
 #include <nativeloader/dlext_namespaces.h>
 #include <private/android_filesystem_config.h>
@@ -210,6 +211,7 @@
 // Render metrics
 static const char *kCodecPlaybackDurationSec = "android.media.mediacodec.playback-duration-sec";
 static const char *kCodecFirstRenderTimeUs = "android.media.mediacodec.first-render-time-us";
+static const char *kCodecLastRenderTimeUs = "android.media.mediacodec.last-render-time-us";
 static const char *kCodecFramesReleased = "android.media.mediacodec.frames-released";
 static const char *kCodecFramesRendered = "android.media.mediacodec.frames-rendered";
 static const char *kCodecFramesDropped = "android.media.mediacodec.frames-dropped";
@@ -879,7 +881,7 @@
             const sp<AMessage> &outputFormat) override;
     virtual void onInputSurfaceDeclined(status_t err) override;
     virtual void onSignaledInputEOS(status_t err) override;
-    virtual void onOutputFramesRendered(const std::list<FrameRenderTracker::Info> &done) override;
+    virtual void onOutputFramesRendered(const std::list<RenderedFrameInfo> &done) override;
     virtual void onOutputBuffersChanged() override;
     virtual void onFirstTunnelFrameReady() override;
 private:
@@ -988,7 +990,7 @@
     notify->post();
 }
 
-void CodecCallback::onOutputFramesRendered(const std::list<FrameRenderTracker::Info> &done) {
+void CodecCallback::onOutputFramesRendered(const std::list<RenderedFrameInfo> &done) {
     sp<AMessage> notify(mNotify->dup());
     notify->setInt32("what", kWhatOutputFramesRendered);
     if (MediaCodec::CreateFramesRenderedMessage(done, notify)) {
@@ -1297,6 +1299,7 @@
         const VideoRenderQualityMetrics &m = mVideoRenderQualityTracker.getMetrics();
         if (m.frameReleasedCount > 0) {
             mediametrics_setInt64(mMetricsHandle, kCodecFirstRenderTimeUs, m.firstRenderTimeUs);
+            mediametrics_setInt64(mMetricsHandle, kCodecLastRenderTimeUs, m.lastRenderTimeUs);
             mediametrics_setInt64(mMetricsHandle, kCodecFramesReleased, m.frameReleasedCount);
             mediametrics_setInt64(mMetricsHandle, kCodecFramesRendered, m.frameRenderedCount);
             mediametrics_setInt64(mMetricsHandle, kCodecFramesSkipped, m.frameSkippedCount);
@@ -6087,12 +6090,10 @@
     return onQueueInputBuffer(msg);
 }
 
-//static
-size_t MediaCodec::CreateFramesRenderedMessage(
-        const std::list<FrameRenderTracker::Info> &done, sp<AMessage> &msg) {
+template<typename T>
+static size_t CreateFramesRenderedMessageInternal(const std::list<T> &done, sp<AMessage> &msg) {
     size_t index = 0;
-    for (std::list<FrameRenderTracker::Info>::const_iterator it = done.cbegin();
-            it != done.cend(); ++it) {
+    for (typename std::list<T>::const_iterator it = done.cbegin(); it != done.cend(); ++it) {
         if (it->getRenderTimeNs() < 0) {
             continue; // dropped frame from tracking
         }
@@ -6103,6 +6104,18 @@
     return index;
 }
 
+//static
+size_t MediaCodec::CreateFramesRenderedMessage(
+        const std::list<RenderedFrameInfo> &done, sp<AMessage> &msg) {
+    return CreateFramesRenderedMessageInternal(done, msg);
+}
+
+//static
+size_t MediaCodec::CreateFramesRenderedMessage(
+        const std::list<FrameRenderTracker::Info> &done, sp<AMessage> &msg) {
+    return CreateFramesRenderedMessageInternal(done, msg);
+}
+
 status_t MediaCodec::onReleaseOutputBuffer(const sp<AMessage> &msg) {
     size_t index;
     CHECK(msg->findSize("index", &index));
@@ -6194,7 +6207,9 @@
             // presentation timestamp is used instead, which almost certainly occurs in the past,
             // since it's almost always a zero-based offset from the start of the stream. In these
             // scenarios, we expect the frame to be rendered with no delay.
-            int64_t delayUs = noRenderTime ? 0 : renderTimeNs / 1000 - ALooper::GetNowUs();
+            int64_t nowUs = ALooper::GetNowUs();
+            int64_t renderTimeUs = renderTimeNs / 1000;
+            int64_t delayUs = renderTimeUs < nowUs ? 0 : renderTimeUs - nowUs;
             delayUs += 100 * 1000; /* 100ms in microseconds */
             status_t err =
                     mMsgPollForRenderedBuffers->postUnique(/* token= */ mMsgPollForRenderedBuffers,
diff --git a/media/libstagefright/VideoRenderQualityTracker.cpp b/media/libstagefright/VideoRenderQualityTracker.cpp
index fbd8577..e920bd1 100644
--- a/media/libstagefright/VideoRenderQualityTracker.cpp
+++ b/media/libstagefright/VideoRenderQualityTracker.cpp
@@ -455,6 +455,8 @@
     if (mMetrics.firstRenderTimeUs == 0) {
         mMetrics.firstRenderTimeUs = actualRenderTimeUs;
     }
+    // Capture the timestamp at which the last frame was rendered
+    mMetrics.lastRenderTimeUs = actualRenderTimeUs;
 
     mMetrics.frameRenderedCount++;
 
diff --git a/media/libstagefright/include/ACodecBufferChannel.h b/media/libstagefright/include/ACodecBufferChannel.h
index 903280f..a464504 100644
--- a/media/libstagefright/include/ACodecBufferChannel.h
+++ b/media/libstagefright/include/ACodecBufferChannel.h
@@ -29,6 +29,7 @@
 #include <media/IOMX.h>
 
 namespace android {
+ struct ACodec;
 namespace hardware {
 class HidlMemory;
 };
@@ -63,7 +64,8 @@
     };
 
     ACodecBufferChannel(
-            const sp<AMessage> &inputBufferFilled, const sp<AMessage> &outputBufferDrained);
+            const sp<AMessage> &inputBufferFilled, const sp<AMessage> &outputBufferDrained,
+            const sp<AMessage> &pollForRenderedBuffers);
     virtual ~ACodecBufferChannel();
 
     // BufferChannelBase interface
@@ -138,6 +140,7 @@
 
     const sp<AMessage> mInputBufferFilled;
     const sp<AMessage> mOutputBufferDrained;
+    const sp<AMessage> mPollForRenderedBuffers;
 
     sp<MemoryDealer> mDealer;
     sp<IMemory> mDecryptDestination;
diff --git a/media/libstagefright/include/media/stagefright/ACodec.h b/media/libstagefright/include/media/stagefright/ACodec.h
index e535d5d..f876bc6 100644
--- a/media/libstagefright/include/media/stagefright/ACodec.h
+++ b/media/libstagefright/include/media/stagefright/ACodec.h
@@ -19,7 +19,7 @@
 
 #include <set>
 #include <stdint.h>
-#include <list>
+#include <deque>
 #include <vector>
 #include <android/native_window.h>
 #include <media/hardware/MetadataBufferType.h>
@@ -27,9 +27,9 @@
 #include <media/IOMX.h>
 #include <media/stagefright/AHierarchicalStateMachine.h>
 #include <media/stagefright/CodecBase.h>
-#include <media/stagefright/FrameRenderTracker.h>
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/SkipCutBuffer.h>
+#include <ui/GraphicBuffer.h>
 #include <utils/NativeHandle.h>
 #include <OMX_Audio.h>
 #include <hardware/gralloc.h>
@@ -156,6 +156,7 @@
         kWhatForceStateTransition    = 'fstt',
         kWhatCheckIfStuck            = 'Cstk',
         kWhatSubmitExtraOutputMetadataBuffer = 'sbxo',
+        kWhatPollForRenderedBuffers  = 'pfrb',
     };
 
     enum {
@@ -177,6 +178,13 @@
                             | static_cast<uint64_t>(BufferUsage::VIDEO_DECODER),
     };
 
+    struct TrackedFrame {
+        int64_t id;
+        int64_t mediaTimeUs;
+        int64_t desiredRenderTimeNs;
+        nsecs_t renderTimeNs;
+    };
+
     struct BufferInfo {
         enum Status {
             OWNED_BY_US,
@@ -204,7 +212,6 @@
         sp<GraphicBuffer> mGraphicBuffer;
         bool mNewGraphicBuffer;
         int mFenceFd;
-        FrameRenderTracker::Info *mRenderInfo;
 
         // The following field and 4 methods are used for debugging only
         bool mIsReadFence;
@@ -251,6 +258,11 @@
     int32_t mNodeGeneration;
     sp<TAllocator> mAllocator[2];
 
+    std::deque<TrackedFrame> mTrackedFrames; // render information for buffers sent to a window
+    bool mAreRenderMetricsEnabled;
+    bool mIsWindowToDisplay;
+    bool mHasPresentFenceTimes;
+
     bool mUsingNativeWindow;
     sp<ANativeWindow> mNativeWindow;
     int mNativeWindowUsageBits;
@@ -267,7 +279,6 @@
     // format updates. This will equal to mOutputFormat until the first actual frame is received.
     sp<AMessage> mBaseOutputFormat;
 
-    FrameRenderTracker mRenderTracker; // render information for buffers rendered by ACodec
     std::vector<BufferInfo> mBuffers[2];
     bool mPortEOS[2];
     status_t mInputEOSResult;
@@ -349,6 +360,10 @@
     status_t freeOutputBuffersNotOwnedByComponent();
     BufferInfo *dequeueBufferFromNativeWindow();
 
+    void initializeFrameTracking();
+    void trackReleasedFrame(int64_t frameId, int64_t mediaTimeUs, int64_t desiredRenderTimeNs);
+    void pollForRenderedFrames();
+
     inline bool storingMetadataInDecodedBuffers() {
         return (mPortMode[kPortIndexOutput] == IOMX::kPortModeDynamicANWBuffer) && !mIsEncoder;
     }
@@ -571,21 +586,6 @@
     void processDeferredMessages();
 
     void onFrameRendered(int64_t mediaTimeUs, nsecs_t systemNano);
-    // called when we have dequeued a buffer |buf| from the native window to track render info.
-    // |fenceFd| is the dequeue fence, and |info| points to the buffer info where this buffer is
-    // stored.
-    void updateRenderInfoForDequeuedBuffer(
-            ANativeWindowBuffer *buf, int fenceFd, BufferInfo *info);
-
-    // Checks to see if any frames have rendered up until |until|, and to notify client
-    // (MediaCodec) of rendered frames up-until the frame pointed to by |until| or the first
-    // unrendered frame. These frames are removed from the render queue.
-    // If |dropIncomplete| is true, unrendered frames up-until |until| will be dropped from the
-    // queue, allowing all rendered framed up till then to be notified of.
-    // (This will effectively clear the render queue up-until (and including) |until|.)
-    // If |until| is NULL, or is not in the rendered queue, this method will check all frames.
-    void notifyOfRenderedFrames(
-            bool dropIncomplete = false, FrameRenderTracker::Info *until = NULL);
 
     void onFirstTunnelFrameReady();
 
diff --git a/media/libstagefright/include/media/stagefright/CodecBase.h b/media/libstagefright/include/media/stagefright/CodecBase.h
index 916d41e..90347f9 100644
--- a/media/libstagefright/include/media/stagefright/CodecBase.h
+++ b/media/libstagefright/include/media/stagefright/CodecBase.h
@@ -41,7 +41,7 @@
 struct BufferProducerWrapper;
 class MediaCodecBuffer;
 struct PersistentSurface;
-struct RenderedFrameInfo;
+class RenderedFrameInfo;
 class Surface;
 struct ICrypto;
 class IMemory;
diff --git a/media/libstagefright/include/media/stagefright/FrameRenderTracker.h b/media/libstagefright/include/media/stagefright/FrameRenderTracker.h
index c14755a..cab7ecc 100644
--- a/media/libstagefright/include/media/stagefright/FrameRenderTracker.h
+++ b/media/libstagefright/include/media/stagefright/FrameRenderTracker.h
@@ -32,61 +32,59 @@
 
 namespace android {
 
-// Tracks the render information about a frame. Frames go through several states while
-// the render information is tracked:
-//
-// 1. queued frame: mMediaTime and mGraphicBuffer are set for the frame. mFence is the
-// queue fence (read fence). mIndex is negative, and mRenderTimeNs is invalid.
-// Key characteristics: mFence is not NULL and mIndex is negative.
-//
-// 2. dequeued frame: mFence is updated with the dequeue fence (write fence). mIndex is set.
-// Key characteristics: mFence is not NULL and mIndex is non-negative. mRenderTime is still
-// invalid.
-//
-// 3. rendered frame or frame: mFence is cleared, mRenderTimeNs is set.
-// Key characteristics: mFence is NULL.
-//
-struct RenderedFrameInfo {
-    // set by client during onFrameQueued or onFrameRendered
-    int64_t getMediaTimeUs() const  { return mMediaTimeUs; }
-
-    // -1 if frame is not yet rendered
-    nsecs_t getRenderTimeNs() const { return mRenderTimeNs; }
-
-    // set by client during updateRenderInfoForDequeuedBuffer; -1 otherwise
-    ssize_t getIndex() const        { return mIndex; }
-
-    // creates information for a queued frame
-    RenderedFrameInfo(int64_t mediaTimeUs, const sp<GraphicBuffer> &graphicBuffer,
-            const sp<Fence> &fence)
-        : mMediaTimeUs(mediaTimeUs),
-          mRenderTimeNs(-1),
-          mIndex(-1),
-          mGraphicBuffer(graphicBuffer),
-          mFence(fence) {
-    }
-
-    // creates information for a frame rendered on a tunneled surface
-    RenderedFrameInfo(int64_t mediaTimeUs, nsecs_t renderTimeNs)
-        : mMediaTimeUs(mediaTimeUs),
-          mRenderTimeNs(renderTimeNs),
-          mIndex(-1),
-          mGraphicBuffer(NULL),
-          mFence(NULL) {
-    }
-
-private:
-    int64_t mMediaTimeUs;
-    nsecs_t mRenderTimeNs;
-    ssize_t mIndex;         // to be used by client
-    sp<GraphicBuffer> mGraphicBuffer;
-    sp<Fence> mFence;
-
-    friend struct FrameRenderTracker;
-};
-
 struct FrameRenderTracker {
-    typedef RenderedFrameInfo Info;
+    // Tracks the render information about a frame. Frames go through several states while
+    // the render information is tracked:
+    //
+    // 1. queued frame: mMediaTime and mGraphicBuffer are set for the frame. mFence is the
+    // queue fence (read fence). mIndex is negative, and mRenderTimeNs is invalid.
+    // Key characteristics: mFence is not NULL and mIndex is negative.
+    //
+    // 2. dequeued frame: mFence is updated with the dequeue fence (write fence). mIndex is set.
+    // Key characteristics: mFence is not NULL and mIndex is non-negative. mRenderTime is still
+    // invalid.
+    //
+    // 3. rendered frame or frame: mFence is cleared, mRenderTimeNs is set.
+    // Key characteristics: mFence is NULL.
+    //
+    struct Info {
+        // set by client during onFrameQueued or onFrameRendered
+        int64_t getMediaTimeUs() const  { return mMediaTimeUs; }
+
+        // -1 if frame is not yet rendered
+        nsecs_t getRenderTimeNs() const { return mRenderTimeNs; }
+
+        // set by client during updateRenderInfoForDequeuedBuffer; -1 otherwise
+        ssize_t getIndex() const        { return mIndex; }
+
+        // creates information for a queued frame
+        Info(int64_t mediaTimeUs, const sp<GraphicBuffer> &graphicBuffer,
+                const sp<Fence> &fence)
+          : mMediaTimeUs(mediaTimeUs),
+            mRenderTimeNs(-1),
+            mIndex(-1),
+            mGraphicBuffer(graphicBuffer),
+            mFence(fence) {
+        }
+
+        // creates information for a frame rendered on a tunneled surface
+        Info(int64_t mediaTimeUs, nsecs_t renderTimeNs)
+            : mMediaTimeUs(mediaTimeUs),
+            mRenderTimeNs(renderTimeNs),
+            mIndex(-1),
+            mGraphicBuffer(NULL),
+            mFence(NULL) {
+        }
+
+    private:
+        int64_t mMediaTimeUs;
+        nsecs_t mRenderTimeNs;
+        ssize_t mIndex;         // to be used by client
+        sp<GraphicBuffer> mGraphicBuffer;
+        sp<Fence> mFence;
+
+        friend struct FrameRenderTracker;
+    };
 
     FrameRenderTracker();
 
diff --git a/media/libstagefright/include/media/stagefright/MediaCodec.h b/media/libstagefright/include/media/stagefright/MediaCodec.h
index bc0f6c5..ceba7d7 100644
--- a/media/libstagefright/include/media/stagefright/MediaCodec.h
+++ b/media/libstagefright/include/media/stagefright/MediaCodec.h
@@ -64,6 +64,7 @@
 class MediaCodecBuffer;
 class IMemory;
 struct PersistentSurface;
+class RenderedFrameInfo;
 class SoftwareRenderer;
 class Surface;
 namespace hardware {
@@ -281,6 +282,8 @@
     // by adding rendered frame information to a base notification message. Returns the number
     // of frames that were rendered.
     static size_t CreateFramesRenderedMessage(
+            const std::list<RenderedFrameInfo> &done, sp<AMessage> &msg);
+    static size_t CreateFramesRenderedMessage(
             const std::list<FrameRenderTracker::Info> &done, sp<AMessage> &msg);
 
     static status_t CanFetchLinearBlock(
diff --git a/media/libstagefright/include/media/stagefright/RenderedFrameInfo.h b/media/libstagefright/include/media/stagefright/RenderedFrameInfo.h
new file mode 100644
index 0000000..4b8a58d
--- /dev/null
+++ b/media/libstagefright/include/media/stagefright/RenderedFrameInfo.h
@@ -0,0 +1,37 @@
+/*
+ * Copyright 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef RENDERED_FRAME_INFO_H
+#define RENDERED_FRAME_INFO_H
+
+namespace android {
+
+class RenderedFrameInfo {
+public:
+    RenderedFrameInfo(int64_t mediaTimeUs, int64_t renderTimeNs)
+        : mMediaTimeUs(mediaTimeUs), mRenderTimeNs(renderTimeNs) {}
+
+    int64_t getMediaTimeUs() const  { return mMediaTimeUs; }
+    nsecs_t getRenderTimeNs() const { return mRenderTimeNs;}
+
+private:
+    int64_t mMediaTimeUs;
+    nsecs_t mRenderTimeNs;
+};
+
+} // android
+
+#endif // RENDERED_FRAME_INFO_H
\ No newline at end of file
diff --git a/media/libstagefright/include/media/stagefright/VideoRenderQualityTracker.h b/media/libstagefright/include/media/stagefright/VideoRenderQualityTracker.h
index 82ba81c..a656e6e 100644
--- a/media/libstagefright/include/media/stagefright/VideoRenderQualityTracker.h
+++ b/media/libstagefright/include/media/stagefright/VideoRenderQualityTracker.h
@@ -38,6 +38,9 @@
     // The render time of the first video frame.
     int64_t firstRenderTimeUs;
 
+    // The render time of the last video frame.
+    int64_t lastRenderTimeUs;
+
     // The number of frames released to be rendered.
     int64_t frameReleasedCount;
 
diff --git a/media/utils/BatteryNotifier.cpp b/media/utils/BatteryNotifier.cpp
index 09bc042..7762c24 100644
--- a/media/utils/BatteryNotifier.cpp
+++ b/media/utils/BatteryNotifier.cpp
@@ -85,8 +85,8 @@
 
 void BatteryNotifier::noteStopAudio(uid_t uid) {
     Mutex::Autolock _l(mLock);
-    if (mAudioRefCounts.find(uid) == mAudioRefCounts.end()) {
-        ALOGW("%s: audio refcount is broken for uid(%d).", __FUNCTION__, (int)uid);
+    if (mAudioRefCounts.find(uid) == mAudioRefCounts.end() || (mAudioRefCounts[uid] == 0)) {
+        ALOGE("%s: audio refcount is broken for uid(%d).", __FUNCTION__, (int)uid);
         return;
     }
 
diff --git a/media/utils/include/mediautils/BatteryNotifier.h b/media/utils/include/mediautils/BatteryNotifier.h
index 3812d7a..73bed4a 100644
--- a/media/utils/include/mediautils/BatteryNotifier.h
+++ b/media/utils/include/mediautils/BatteryNotifier.h
@@ -68,6 +68,38 @@
     sp<IBatteryStats> getBatteryService_l();
 };
 
+namespace mediautils {
+class BatteryStatsAudioHandle {
+  public:
+    static constexpr uid_t INVALID_UID = static_cast<uid_t>(-1);
+
+    explicit BatteryStatsAudioHandle(uid_t uid) : mUid(uid) {
+        if (uid != INVALID_UID) {
+            BatteryNotifier::getInstance().noteStartAudio(mUid);
+        }
+    }
+
+    BatteryStatsAudioHandle(BatteryStatsAudioHandle&& other) : mUid(other.mUid) {
+        other.mUid = INVALID_UID;
+    }
+
+    BatteryStatsAudioHandle(const BatteryStatsAudioHandle& other) = delete;
+
+    BatteryStatsAudioHandle& operator=(const BatteryStatsAudioHandle& other) = delete;
+
+    BatteryStatsAudioHandle& operator=(BatteryStatsAudioHandle&& other) = delete;
+
+    ~BatteryStatsAudioHandle() {
+        if (mUid != INVALID_UID) {
+            BatteryNotifier::getInstance().noteStopAudio(mUid);
+        }
+    }
+
+  private:
+    // Logically const
+    uid_t mUid = INVALID_UID;
+};
+}  // namespace mediautils
 }  // namespace android
 
 #endif // MEDIA_BATTERY_NOTIFIER_H
diff --git a/services/audioflinger/Android.bp b/services/audioflinger/Android.bp
index 17e6d15..0cd6243 100644
--- a/services/audioflinger/Android.bp
+++ b/services/audioflinger/Android.bp
@@ -149,6 +149,7 @@
 
     srcs: [
         "AudioFlinger.cpp",
+        "Client.cpp",
         "DeviceEffectManager.cpp",
         "Effects.cpp",
         "MelReporter.cpp",
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index a0985c7..4842d0b 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -86,7 +86,9 @@
 #include <private/android_filesystem_config.h>
 
 //#define BUFLOG_NDEBUG 0
+#include <afutils/DumpTryLock.h>
 #include <afutils/BufLog.h>
+#include <afutils/Permission.h>
 #include <afutils/TypedLogger.h>
 
 // ----------------------------------------------------------------------------
@@ -126,8 +128,6 @@
 
 nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
 
-uint32_t AudioFlinger::mScreenState;
-
 // In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off
 // we define a minimum time during which a global effect is considered enabled.
 static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200);
@@ -269,33 +269,6 @@
     }
 };
 
-// TODO b/182392769: use attribution source util
-/* static */
-AttributionSourceState AudioFlinger::checkAttributionSourcePackage(
-        const AttributionSourceState& attributionSource) {
-    Vector<String16> packages;
-    PermissionController{}.getPackagesForUid(attributionSource.uid, packages);
-
-    AttributionSourceState checkedAttributionSource = attributionSource;
-    if (!attributionSource.packageName.has_value()
-            || attributionSource.packageName.value().size() == 0) {
-        if (!packages.isEmpty()) {
-            checkedAttributionSource.packageName =
-                std::move(legacy2aidl_String16_string(packages[0]).value());
-        }
-    } else {
-        String16 opPackageLegacy = VALUE_OR_FATAL(
-            aidl2legacy_string_view_String16(attributionSource.packageName.value_or("")));
-        if (std::find_if(packages.begin(), packages.end(),
-                [&opPackageLegacy](const auto& package) {
-                return opPackageLegacy == package; }) == packages.end()) {
-            ALOGW("The package name(%s) provided does not correspond to the uid %d",
-                    attributionSource.packageName.value_or("").c_str(), attributionSource.uid);
-        }
-    }
-    return checkedAttributionSource;
-}
-
 // ----------------------------------------------------------------------------
 
 std::string formatToString(audio_format_t format) {
@@ -328,10 +301,7 @@
       mTotalMemory(0),
       mClientSharedHeapSize(kMinimumClientSharedHeapSizeBytes),
       mGlobalEffectEnableTime(0),
-      mPatchPanel(this),
       mPatchCommandThread(sp<PatchCommandThread>::make()),
-      mDeviceEffectManager(sp<DeviceEffectManager>::make(*this)),
-      mMelReporter(sp<MelReporter>::make(*this)),
       mSystemReady(false),
       mBluetoothLatencyModesEnabled(true)
 {
@@ -405,7 +375,8 @@
     mMode = AUDIO_MODE_NORMAL;
 
     gAudioFlinger = this;  // we are already refcounted, store into atomic pointer.
-
+    mDeviceEffectManager = sp<DeviceEffectManager>::make(
+            sp<IAfDeviceEffectManagerCallback>::fromExisting(this)),
     mDevicesFactoryHalCallback = new DevicesFactoryHalCallbackImpl;
     mDevicesFactoryHal->setCallbackOnce(mDevicesFactoryHalCallback);
 
@@ -413,6 +384,9 @@
         mAAudioBurstsPerBuffer = getAAudioMixerBurstCountFromSystemProperty();
         mAAudioHwBurstMinMicros = getAAudioHardwareBurstMinUsecFromSystemProperty();
     }
+
+    mPatchPanel = IAfPatchPanel::create(sp<IAfPatchPanelCallback>::fromExisting(this));
+    mMelReporter = sp<MelReporter>::make(sp<IAfMelReporterCallback>::fromExisting(this));
 }
 
 status_t AudioFlinger::setAudioHalPids(const std::vector<pid_t>& pids) {
@@ -433,9 +407,9 @@
     for (const auto& [trackId, secondaryOutputs] : trackSecondaryOutputs) {
         size_t i = 0;
         for (; i < mPlaybackThreads.size(); ++i) {
-            PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
-            Mutex::Autolock _tl(thread->mLock);
-            sp<PlaybackThread::Track> track = thread->getTrackById_l(trackId);
+            IAfPlaybackThread* thread = mPlaybackThreads.valueAt(i).get();
+            Mutex::Autolock _tl(thread->mutex());
+            sp<IAfTrack> track = thread->getTrackById_l(trackId);
             if (track != nullptr) {
                 ALOGD("%s trackId: %u", __func__, trackId);
                 updateSecondaryOutputsForTrack_l(track.get(), thread, secondaryOutputs);
@@ -476,12 +450,12 @@
     return NO_ERROR;
 }
 
-int32_t AudioFlinger::getAAudioMixerBurstCount() {
+int32_t AudioFlinger::getAAudioMixerBurstCount() const {
     Mutex::Autolock _l(mLock);
     return mAAudioBurstsPerBuffer;
 }
 
-int32_t AudioFlinger::getAAudioHardwareBurstMinUsec() {
+int32_t AudioFlinger::getAAudioHardwareBurstMinUsec() const {
     Mutex::Autolock _l(mLock);
     return mAAudioHwBurstMinMicros;
 }
@@ -527,7 +501,7 @@
 }
 
 // getDefaultVibratorInfo_l must be called with AudioFlinger lock held.
-std::optional<media::AudioVibratorInfo> AudioFlinger::getDefaultVibratorInfo_l() {
+std::optional<media::AudioVibratorInfo> AudioFlinger::getDefaultVibratorInfo_l() const {
     if (mAudioVibratorInfos.empty()) {
         return {};
     }
@@ -638,7 +612,7 @@
                  __func__, callingUid, callingPid, clientPid);
         adjAttributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
     }
-    adjAttributionSource = AudioFlinger::checkAttributionSourcePackage(
+    adjAttributionSource = afutils::checkAttributionSourcePackage(
             adjAttributionSource);
 
     if (direction == MmapStreamInterface::DIRECTION_OUTPUT) {
@@ -678,9 +652,9 @@
 
     // at this stage, a MmapThread was created when openOutput() or openInput() was called by
     // audio policy manager and we can retrieve it
-    sp<MmapThread> thread = mMmapThreads.valueFor(io);
+    const sp<IAfMmapThread> thread = mMmapThreads.valueFor(io);
     if (thread != 0) {
-        interface = new MmapThreadHandle(thread);
+        interface = IAfMmapThread::createMmapStreamInterfaceAdapter(thread);
         thread->configure(&localAttr, streamType, actualSessionId, callback, *deviceId, portId);
         *handle = portId;
         *sessionId = actualSessionId;
@@ -857,12 +831,6 @@
     write(fd, result.string(), result.size());
 }
 
-bool AudioFlinger::dumpTryLock(Mutex& mutex)
-{
-    status_t err = mutex.timedLock(kDumpLockTimeoutNs);
-    return err == NO_ERROR;
-}
-
 status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
 NO_THREAD_SAFETY_ANALYSIS  // conditional try lock
 {
@@ -870,7 +838,7 @@
         dumpPermissionDenial(fd, args);
     } else {
         // get state of hardware lock
-        bool hardwareLocked = dumpTryLock(mHardwareLock);
+        const bool hardwareLocked = afutils::dumpTryLock(mHardwareLock);
         if (!hardwareLocked) {
             String8 result(kHardwareLockedString);
             write(fd, result.string(), result.size());
@@ -878,7 +846,7 @@
             mHardwareLock.unlock();
         }
 
-        const bool locked = dumpTryLock(mLock);
+        const bool locked = afutils::dumpTryLock(mLock);
 
         // failed to lock - AudioFlinger is probably deadlocked
         if (!locked) {
@@ -886,7 +854,7 @@
             write(fd, result.string(), result.size());
         }
 
-        bool clientLocked = dumpTryLock(mClientLock);
+        const bool clientLocked = afutils::dumpTryLock(mClientLock);
         if (!clientLocked) {
             String8 result(kClientLockedString);
             write(fd, result.string(), result.size());
@@ -934,7 +902,7 @@
             dev->dump(fd, args);
         }
 
-        mPatchPanel.dump(fd);
+        mPatchPanel->dump(fd);
 
         mDeviceEffectManager->dump(fd);
 
@@ -1036,7 +1004,7 @@
     // (for which promote() is always != 0), otherwise create a new entry and Client.
     sp<Client> client = mClients.valueFor(pid).promote();
     if (client == 0) {
-        client = new Client(this, pid);
+        client = sp<Client>::make(sp<IAfClientCallback>::fromExisting(this), pid);
         mClients.add(pid, client);
     }
 
@@ -1107,7 +1075,7 @@
     CreateTrackInput input = VALUE_OR_RETURN_STATUS(CreateTrackInput::fromAidl(_input));
     CreateTrackOutput output;
 
-    sp<PlaybackThread::Track> track;
+    sp<IAfTrack> track;
     sp<Client> client;
     status_t lStatus;
     audio_stream_type_t streamType;
@@ -1144,7 +1112,7 @@
         clientPid = callingPid;
         adjAttributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
     }
-    adjAttributionSource = AudioFlinger::checkAttributionSourcePackage(
+    adjAttributionSource = afutils::checkAttributionSourcePackage(
             adjAttributionSource);
 
     audio_session_t sessionId = input.sessionId;
@@ -1191,7 +1159,7 @@
 
     {
         Mutex::Autolock _l(mLock);
-        PlaybackThread *thread = checkPlaybackThread_l(output.outputId);
+        IAfPlaybackThread* thread = checkPlaybackThread_l(output.outputId);
         if (thread == NULL) {
             ALOGE("no playback thread found for output handle %d", output.outputId);
             lStatus = BAD_VALUE;
@@ -1200,14 +1168,14 @@
 
         client = registerPid(clientPid);
 
-        PlaybackThread *effectThread = NULL;
+        IAfPlaybackThread* effectThread = nullptr;
         // check if an effect chain with the same session ID is present on another
         // output thread and move it here.
         for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
-            sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
+            sp<IAfPlaybackThread> t = mPlaybackThreads.valueAt(i);
             if (mPlaybackThreads.keyAt(i) != output.outputId) {
                 uint32_t sessions = t->hasAudioSession(sessionId);
-                if (sessions & ThreadBase::EFFECT_SESSION) {
+                if (sessions & IAfThreadBase::EFFECT_SESSION) {
                     effectThread = t.get();
                     break;
                 }
@@ -1242,7 +1210,7 @@
 
         if (lStatus == NO_ERROR) {
             // no risk of deadlock because AudioFlinger::mLock is held
-            Mutex::Autolock _dl(thread->mLock);
+            Mutex::Autolock _dl(thread->mutex());
             // Connect secondary outputs. Failure on a secondary output must not imped the primary
             // Any secondary output setup failure will lead to a desync between the AP and AF until
             // the track is destroyed.
@@ -1250,7 +1218,7 @@
             // move effect chain to this output thread if an effect on same session was waiting
             // for a track to be created
             if (effectThread != nullptr) {
-                Mutex::Autolock _sl(effectThread->mLock);
+                Mutex::Autolock _sl(effectThread->mutex());
                 if (moveEffectChain_l(sessionId, effectThread, thread) == NO_ERROR) {
                     effectThreadId = thread->id();
                     effectIds = thread->getEffectIds_l(sessionId);
@@ -1297,7 +1265,7 @@
         AudioSystem::moveEffectsToIo(effectIds, effectThreadId);
     }
 
-    output.audioTrack = PlaybackThread::Track::createIAudioTrackAdapter(track);
+    output.audioTrack = IAfTrack::createIAudioTrackAdapter(track);
     _output = VALUE_OR_FATAL(output.toAidl());
 
 Exit:
@@ -1310,7 +1278,7 @@
 uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const
 {
     Mutex::Autolock _l(mLock);
-    ThreadBase *thread = checkThread_l(ioHandle);
+    IAfThreadBase* const thread = checkThread_l(ioHandle);
     if (thread == NULL) {
         ALOGW("sampleRate() unknown thread %d", ioHandle);
         return 0;
@@ -1321,7 +1289,7 @@
 audio_format_t AudioFlinger::format(audio_io_handle_t output) const
 {
     Mutex::Autolock _l(mLock);
-    PlaybackThread *thread = checkPlaybackThread_l(output);
+    IAfPlaybackThread* const thread = checkPlaybackThread_l(output);
     if (thread == NULL) {
         ALOGW("format() unknown thread %d", output);
         return AUDIO_FORMAT_INVALID;
@@ -1332,7 +1300,7 @@
 size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const
 {
     Mutex::Autolock _l(mLock);
-    ThreadBase *thread = checkThread_l(ioHandle);
+    IAfThreadBase* const thread = checkThread_l(ioHandle);
     if (thread == NULL) {
         ALOGW("frameCount() unknown thread %d", ioHandle);
         return 0;
@@ -1345,7 +1313,7 @@
 size_t AudioFlinger::frameCountHAL(audio_io_handle_t ioHandle) const
 {
     Mutex::Autolock _l(mLock);
-    ThreadBase *thread = checkThread_l(ioHandle);
+    IAfThreadBase* const thread = checkThread_l(ioHandle);
     if (thread == NULL) {
         ALOGW("frameCountHAL() unknown thread %d", ioHandle);
         return 0;
@@ -1356,7 +1324,7 @@
 uint32_t AudioFlinger::latency(audio_io_handle_t output) const
 {
     Mutex::Autolock _l(mLock);
-    PlaybackThread *thread = checkPlaybackThread_l(output);
+    IAfPlaybackThread* const thread = checkPlaybackThread_l(output);
     if (thread == NULL) {
         ALOGW("latency(): no playback thread found for output handle %d", output);
         return 0;
@@ -1585,7 +1553,7 @@
     // assigned to HALs which do not have master mute support will apply master mute
     // during the mix operation.  Threads with HALs which do support master mute
     // will simply ignore the setting.
-    Vector<VolumeInterface *> volumeInterfaces = getAllVolumeInterfaces_l();
+    std::vector<sp<VolumeInterface>> volumeInterfaces = getAllVolumeInterfaces_l();
     for (size_t i = 0; i < volumeInterfaces.size(); i++) {
         volumeInterfaces[i]->setMasterMute(muted);
     }
@@ -1661,7 +1629,7 @@
                         "AUDIO_STREAM_PATCH must have full scale volume");
 
     AutoMutex lock(mLock);
-    VolumeInterface *volumeInterface = getVolumeInterface_l(output);
+    sp<VolumeInterface> volumeInterface = getVolumeInterface_l(output);
     if (volumeInterface == NULL) {
         return BAD_VALUE;
     }
@@ -1676,7 +1644,7 @@
         return BAD_VALUE;
     }
     AutoMutex lock(mLock);
-    PlaybackThread *thread = checkPlaybackThread_l(output);
+    IAfPlaybackThread* const thread = checkPlaybackThread_l(output);
     if (thread == nullptr) {
         return BAD_VALUE;
     }
@@ -1684,12 +1652,12 @@
 }
 
 status_t AudioFlinger::getSupportedLatencyModes(audio_io_handle_t output,
-            std::vector<audio_latency_mode_t>* modes) {
+            std::vector<audio_latency_mode_t>* modes) const {
     if (output == AUDIO_IO_HANDLE_NONE) {
         return BAD_VALUE;
     }
     AutoMutex lock(mLock);
-    PlaybackThread *thread = checkPlaybackThread_l(output);
+    IAfPlaybackThread* const thread = checkPlaybackThread_l(output);
     if (thread == nullptr) {
         return BAD_VALUE;
     }
@@ -1711,7 +1679,7 @@
     return status;
 }
 
-status_t AudioFlinger::isBluetoothVariableLatencyEnabled(bool *enabled) {
+status_t AudioFlinger::isBluetoothVariableLatencyEnabled(bool* enabled) const {
     if (enabled == nullptr) {
         return BAD_VALUE;
     }
@@ -1719,7 +1687,7 @@
     return NO_ERROR;
 }
 
-status_t AudioFlinger::supportsBluetoothVariableLatency(bool* support) {
+status_t AudioFlinger::supportsBluetoothVariableLatency(bool* support) const {
     if (support == nullptr) {
         return BAD_VALUE;
     }
@@ -1735,7 +1703,7 @@
 }
 
 status_t AudioFlinger::getSoundDoseInterface(const sp<media::ISoundDoseCallback>& callback,
-                                             sp<media::ISoundDose>* soundDose) {
+                                             sp<media::ISoundDose>* soundDose) const {
     if (soundDose == nullptr) {
         return BAD_VALUE;
     }
@@ -1764,7 +1732,7 @@
 
     AutoMutex lock(mLock);
     mStreamTypes[stream].mute = muted;
-    Vector<VolumeInterface *> volumeInterfaces = getAllVolumeInterfaces_l();
+    std::vector<sp<VolumeInterface>> volumeInterfaces = getAllVolumeInterfaces_l();
     for (size_t i = 0; i < volumeInterfaces.size(); i++) {
         volumeInterfaces[i]->setStreamMute(stream, muted);
     }
@@ -1783,7 +1751,7 @@
     }
 
     AutoMutex lock(mLock);
-    VolumeInterface *volumeInterface = getVolumeInterface_l(output);
+    sp<VolumeInterface> volumeInterface = getVolumeInterface_l(output);
     if (volumeInterface == NULL) {
         return 0.0f;
     }
@@ -1820,14 +1788,15 @@
 // forwardAudioHwSyncToDownstreamPatches_l() must be called with AudioFlinger::mLock held
 void AudioFlinger::forwardParametersToDownstreamPatches_l(
         audio_io_handle_t upStream, const String8& keyValuePairs,
-        const std::function<bool(const sp<PlaybackThread>&)>& useThread)
+        const std::function<bool(const sp<IAfPlaybackThread>&)>& useThread)
 {
-    std::vector<PatchPanel::SoftwarePatch> swPatches;
-    if (mPatchPanel.getDownstreamSoftwarePatches(upStream, &swPatches) != OK) return;
+    std::vector<SoftwarePatch> swPatches;
+    if (mPatchPanel->getDownstreamSoftwarePatches(upStream, &swPatches) != OK) return;
     ALOGV_IF(!swPatches.empty(), "%s found %zu downstream patches for stream ID %d",
             __func__, swPatches.size(), upStream);
     for (const auto& swPatch : swPatches) {
-        sp<PlaybackThread> downStream = checkPlaybackThread_l(swPatch.getPlaybackThreadHandle());
+        const sp<IAfPlaybackThread> downStream =
+                checkPlaybackThread_l(swPatch.getPlaybackThreadHandle());
         if (downStream != NULL && (useThread == nullptr || useThread(downStream))) {
             downStream->setParameters(keyValuePairs);
         }
@@ -1839,7 +1808,7 @@
                                              const std::set<audio_io_handle_t>& streams)
 {
     for (const audio_io_handle_t stream : streams) {
-        PlaybackThread *playbackThread = checkPlaybackThread_l(stream);
+        IAfPlaybackThread* const playbackThread = checkPlaybackThread_l(stream);
         if (playbackThread == nullptr || !playbackThread->isMsdDevice()) {
             continue;
         }
@@ -1953,8 +1922,8 @@
         String8 screenState;
         if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
             bool isOff = (screenState == AudioParameter::valueOff);
-            if (isOff != (AudioFlinger::mScreenState & 1)) {
-                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
+            if (isOff != (mScreenState & 1)) {
+                mScreenState = ((mScreenState & ~1) + 2) | isOff;
             }
         }
         return final_result;
@@ -1962,7 +1931,7 @@
 
     // hold a strong ref on thread in case closeOutput() or closeInput() is called
     // and the thread is exited once the lock is released
-    sp<ThreadBase> thread;
+    sp<IAfThreadBase> thread;
     {
         Mutex::Autolock _l(mLock);
         thread = checkPlaybackThread_l(ioHandle);
@@ -2011,11 +1980,11 @@
         return out_s8;
     }
 
-    ThreadBase *thread = (ThreadBase *)checkPlaybackThread_l(ioHandle);
+    IAfThreadBase* thread = checkPlaybackThread_l(ioHandle);
     if (thread == NULL) {
-        thread = (ThreadBase *)checkRecordThread_l(ioHandle);
+        thread = checkRecordThread_l(ioHandle);
         if (thread == NULL) {
-            thread = (ThreadBase *)checkMmapThread_l(ioHandle);
+            thread = checkMmapThread_l(ioHandle);
             if (thread == NULL) {
                 return String8("");
             }
@@ -2111,7 +2080,7 @@
 {
     Mutex::Autolock _l(mLock);
 
-    RecordThread *recordThread = checkRecordThread_l(ioHandle);
+    IAfRecordThread* const recordThread = checkRecordThread_l(ioHandle);
     if (recordThread != NULL) {
         return recordThread->getInputFramesLost();
     }
@@ -2151,7 +2120,7 @@
 {
     Mutex::Autolock _l(mLock);
 
-    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
+    IAfPlaybackThread* const playbackThread = checkPlaybackThread_l(output);
     if (playbackThread != NULL) {
         return playbackThread->getRenderPosition(halFrames, dspFrames);
     }
@@ -2274,10 +2243,10 @@
 }
 
 // getEffectThread_l() must be called with AudioFlinger::mLock held
-sp<AudioFlinger::ThreadBase> AudioFlinger::getEffectThread_l(audio_session_t sessionId,
+sp<IAfThreadBase> AudioFlinger::getEffectThread_l(audio_session_t sessionId,
         int effectId)
 {
-    sp<ThreadBase> thread;
+    sp<IAfThreadBase> thread;
 
     for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
         if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, effectId) != 0) {
@@ -2306,27 +2275,6 @@
     return thread;
 }
 
-
-
-// ----------------------------------------------------------------------------
-
-Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
-    :   RefBase(),
-        mAudioFlinger(audioFlinger),
-        mPid(pid),
-        mClientAllocator(AllocatorFactory::getClientAllocator()) {}
-
-// Client destructor must be called with AudioFlinger::mClientLock held
-Client::~Client()
-{
-    mAudioFlinger->removeClient_l(mPid);
-}
-
-AllocatorFactory::ClientAllocator& Client::allocator()
-{
-    return mClientAllocator;
-}
-
 // ----------------------------------------------------------------------------
 
 AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
@@ -2391,7 +2339,7 @@
     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;
@@ -2423,7 +2371,7 @@
                  __func__, callingUid, callingPid, currentPid);
         adjAttributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
     }
-    adjAttributionSource = AudioFlinger::checkAttributionSourcePackage(
+    adjAttributionSource = afutils::checkAttributionSourcePackage(
             adjAttributionSource);
     // we don't yet support anything other than linear PCM
     if (!audio_is_valid_format(input.config.format) || !audio_is_linear_pcm(input.config.format)) {
@@ -2480,7 +2428,7 @@
 
     {
         Mutex::Autolock _l(mLock);
-        RecordThread *thread = checkRecordThread_l(output.inputId);
+        IAfRecordThread* const thread = checkRecordThread_l(output.inputId);
         if (thread == NULL) {
             ALOGW("createRecord() checkRecordThread_l failed, input handle %d", output.inputId);
             lStatus = FAILED_TRANSACTION;
@@ -2536,7 +2484,7 @@
         // session and move it to this thread.
         sp<IAfEffectChain> chain = getOrphanEffectChain_l(sessionId);
         if (chain != 0) {
-            Mutex::Autolock _l2(thread->mLock);
+            Mutex::Autolock _l2(thread->mutex());
             thread->addEffectChain_l(chain);
         }
         break;
@@ -2549,7 +2497,7 @@
     output.buffers = recordTrack->getBuffers();
     output.portId = portId;
 
-    output.audioRecord = RecordThread::RecordTrack::createIAudioRecordAdapter(recordTrack);
+    output.audioRecord = IAfRecordTrack::createIAudioRecordAdapter(recordTrack);
     _output = VALUE_OR_FATAL(output.toAidl());
 
 Exit:
@@ -2735,17 +2683,17 @@
 
 // ----------------------------------------------------------------------------
 
-uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
+uint32_t AudioFlinger::getPrimaryOutputSamplingRate() const
 {
     Mutex::Autolock _l(mLock);
-    PlaybackThread *thread = fastPlaybackThread_l();
+    IAfPlaybackThread* const thread = fastPlaybackThread_l();
     return thread != NULL ? thread->sampleRate() : 0;
 }
 
-size_t AudioFlinger::getPrimaryOutputFrameCount()
+size_t AudioFlinger::getPrimaryOutputFrameCount() const
 {
     Mutex::Autolock _l(mLock);
-    PlaybackThread *thread = fastPlaybackThread_l();
+    IAfPlaybackThread* const thread = fastPlaybackThread_l();
     return thread != NULL ? thread->frameCountHAL() : 0;
 }
 
@@ -2870,15 +2818,15 @@
     mHwAvSyncIds.add(sessionId, value);
 
     for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
-        sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i);
+        const sp<IAfPlaybackThread> thread = mPlaybackThreads.valueAt(i);
         uint32_t sessions = thread->hasAudioSession(sessionId);
-        if (sessions & ThreadBase::TRACK_SESSION) {
+        if (sessions & IAfThreadBase::TRACK_SESSION) {
             AudioParameter param = AudioParameter();
             param.addInt(String8(AudioParameter::keyStreamHwAvSync), value);
             String8 keyValuePairs = param.toString();
             thread->setParameters(keyValuePairs);
             forwardParametersToDownstreamPatches_l(thread->id(), keyValuePairs,
-                    [](const sp<PlaybackThread>& thread) { return thread->usesHwAvSync(); });
+                    [](const sp<IAfPlaybackThread>& thread) { return thread->usesHwAvSync(); });
             break;
         }
     }
@@ -2897,15 +2845,15 @@
     }
     mSystemReady = true;
     for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
-        ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get();
+        IAfThreadBase* const thread = mPlaybackThreads.valueAt(i).get();
         thread->systemReady();
     }
     for (size_t i = 0; i < mRecordThreads.size(); i++) {
-        ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get();
+        IAfThreadBase* const thread = mRecordThreads.valueAt(i).get();
         thread->systemReady();
     }
     for (size_t i = 0; i < mMmapThreads.size(); i++) {
-        ThreadBase *thread = (ThreadBase *)mMmapThreads.valueAt(i).get();
+        IAfThreadBase* const thread = mMmapThreads.valueAt(i).get();
         thread->systemReady();
     }
 
@@ -2930,7 +2878,7 @@
     return mAudioManager.load();
 }
 
-status_t AudioFlinger::getMicrophones(std::vector<media::MicrophoneInfoFw> *microphones)
+status_t AudioFlinger::getMicrophones(std::vector<media::MicrophoneInfoFw>* microphones) const
 {
     AutoMutex lock(mHardwareLock);
     status_t status = INVALID_OPERATION;
@@ -2957,7 +2905,8 @@
 }
 
 // setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId)
+void AudioFlinger::setAudioHwSyncForSession_l(
+        IAfPlaybackThread* const thread, audio_session_t sessionId)
 {
     ssize_t index = mHwAvSyncIds.indexOfKey(sessionId);
     if (index >= 0) {
@@ -2968,7 +2917,7 @@
         String8 keyValuePairs = param.toString();
         thread->setParameters(keyValuePairs);
         forwardParametersToDownstreamPatches_l(thread->id(), keyValuePairs,
-                [](const sp<PlaybackThread>& thread) { return thread->usesHwAvSync(); });
+                [](const sp<IAfPlaybackThread>& thread) { return thread->usesHwAvSync(); });
     }
 }
 
@@ -2976,7 +2925,7 @@
 // ----------------------------------------------------------------------------
 
 
-sp<AudioFlinger::ThreadBase> AudioFlinger::openOutput_l(audio_module_handle_t module,
+sp<IAfThreadBase> AudioFlinger::openOutput_l(audio_module_handle_t module,
                                                         audio_io_handle_t *output,
                                                         audio_config_t *halConfig,
                                                         audio_config_base_t *mixerConfig,
@@ -3034,43 +2983,45 @@
 
     if (status == NO_ERROR) {
         if (flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) {
-            sp<MmapPlaybackThread> thread =
-                    new MmapPlaybackThread(this, *output, outHwDev, outputStream, mSystemReady);
+            const sp<IAfMmapPlaybackThread> thread = IAfMmapPlaybackThread::create(
+                    this, *output, outHwDev, outputStream, mSystemReady);
             mMmapThreads.add(*output, thread);
             ALOGV("openOutput_l() created mmap playback thread: ID %d thread %p",
                   *output, thread.get());
             return thread;
         } else {
-            sp<PlaybackThread> thread;
+            sp<IAfPlaybackThread> thread;
             if (flags & AUDIO_OUTPUT_FLAG_BIT_PERFECT) {
-                thread = sp<BitPerfectThread>::make(this, outputStream, *output, mSystemReady);
+                thread = IAfPlaybackThread::createBitPerfectThread(
+                        this, outputStream, *output, mSystemReady);
                 ALOGV("%s() created bit-perfect output: ID %d thread %p",
                       __func__, *output, thread.get());
             } else if (flags & AUDIO_OUTPUT_FLAG_SPATIALIZER) {
-                thread = new SpatializerThread(this, outputStream, *output,
+                thread = IAfPlaybackThread::createSpatializerThread(this, outputStream, *output,
                                                     mSystemReady, mixerConfig);
                 ALOGV("openOutput_l() created spatializer output: ID %d thread %p",
                       *output, thread.get());
             } else if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
-                thread = new OffloadThread(this, outputStream, *output,
+                thread = IAfPlaybackThread::createOffloadThread(this, outputStream, *output,
                         mSystemReady, halConfig->offload_info);
                 ALOGV("openOutput_l() created offload output: ID %d thread %p",
                       *output, thread.get());
             } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
                     || !isValidPcmSinkFormat(halConfig->format)
                     || !isValidPcmSinkChannelMask(halConfig->channel_mask)) {
-                thread = new DirectOutputThread(this, outputStream, *output,
+                thread = IAfPlaybackThread::createDirectOutputThread(this, outputStream, *output,
                         mSystemReady, halConfig->offload_info);
                 ALOGV("openOutput_l() created direct output: ID %d thread %p",
                       *output, thread.get());
             } else {
-                thread = new MixerThread(this, outputStream, *output, mSystemReady);
+                thread = IAfPlaybackThread::createMixerThread(
+                        this, outputStream, *output, mSystemReady);
                 ALOGV("openOutput_l() created mixer output: ID %d thread %p",
                       *output, thread.get());
             }
             mPlaybackThreads.add(*output, thread);
             struct audio_patch patch;
-            mPatchPanel.notifyStreamOpened(outHwDev, *output, &patch);
+            mPatchPanel->notifyStreamOpened(outHwDev, *output, &patch);
             if (thread->isMsdDevice()) {
                 thread->setDownStreamPatch(&patch);
             }
@@ -3116,12 +3067,12 @@
 
     Mutex::Autolock _l(mLock);
 
-    sp<ThreadBase> thread = openOutput_l(module, &output, &halConfig,
+    const sp<IAfThreadBase> thread = openOutput_l(module, &output, &halConfig,
             &mixerConfig, deviceType, address, flags);
     if (thread != 0) {
         uint32_t latencyMs = 0;
         if ((flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0) {
-            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+            const auto playbackThread = thread->asIAfPlaybackThread();
             latencyMs = playbackThread->latency();
 
             // notify client processes of the new output creation
@@ -3139,8 +3090,7 @@
                 mHardwareStatus = AUDIO_HW_IDLE;
             }
         } else {
-            MmapThread *mmapThread = (MmapThread *)thread.get();
-            mmapThread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
+            thread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
         }
         response->output = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
         response->config = VALUE_OR_RETURN_STATUS(
@@ -3158,8 +3108,8 @@
         audio_io_handle_t output2)
 {
     Mutex::Autolock _l(mLock);
-    MixerThread *thread1 = checkMixerThread_l(output1);
-    MixerThread *thread2 = checkMixerThread_l(output2);
+    IAfPlaybackThread* const thread1 = checkMixerThread_l(output1);
+    IAfPlaybackThread* const thread2 = checkMixerThread_l(output2);
 
     if (thread1 == NULL || thread2 == NULL) {
         ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
@@ -3168,7 +3118,8 @@
     }
 
     audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT);
-    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady);
+    const sp<IAfDuplicatingThread> thread = IAfDuplicatingThread::create(
+            this, thread1, id, mSystemReady);
     thread->addOutputTrack(thread2);
     mPlaybackThreads.add(id, thread);
     // notify client processes of the new output creation
@@ -3185,8 +3136,8 @@
 {
     // keep strong reference on the playback thread so that
     // it is not destroyed while exit() is executed
-    sp<PlaybackThread> playbackThread;
-    sp<MmapPlaybackThread> mmapThread;
+    sp<IAfPlaybackThread> playbackThread;
+    sp<IAfMmapPlaybackThread> mmapThread;
     {
         Mutex::Autolock _l(mLock);
         playbackThread = checkPlaybackThread_l(output);
@@ -3195,12 +3146,12 @@
 
             dumpToThreadLog_l(playbackThread);
 
-            if (playbackThread->type() == ThreadBase::MIXER) {
+            if (playbackThread->type() == IAfThreadBase::MIXER) {
                 for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
                     if (mPlaybackThreads.valueAt(i)->isDuplicating()) {
-                        DuplicatingThread *dupThread =
-                                (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
-                        dupThread->removeOutputTrack((MixerThread *)playbackThread.get());
+                        IAfDuplicatingThread* const dupThread =
+                                mPlaybackThreads.valueAt(i)->asIAfDuplicatingThread().get();
+                        dupThread->removeOutputTrack(playbackThread.get());
                     }
                 }
             }
@@ -3209,11 +3160,12 @@
             mPlaybackThreads.removeItem(output);
             // save all effects to the default thread
             if (mPlaybackThreads.size()) {
-                PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
+                IAfPlaybackThread* const dstThread =
+                        checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
                 if (dstThread != NULL) {
                     // audioflinger lock is held so order of thread lock acquisition doesn't matter
-                    Mutex::Autolock _dl(dstThread->mLock);
-                    Mutex::Autolock _sl(playbackThread->mLock);
+                    Mutex::Autolock _dl(dstThread->mutex());
+                    Mutex::Autolock _sl(playbackThread->mutex());
                     Vector<sp<IAfEffectChain>> effectChains = playbackThread->getEffectChains_l();
                     for (size_t i = 0; i < effectChains.size(); i ++) {
                         moveEffectChain_l(effectChains[i]->sessionId(), playbackThread.get(),
@@ -3222,7 +3174,8 @@
                 }
             }
         } else {
-            mmapThread = (MmapPlaybackThread *)checkMmapThread_l(output);
+            const sp<IAfMmapThread> mt = checkMmapThread_l(output);
+            mmapThread = mt ? mt->asIAfMmapPlaybackThread().get() : nullptr;
             if (mmapThread == 0) {
                 return BAD_VALUE;
             }
@@ -3231,10 +3184,10 @@
             ALOGD("closing mmapThread %p", mmapThread.get());
         }
         ioConfigChanged(AUDIO_OUTPUT_CLOSED, sp<AudioIoDescriptor>::make(output));
-        mPatchPanel.notifyStreamClosed(output);
+        mPatchPanel->notifyStreamClosed(output);
     }
     // The thread entity (active unit of execution) is no longer running here,
-    // but the ThreadBase container still exists.
+    // but the IAfThreadBase container still exists.
 
     if (playbackThread != 0) {
         playbackThread->exit();
@@ -3252,7 +3205,7 @@
     return NO_ERROR;
 }
 
-void AudioFlinger::closeOutputFinish(const sp<PlaybackThread>& thread)
+void AudioFlinger::closeOutputFinish(const sp<IAfPlaybackThread>& thread)
 {
     AudioStreamOut *out = thread->clearOutput();
     ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
@@ -3260,9 +3213,9 @@
     delete out;
 }
 
-void AudioFlinger::closeThreadInternal_l(const sp<PlaybackThread>& thread)
+void AudioFlinger::closeThreadInternal_l(const sp<IAfPlaybackThread>& thread)
 {
-    mPlaybackThreads.removeItem(thread->mId);
+    mPlaybackThreads.removeItem(thread->id());
     thread->exit();
     closeOutputFinish(thread);
 }
@@ -3270,7 +3223,7 @@
 status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
 {
     Mutex::Autolock _l(mLock);
-    PlaybackThread *thread = checkPlaybackThread_l(output);
+    IAfPlaybackThread* const thread = checkPlaybackThread_l(output);
 
     if (thread == NULL) {
         return BAD_VALUE;
@@ -3285,7 +3238,7 @@
 status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
 {
     Mutex::Autolock _l(mLock);
-    PlaybackThread *thread = checkPlaybackThread_l(output);
+    IAfPlaybackThread* const thread = checkPlaybackThread_l(output);
 
     if (thread == NULL) {
         return BAD_VALUE;
@@ -3314,7 +3267,7 @@
     audio_config_t config = VALUE_OR_RETURN_STATUS(
             aidl2legacy_AudioConfig_audio_config_t(request.config, true /*isInput*/));
 
-    sp<ThreadBase> thread = openInput_l(
+    const sp<IAfThreadBase> thread = openInput_l(
             VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_module_handle_t(request.module)),
             &input,
             &config,
@@ -3338,7 +3291,7 @@
     return NO_INIT;
 }
 
-sp<AudioFlinger::ThreadBase> AudioFlinger::openInput_l(audio_module_handle_t module,
+sp<IAfThreadBase> AudioFlinger::openInput_l(audio_module_handle_t module,
                                                          audio_io_handle_t *input,
                                                          audio_config_t *config,
                                                          audio_devices_t devices,
@@ -3404,17 +3357,18 @@
     if (status == NO_ERROR && inStream != 0) {
         AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream, flags);
         if ((flags & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0) {
-            sp<MmapCaptureThread> thread =
-                    new MmapCaptureThread(this, *input, inHwDev, inputStream, mSystemReady);
+            const sp<IAfMmapCaptureThread> thread =
+                    IAfMmapCaptureThread::create(this, *input, inHwDev, inputStream, mSystemReady);
             mMmapThreads.add(*input, thread);
             ALOGV("openInput_l() created mmap capture thread: ID %d thread %p", *input,
                     thread.get());
             return thread;
         } else {
             // Start record thread
-            // RecordThread requires both input and output device indication to forward to audio
-            // pre processing modules
-            sp<RecordThread> thread = new RecordThread(this, inputStream, *input, mSystemReady);
+            // IAfRecordThread requires both input and output device indication
+            // to forward to audio pre processing modules
+            const sp<IAfRecordThread> thread =
+                    IAfRecordThread::create(this, inputStream, *input, mSystemReady);
             mRecordThreads.add(*input, thread);
             ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get());
             return thread;
@@ -3434,8 +3388,8 @@
 {
     // keep strong reference on the record thread so that
     // it is not destroyed while exit() is executed
-    sp<RecordThread> recordThread;
-    sp<MmapCaptureThread> mmapThread;
+    sp<IAfRecordThread> recordThread;
+    sp<IAfMmapCaptureThread> mmapThread;
     {
         Mutex::Autolock _l(mLock);
         recordThread = checkRecordThread_l(input);
@@ -3450,8 +3404,8 @@
             // new capture on the same session
             sp<IAfEffectChain> chain;
             {
-                Mutex::Autolock _sl(recordThread->mLock);
-                Vector< sp<IAfEffectChain> > effectChains = recordThread->getEffectChains_l();
+                Mutex::Autolock _sl(recordThread->mutex());
+                const Vector<sp<IAfEffectChain>> effectChains = recordThread->getEffectChains_l();
                 // Note: maximum one chain per record thread
                 if (effectChains.size() != 0) {
                     chain = effectChains[0];
@@ -3463,12 +3417,12 @@
                 // creation of its replacement
                 size_t i;
                 for (i = 0; i < mRecordThreads.size(); i++) {
-                    sp<RecordThread> t = mRecordThreads.valueAt(i);
+                    const sp<IAfRecordThread> t = mRecordThreads.valueAt(i);
                     if (t == recordThread) {
                         continue;
                     }
                     if (t->hasAudioSession(chain->sessionId()) != 0) {
-                        Mutex::Autolock _l2(t->mLock);
+                        Mutex::Autolock _l2(t->mutex());
                         ALOGV("closeInput() found thread %d for effect session %d",
                               t->id(), chain->sessionId());
                         t->addEffectChain_l(chain);
@@ -3482,7 +3436,8 @@
             }
             mRecordThreads.removeItem(input);
         } else {
-            mmapThread = (MmapCaptureThread *)checkMmapThread_l(input);
+            const sp<IAfMmapThread> mt = checkMmapThread_l(input);
+            mmapThread = mt ? mt->asIAfMmapCaptureThread().get() : nullptr;
             if (mmapThread == 0) {
                 return BAD_VALUE;
             }
@@ -3505,7 +3460,7 @@
     return NO_ERROR;
 }
 
-void AudioFlinger::closeInputFinish(const sp<RecordThread>& thread)
+void AudioFlinger::closeInputFinish(const sp<IAfRecordThread>& thread)
 {
     thread->exit();
     AudioStreamIn *in = thread->clearInput();
@@ -3514,9 +3469,9 @@
     delete in;
 }
 
-void AudioFlinger::closeThreadInternal_l(const sp<RecordThread>& thread)
+void AudioFlinger::closeThreadInternal_l(const sp<IAfRecordThread>& thread)
 {
-    mRecordThreads.removeItem(thread->mId);
+    mRecordThreads.removeItem(thread->id());
     closeInputFinish(thread);
 }
 
@@ -3526,7 +3481,7 @@
 
     std::set<audio_port_handle_t> portIdSet(portIds.begin(), portIds.end());
     for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
-        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
+        IAfPlaybackThread* const thread = mPlaybackThreads.valueAt(i).get();
         thread->invalidateTracks(portIdSet);
         if (portIdSet.empty()) {
             return NO_ERROR;
@@ -3646,14 +3601,15 @@
 
     ALOGV("purging stale effects");
 
-    Vector< sp<IAfEffectChain> > chains;
+    Vector<sp<IAfEffectChain>> chains;
     std::vector< sp<IAfEffectModule> > removedEffects;
 
     for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
-        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
-        Mutex::Autolock _l(t->mLock);
-        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
-            sp<IAfEffectChain> ec = t->mEffectChains[j];
+        sp<IAfPlaybackThread> t = mPlaybackThreads.valueAt(i);
+        Mutex::Autolock _l(t->mutex());
+        const Vector<sp<IAfEffectChain>> threadChains = t->getEffectChains_l();
+        for (size_t j = 0; j < threadChains.size(); j++) {
+            sp<IAfEffectChain> ec = threadChains[j];
             if (!audio_is_global_session(ec->sessionId())) {
                 chains.push(ec);
             }
@@ -3661,19 +3617,21 @@
     }
 
     for (size_t i = 0; i < mRecordThreads.size(); i++) {
-        sp<RecordThread> t = mRecordThreads.valueAt(i);
-        Mutex::Autolock _l(t->mLock);
-        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
-            sp<IAfEffectChain> ec = t->mEffectChains[j];
+        sp<IAfRecordThread> t = mRecordThreads.valueAt(i);
+        Mutex::Autolock _l(t->mutex());
+        const Vector<sp<IAfEffectChain>> threadChains = t->getEffectChains_l();
+        for (size_t j = 0; j < threadChains.size(); j++) {
+            sp<IAfEffectChain> ec = threadChains[j];
             chains.push(ec);
         }
     }
 
     for (size_t i = 0; i < mMmapThreads.size(); i++) {
-        sp<MmapThread> t = mMmapThreads.valueAt(i);
-        Mutex::Autolock _l(t->mLock);
-        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
-            sp<IAfEffectChain> ec = t->mEffectChains[j];
+        const sp<IAfMmapThread> t = mMmapThreads.valueAt(i);
+        Mutex::Autolock _l(t->mutex());
+        const Vector<sp<IAfEffectChain>> threadChains = t->getEffectChains_l();
+        for (size_t j = 0; j < threadChains.size(); j++) {
+            sp<IAfEffectChain> ec = threadChains[j];
             chains.push(ec);
         }
     }
@@ -3682,7 +3640,7 @@
          // clang-tidy suggests const ref
         sp<IAfEffectChain> ec = chains[i];  // NOLINT(performance-unnecessary-copy-initialization)
         int sessionid = ec->sessionId();
-        sp<ThreadBase> t = sp<ThreadBase>::cast(ec->thread().promote()); // TODO(b/288339104)
+        const auto t = ec->thread().promote();
         if (t == 0) {
             continue;
         }
@@ -3698,7 +3656,7 @@
             }
         }
         if (!found) {
-            Mutex::Autolock _l(t->mLock);
+            Mutex::Autolock _l(t->mutex());
             // remove all effects from the chain
             while (ec->numberOfEffects()) {
                 sp<IAfEffectModule> effect = ec->getEffectModule(0);
@@ -3715,7 +3673,7 @@
 }
 
 // dumpToThreadLog_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::dumpToThreadLog_l(const sp<ThreadBase> &thread)
+void AudioFlinger::dumpToThreadLog_l(const sp<IAfThreadBase> &thread)
 {
     constexpr int THREAD_DUMP_TIMEOUT_MS = 2;
     audio_utils::FdToString fdToString("- ", THREAD_DUMP_TIMEOUT_MS);
@@ -3727,9 +3685,9 @@
 }
 
 // checkThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const
+IAfThreadBase* AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const
 {
-    ThreadBase *thread = checkMmapThread_l(ioHandle);
+    IAfThreadBase* thread = checkMmapThread_l(ioHandle);
     if (thread == 0) {
         switch (audio_unique_id_get_use(ioHandle)) {
         case AUDIO_UNIQUE_ID_USE_OUTPUT:
@@ -3746,13 +3704,13 @@
 }
 
 // checkOutputThread_l() must be called with AudioFlinger::mLock held
-sp<AudioFlinger::ThreadBase> AudioFlinger::checkOutputThread_l(audio_io_handle_t ioHandle) const
+sp<IAfThreadBase> AudioFlinger::checkOutputThread_l(audio_io_handle_t ioHandle) const
 {
     if (audio_unique_id_get_use(ioHandle) != AUDIO_UNIQUE_ID_USE_OUTPUT) {
         return nullptr;
     }
 
-    sp<AudioFlinger::ThreadBase> thread = mPlaybackThreads.valueFor(ioHandle);
+    sp<IAfThreadBase> thread = mPlaybackThreads.valueFor(ioHandle);
     if (thread == nullptr) {
         thread = mMmapThreads.valueFor(ioHandle);
     }
@@ -3760,41 +3718,41 @@
 }
 
 // checkPlaybackThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
+IAfPlaybackThread* AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
 {
     return mPlaybackThreads.valueFor(output).get();
 }
 
 // checkMixerThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
+IAfPlaybackThread* AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
 {
-    PlaybackThread *thread = checkPlaybackThread_l(output);
-    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
+    IAfPlaybackThread * const thread = checkPlaybackThread_l(output);
+    return thread != nullptr && thread->type() != IAfThreadBase::DIRECT ? thread : nullptr;
 }
 
 // checkRecordThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
+IAfRecordThread* AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
 {
     return mRecordThreads.valueFor(input).get();
 }
 
 // checkMmapThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::MmapThread *AudioFlinger::checkMmapThread_l(audio_io_handle_t io) const
+IAfMmapThread* AudioFlinger::checkMmapThread_l(audio_io_handle_t io) const
 {
     return mMmapThreads.valueFor(io).get();
 }
 
 
 // checkPlaybackThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::VolumeInterface *AudioFlinger::getVolumeInterface_l(audio_io_handle_t output) const
+sp<VolumeInterface> AudioFlinger::getVolumeInterface_l(audio_io_handle_t output) const
 {
-    VolumeInterface *volumeInterface = mPlaybackThreads.valueFor(output).get();
+    sp<VolumeInterface> volumeInterface = mPlaybackThreads.valueFor(output).get();
     if (volumeInterface == nullptr) {
-        MmapThread *mmapThread = mMmapThreads.valueFor(output).get();
+        IAfMmapThread* const mmapThread = mMmapThreads.valueFor(output).get();
         if (mmapThread != nullptr) {
             if (mmapThread->isOutput()) {
-                MmapPlaybackThread *mmapPlaybackThread =
-                        static_cast<MmapPlaybackThread *>(mmapThread);
+                IAfMmapPlaybackThread* const mmapPlaybackThread =
+                        mmapThread->asIAfMmapPlaybackThread().get();
                 volumeInterface = mmapPlaybackThread;
             }
         }
@@ -3802,17 +3760,17 @@
     return volumeInterface;
 }
 
-Vector <AudioFlinger::VolumeInterface *> AudioFlinger::getAllVolumeInterfaces_l() const
+std::vector<sp<VolumeInterface>> AudioFlinger::getAllVolumeInterfaces_l() const
 {
-    Vector <VolumeInterface *> volumeInterfaces;
+    std::vector<sp<VolumeInterface>> volumeInterfaces;
     for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
-        volumeInterfaces.add(mPlaybackThreads.valueAt(i).get());
+        volumeInterfaces.push_back(mPlaybackThreads.valueAt(i).get());
     }
     for (size_t i = 0; i < mMmapThreads.size(); i++) {
         if (mMmapThreads.valueAt(i)->isOutput()) {
-            MmapPlaybackThread *mmapPlaybackThread =
-                    static_cast<MmapPlaybackThread *>(mMmapThreads.valueAt(i).get());
-            volumeInterfaces.add(mmapPlaybackThread);
+            IAfMmapPlaybackThread* const mmapPlaybackThread =
+                    mMmapThreads.valueAt(i)->asIAfMmapPlaybackThread().get();
+            volumeInterfaces.push_back(mmapPlaybackThread);
         }
     }
     return volumeInterfaces;
@@ -3839,14 +3797,14 @@
     // TODO Use a floor after wraparound.  This may need a mutex.
 }
 
-AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
+IAfPlaybackThread* AudioFlinger::primaryPlaybackThread_l() const
 {
     AutoMutex lock(mHardwareLock);
     if (mPrimaryHardwareDev == nullptr) {
         return nullptr;
     }
     for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
-        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
+        IAfPlaybackThread* const thread = mPlaybackThreads.valueAt(i).get();
         if(thread->isDuplicating()) {
             continue;
         }
@@ -3860,7 +3818,7 @@
 
 DeviceTypeSet AudioFlinger::primaryOutputDevice_l() const
 {
-    PlaybackThread *thread = primaryPlaybackThread_l();
+    IAfPlaybackThread* const thread = primaryPlaybackThread_l();
 
     if (thread == NULL) {
         return {};
@@ -3869,12 +3827,12 @@
     return thread->outDeviceTypes();
 }
 
-AudioFlinger::PlaybackThread *AudioFlinger::fastPlaybackThread_l() const
+IAfPlaybackThread* AudioFlinger::fastPlaybackThread_l() const
 {
     size_t minFrameCount = 0;
-    PlaybackThread *minThread = NULL;
+    IAfPlaybackThread* minThread = nullptr;
     for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
-        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
+        IAfPlaybackThread* const thread = mPlaybackThreads.valueAt(i).get();
         if (!thread->isDuplicating()) {
             size_t frameCount = thread->frameCountHAL();
             if (frameCount != 0 && (minFrameCount == 0 || frameCount < minFrameCount ||
@@ -3888,9 +3846,9 @@
     return minThread;
 }
 
-AudioFlinger::ThreadBase *AudioFlinger::hapticPlaybackThread_l() const {
+IAfThreadBase* AudioFlinger::hapticPlaybackThread_l() const {
     for (size_t i  = 0; i < mPlaybackThreads.size(); ++i) {
-        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
+        IAfPlaybackThread* const thread = mPlaybackThreads.valueAt(i).get();
         if (thread->hapticChannelMask() != AUDIO_CHANNEL_NONE) {
             return thread;
         }
@@ -3899,12 +3857,12 @@
 }
 
 void AudioFlinger::updateSecondaryOutputsForTrack_l(
-        PlaybackThread::Track* track,
-        PlaybackThread* thread,
+        IAfTrack* track,
+        IAfPlaybackThread* thread,
         const std::vector<audio_io_handle_t> &secondaryOutputs) const {
     TeePatches teePatches;
     for (audio_io_handle_t secondaryOutput : secondaryOutputs) {
-        PlaybackThread *secondaryThread = checkPlaybackThread_l(secondaryOutput);
+        IAfPlaybackThread* const secondaryThread = checkPlaybackThread_l(secondaryOutput);
         if (secondaryThread == nullptr) {
             ALOGE("no playback thread found for secondary output %d", thread->id());
             continue;
@@ -3930,10 +3888,10 @@
         // The frameCount should also not be smaller than the secondary thread min frame
         // count
         size_t minFrameCount = AudioSystem::calculateMinFrameCount(
-                    [&] { Mutex::Autolock _l(secondaryThread->mLock);
+                    [&] { Mutex::Autolock _l(secondaryThread->mutex());
                           return secondaryThread->latency_l(); }(),
-                    secondaryThread->mNormalFrameCount,
-                    secondaryThread->mSampleRate,
+                    secondaryThread->frameCount(), // normal frame count
+                    secondaryThread->sampleRate(),
                     track->sampleRate(),
                     track->getSpeed());
         frameCount = std::max(frameCount, minFrameCount);
@@ -3946,7 +3904,7 @@
             // 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 +3924,7 @@
         // 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(),
@@ -3989,14 +3947,14 @@
         patchTrack->setPeerProxy(patchRecord, true /* holdReference */);
         patchRecord->setPeerProxy(patchTrack, false /* holdReference */);
     }
-    track->setTeePatchesToUpdate(std::move(teePatches));
+    track->setTeePatchesToUpdate_l(std::move(teePatches));
 }
 
 sp<audioflinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
                                     audio_session_t triggerSession,
                                     audio_session_t listenerSession,
                                     const audioflinger::SyncEventCallback& callBack,
-                                    const wp<RefBase>& cookie)
+                                    const wp<IAfTrackBase>& cookie)
 {
     Mutex::Autolock _l(mLock);
 
@@ -4157,7 +4115,7 @@
         adjAttributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
         currentPid = callingPid;
     }
-    adjAttributionSource = AudioFlinger::checkAttributionSourcePackage(adjAttributionSource);
+    adjAttributionSource = afutils::checkAttributionSourcePackage(adjAttributionSource);
 
     ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d, factory %p",
           adjAttributionSource.pid, effectClient.get(), priority, sessionId, io,
@@ -4182,7 +4140,7 @@
             lStatus = BAD_VALUE;
             goto Exit;
         }
-        PlaybackThread *thread = checkPlaybackThread_l(io);
+        IAfPlaybackThread* const thread = checkPlaybackThread_l(io);
         if (thread == nullptr) {
             ALOGE("%s: invalid output %d specified for AUDIO_SESSION_OUTPUT_STAGE", __func__, io);
             lStatus = BAD_VALUE;
@@ -4279,7 +4237,7 @@
             sp<Client> client = registerPid(currentPid);
             ALOGV("%s device type %#x address %s", __func__, device.mType, device.getAddress());
             handle = mDeviceEffectManager->createEffect_l(
-                    &descOut, device, client, effectClient, mPatchPanel.patches_l(),
+                    &descOut, device, client, effectClient, mPatchPanel->patches_l(),
                     &enabledOut, &lStatus, probe, request.notifyFramesProcessed);
             if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
                 // remove local strong reference to Client with mClientLock held
@@ -4351,7 +4309,7 @@
                 }
                 const uint32_t sessionType =
                         mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId);
-                if ((sessionType & ThreadBase::EFFECT_SESSION) != 0) {
+                if ((sessionType & IAfThreadBase::EFFECT_SESSION) != 0) {
                     ALOGE("%s: effect %s io %d denied because session %d effect exists on io %d",
                           __func__, descOut.name, (int) io, (int) sessionId, (int) checkIo);
                     android_errorWriteLog(0x534e4554, "123237974");
@@ -4360,7 +4318,7 @@
                 }
             }
         }
-        ThreadBase *thread = checkRecordThread_l(io);
+        IAfThreadBase* thread = checkRecordThread_l(io);
         if (thread == NULL) {
             thread = checkPlaybackThread_l(io);
             if (thread == NULL) {
@@ -4376,7 +4334,7 @@
             // session and used it instead of creating a new one.
             sp<IAfEffectChain> chain = getOrphanEffectChain_l(sessionId);
             if (chain != 0) {
-                Mutex::Autolock _l2(thread->mLock);
+                Mutex::Autolock _l2(thread->mutex());
                 thread->addEffectChain_l(chain);
             }
         }
@@ -4385,9 +4343,9 @@
 
         // create effect on selected output thread
         bool pinned = !audio_is_global_session(sessionId) && isSessionAcquired_l(sessionId);
-        ThreadBase *oriThread = nullptr;
+        IAfThreadBase* oriThread = nullptr;
         if (hapticPlaybackRequired && thread->hapticChannelMask() == AUDIO_CHANNEL_NONE) {
-            ThreadBase *hapticThread = hapticPlaybackThread_l();
+            IAfThreadBase* const hapticThread = hapticPlaybackThread_l();
             if (hapticThread == nullptr) {
                 ALOGE("%s haptic thread not found while it is required", __func__);
                 lStatus = INVALID_OPERATION;
@@ -4450,26 +4408,26 @@
 status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput,
         audio_io_handle_t dstOutput)
 {
-    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
-            sessionId, srcOutput, dstOutput);
+    ALOGV("%s() session %d, srcOutput %d, dstOutput %d",
+            __func__, sessionId, srcOutput, dstOutput);
     Mutex::Autolock _l(mLock);
     if (srcOutput == dstOutput) {
-        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
+        ALOGW("%s() same dst and src outputs %d", __func__, dstOutput);
         return NO_ERROR;
     }
-    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
-    if (srcThread == NULL) {
-        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
+    IAfPlaybackThread* const srcThread = checkPlaybackThread_l(srcOutput);
+    if (srcThread == nullptr) {
+        ALOGW("%s() bad srcOutput %d", __func__, srcOutput);
         return BAD_VALUE;
     }
-    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
-    if (dstThread == NULL) {
-        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
+    IAfPlaybackThread* const dstThread = checkPlaybackThread_l(dstOutput);
+    if (dstThread == nullptr) {
+        ALOGW("%s() bad dstOutput %d", __func__, dstOutput);
         return BAD_VALUE;
     }
 
-    Mutex::Autolock _dl(dstThread->mLock);
-    Mutex::Autolock _sl(srcThread->mLock);
+    Mutex::Autolock _dl(dstThread->mutex());
+    Mutex::Autolock _sl(srcThread->mutex());
     return moveEffectChain_l(sessionId, srcThread, dstThread);
 }
 
@@ -4480,11 +4438,11 @@
 {
     Mutex::Autolock _l(mLock);
 
-    sp<ThreadBase> thread = getEffectThread_l(sessionId, effectId);
+    sp<IAfThreadBase> thread = getEffectThread_l(sessionId, effectId);
     if (thread == nullptr) {
       return;
     }
-    Mutex::Autolock _sl(thread->mLock);
+    Mutex::Autolock _sl(thread->mutex());
     sp<IAfEffectModule> effect = thread->getEffect_l(sessionId, effectId);
     thread->setEffectSuspended_l(&effect->desc().type, suspended, sessionId);
 }
@@ -4492,8 +4450,7 @@
 
 // moveEffectChain_l must be called with both srcThread and dstThread mLocks held
 status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId,
-                                   AudioFlinger::PlaybackThread *srcThread,
-                                   AudioFlinger::PlaybackThread *dstThread)
+        IAfPlaybackThread* srcThread, IAfPlaybackThread* dstThread)
 NO_THREAD_SAFETY_ANALYSIS // requires srcThread and dstThread locks
 {
     ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
@@ -4603,17 +4560,16 @@
 }
 
 status_t AudioFlinger::moveAuxEffectToIo(int EffectId,
-                                         const sp<PlaybackThread>& dstThread,
-                                         sp<PlaybackThread> *srcThread)
+        const sp<IAfPlaybackThread>& dstThread, sp<IAfPlaybackThread>* srcThread)
 {
     status_t status = NO_ERROR;
     Mutex::Autolock _l(mLock);
-    sp<PlaybackThread> thread =
-        static_cast<PlaybackThread *>(getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId).get());
+    const sp<IAfThreadBase> threadBase = getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
+    const sp<IAfPlaybackThread> thread = threadBase ? threadBase->asIAfPlaybackThread() : nullptr;
 
     if (EffectId != 0 && thread != 0 && dstThread != thread.get()) {
-        Mutex::Autolock _dl(dstThread->mLock);
-        Mutex::Autolock _sl(thread->mLock);
+        Mutex::Autolock _dl(dstThread->mutex());
+        Mutex::Autolock _sl(thread->mutex());
         sp<IAfEffectChain> srcChain = thread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
         sp<IAfEffectChain> dstChain;
         if (srcChain == 0) {
@@ -4652,7 +4608,7 @@
     return status;
 }
 
-bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
+bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() const
 NO_THREAD_SAFETY_ANALYSIS  // thread lock for getEffectChain_l.
 {
     if (mGlobalEffectEnableTime != 0 &&
@@ -4677,8 +4633,8 @@
     mGlobalEffectEnableTime = systemTime();
 
     for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
-        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
-        if (t->mType == ThreadBase::OFFLOAD) {
+        const sp<IAfPlaybackThread> t = mPlaybackThreads.valueAt(i);
+        if (t->type() == IAfThreadBase::OFFLOAD) {
             t->invalidateTracks(AUDIO_STREAM_MUSIC);
         }
     }
@@ -4732,6 +4688,55 @@
     return false;
 }
 
+// ----------------------------------------------------------------------------
+// from PatchPanel
+
+/* List connected audio ports and their attributes */
+status_t AudioFlinger::listAudioPorts(unsigned int* num_ports,
+        struct audio_port* ports) const
+{
+    Mutex::Autolock _l(mLock);
+    return mPatchPanel->listAudioPorts(num_ports, ports);
+}
+
+/* Get supported attributes for a given audio port */
+status_t AudioFlinger::getAudioPort(struct audio_port_v7* port) const {
+    const status_t status = AudioValidator::validateAudioPort(*port);
+    if (status != NO_ERROR) {
+        return status;
+    }
+
+    Mutex::Autolock _l(mLock);
+    return mPatchPanel->getAudioPort(port);
+}
+
+/* Connect a patch between several source and sink ports */
+status_t AudioFlinger::createAudioPatch(
+        const struct audio_patch* patch, audio_patch_handle_t* handle)
+{
+    const status_t status = AudioValidator::validateAudioPatch(*patch);
+    if (status != NO_ERROR) {
+        return status;
+    }
+
+    Mutex::Autolock _l(mLock);
+    return mPatchPanel->createAudioPatch(patch, handle);
+}
+
+/* Disconnect a patch */
+status_t AudioFlinger::releaseAudioPatch(audio_patch_handle_t handle)
+{
+    Mutex::Autolock _l(mLock);
+    return mPatchPanel->releaseAudioPatch(handle);
+}
+
+/* List connected audio ports and they attributes */
+status_t AudioFlinger::listAudioPatches(
+        unsigned int* num_patches, struct audio_patch* patches) const
+{
+    Mutex::Autolock _l(mLock);
+    return mPatchPanel->listAudioPatches(num_patches, patches);
+}
 
 // ----------------------------------------------------------------------------
 
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index ba6200f..12180f9 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -74,6 +74,7 @@
 #include <media/DeviceDescriptorBase.h>
 #include <media/ExtendedAudioBufferProvider.h>
 #include <media/VolumeShaper.h>
+#include <mediautils/BatteryNotifier.h>
 #include <mediautils/ServiceUtilities.h>
 #include <mediautils/SharedMemoryAllocator.h>
 #include <mediautils/Synchronization.h>
@@ -97,10 +98,12 @@
 #include <timing/SynchronizedRecordState.h>
 
 #include <datapath/AudioHwDevice.h>
+#include <datapath/AudioStreamIn.h>
 #include <datapath/AudioStreamOut.h>
 #include <datapath/SpdifStreamOut.h>
 #include <datapath/ThreadMetrics.h>
 #include <datapath/TrackMetrics.h>
+#include <datapath/VolumeInterface.h>
 #include <fastpath/FastCapture.h>
 #include <fastpath/FastMixer.h>
 #include <media/nbaio/NBAIO.h>
@@ -118,9 +121,18 @@
 #include "android/media/BnEffect.h"
 
 #include "Client.h"
+#include "ResamplerBufferProvider.h"
 
 // include AudioFlinger component interfaces
+#include "IAfPatchPanel.h"  // this should be listed before other IAf* interfaces.
 #include "IAfEffect.h"
+#include "IAfThread.h"
+#include "IAfTrack.h"
+
+// Classes that depend on IAf* interfaces but are not cross-dependent.
+#include "PatchCommandThread.h"
+#include "DeviceEffectManager.h"
+#include "MelReporter.h"
 
 namespace android {
 
@@ -134,212 +146,332 @@
 class FastMixer;
 class IAudioManager;
 class PassthruBufferProvider;
-class RecordBufferConverter;
 class ServerProxy;
 
 // ----------------------------------------------------------------------------
 
 static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
 
-#define INCLUDING_FROM_AUDIOFLINGER_H
-
 using android::content::AttributionSourceState;
 
-class AudioFlinger : public AudioFlingerServerAdapter::Delegate
+struct stream_type_t {
+    float volume = 1.f;
+    bool mute = false;
+};
+
+class AudioFlinger
+    : public AudioFlingerServerAdapter::Delegate  // IAudioFlinger client interface
+    , public IAfClientCallback
+    , public IAfDeviceEffectManagerCallback
+    , public IAfMelReporterCallback
+    , public IAfPatchPanelCallback
+    , public IAfThreadCallback
 {
     friend class sp<AudioFlinger>;
-    friend class Client; // removeClient_l();
 public:
     static void instantiate() ANDROID_API;
 
-    static AttributionSourceState checkAttributionSourcePackage(
-        const AttributionSourceState& attributionSource);
+private:
 
-    status_t dump(int fd, const Vector<String16>& args) override;
+    // ---- begin IAudioFlinger interface
 
-    // IAudioFlinger interface, in binder opcode order
+    status_t dump(int fd, const Vector<String16>& args) final;
+
     status_t createTrack(const media::CreateTrackRequest& input,
-                         media::CreateTrackResponse& output) override;
+                         media::CreateTrackResponse& output) final;
 
     status_t createRecord(const media::CreateRecordRequest& input,
-                          media::CreateRecordResponse& output) override;
+                          media::CreateRecordResponse& output) final;
 
-    virtual     uint32_t    sampleRate(audio_io_handle_t ioHandle) const;
-    virtual     audio_format_t format(audio_io_handle_t output) const;
-    virtual     size_t      frameCount(audio_io_handle_t ioHandle) const;
-    virtual     size_t      frameCountHAL(audio_io_handle_t ioHandle) const;
-    virtual     uint32_t    latency(audio_io_handle_t output) const;
+    uint32_t sampleRate(audio_io_handle_t ioHandle) const final;
+    audio_format_t format(audio_io_handle_t output) const final;
+    size_t frameCount(audio_io_handle_t ioHandle) const final;
+    size_t frameCountHAL(audio_io_handle_t ioHandle) const final;
+    uint32_t latency(audio_io_handle_t output) const final;
 
-    virtual     status_t    setMasterVolume(float value);
-    virtual     status_t    setMasterMute(bool muted);
-
-    virtual     float       masterVolume() const;
-    virtual     bool        masterMute() const;
+    status_t setMasterVolume(float value) final;
+    status_t setMasterMute(bool muted) final;
+    float masterVolume() const final;
+    bool masterMute() const final;
 
     // Balance value must be within -1.f (left only) to 1.f (right only) inclusive.
-                status_t    setMasterBalance(float balance) override;
-                status_t    getMasterBalance(float *balance) const override;
+    status_t setMasterBalance(float balance) final;
+    status_t getMasterBalance(float* balance) const final;
 
-    virtual     status_t    setStreamVolume(audio_stream_type_t stream, float value,
-                                            audio_io_handle_t output);
-    virtual     status_t    setStreamMute(audio_stream_type_t stream, bool muted);
+    status_t setStreamVolume(audio_stream_type_t stream, float value,
+            audio_io_handle_t output) final;
+    status_t setStreamMute(audio_stream_type_t stream, bool muted) final;
 
-    virtual     float       streamVolume(audio_stream_type_t stream,
-                                         audio_io_handle_t output) const;
-    virtual     bool        streamMute(audio_stream_type_t stream) const;
+    float streamVolume(audio_stream_type_t stream,
+            audio_io_handle_t output) const final;
+    bool streamMute(audio_stream_type_t stream) const final;
 
-    virtual     status_t    setMode(audio_mode_t mode);
+    status_t setMode(audio_mode_t mode) final;
 
-    virtual     status_t    setMicMute(bool state);
-    virtual     bool        getMicMute() const;
+    status_t setMicMute(bool state) final;
+    bool getMicMute() const final;
 
-    virtual     void        setRecordSilenced(audio_port_handle_t portId, bool silenced);
+    void setRecordSilenced(audio_port_handle_t portId, bool silenced) final;
 
-    virtual     status_t    setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
-    virtual     String8     getParameters(audio_io_handle_t ioHandle, const String8& keys) const;
+    status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) final;
+    String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const final;
 
-    virtual     void        registerClient(const sp<media::IAudioFlingerClient>& client);
+    void registerClient(const sp<media::IAudioFlingerClient>& client) final;
+    size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
+            audio_channel_mask_t channelMask) const final;
 
-    virtual     size_t      getInputBufferSize(uint32_t sampleRate, audio_format_t format,
-                                               audio_channel_mask_t channelMask) const;
+    status_t openOutput(const media::OpenOutputRequest& request,
+            media::OpenOutputResponse* response) final;
 
-    virtual status_t openOutput(const media::OpenOutputRequest& request,
-                                media::OpenOutputResponse* response);
+    audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
+            audio_io_handle_t output2) final;
 
-    virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
-                                                  audio_io_handle_t output2);
+    status_t closeOutput(audio_io_handle_t output) final;
 
-    virtual status_t closeOutput(audio_io_handle_t output);
+    status_t suspendOutput(audio_io_handle_t output) final;
 
-    virtual status_t suspendOutput(audio_io_handle_t output);
+    status_t restoreOutput(audio_io_handle_t output) final;
 
-    virtual status_t restoreOutput(audio_io_handle_t output);
+    status_t openInput(const media::OpenInputRequest& request,
+            media::OpenInputResponse* response) final;
 
-    virtual status_t openInput(const media::OpenInputRequest& request,
-                               media::OpenInputResponse* response);
+    status_t closeInput(audio_io_handle_t input) final;
 
-    virtual status_t closeInput(audio_io_handle_t input);
+    status_t setVoiceVolume(float volume) final;
 
-    virtual status_t setVoiceVolume(float volume);
+    status_t getRenderPosition(uint32_t* halFrames, uint32_t* dspFrames,
+            audio_io_handle_t output) const final;
 
-    virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
-                                       audio_io_handle_t output) const;
-
-    virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const;
+    uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const final;
 
     // This is the binder API.  For the internal API see nextUniqueId().
-    virtual audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use);
+    audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use) final;
 
-    void acquireAudioSessionId(audio_session_t audioSession, pid_t pid, uid_t uid) override;
+    void acquireAudioSessionId(audio_session_t audioSession, pid_t pid, uid_t uid) final;
 
-    virtual void releaseAudioSessionId(audio_session_t audioSession, pid_t pid);
+    void releaseAudioSessionId(audio_session_t audioSession, pid_t pid) final;
 
-    virtual status_t queryNumberEffects(uint32_t *numEffects) const;
+    status_t queryNumberEffects(uint32_t* numEffects) const final;
 
-    virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const;
+    status_t queryEffect(uint32_t index, effect_descriptor_t* descriptor) const final;
 
-    virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid,
-                                         const effect_uuid_t *pTypeUuid,
-                                         uint32_t preferredTypeFlag,
-                                         effect_descriptor_t *descriptor) const;
+    status_t getEffectDescriptor(const effect_uuid_t* pUuid,
+            const effect_uuid_t* pTypeUuid,
+            uint32_t preferredTypeFlag,
+            effect_descriptor_t* descriptor) const final;
 
-    virtual status_t createEffect(const media::CreateEffectRequest& request,
-                                  media::CreateEffectResponse* response);
+    status_t createEffect(const media::CreateEffectRequest& request,
+            media::CreateEffectResponse* response) final;
 
-    virtual status_t moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput,
-                        audio_io_handle_t dstOutput);
+    status_t moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput,
+            audio_io_handle_t dstOutput) final;
 
-            void setEffectSuspended(int effectId,
-                                    audio_session_t sessionId,
-                                    bool suspended) override;
+    void setEffectSuspended(int effectId,
+            audio_session_t sessionId,
+            bool suspended) final;
 
-    virtual audio_module_handle_t loadHwModule(const char *name);
+    audio_module_handle_t loadHwModule(const char* name) final;
 
-    virtual uint32_t getPrimaryOutputSamplingRate();
-    virtual size_t getPrimaryOutputFrameCount();
+    uint32_t getPrimaryOutputSamplingRate() const final;
+    size_t getPrimaryOutputFrameCount() const final;
 
-    virtual status_t setLowRamDevice(bool isLowRamDevice, int64_t totalMemory) override;
-
-    /* List available audio ports and their attributes */
-    virtual status_t listAudioPorts(unsigned int *num_ports,
-                                    struct audio_port *ports);
+    status_t setLowRamDevice(bool isLowRamDevice, int64_t totalMemory) final;
 
     /* Get attributes for a given audio port */
-    virtual status_t getAudioPort(struct audio_port_v7 *port);
+    status_t getAudioPort(struct audio_port_v7* port) const final;
 
     /* Create an audio patch between several source and sink ports */
-    virtual status_t createAudioPatch(const struct audio_patch *patch,
-                                       audio_patch_handle_t *handle);
+    status_t createAudioPatch(const struct audio_patch *patch,
+            audio_patch_handle_t* handle) final;
 
     /* Release an audio patch */
-    virtual status_t releaseAudioPatch(audio_patch_handle_t handle);
+    status_t releaseAudioPatch(audio_patch_handle_t handle) final;
 
     /* List existing audio patches */
-    virtual status_t listAudioPatches(unsigned int *num_patches,
-                                      struct audio_patch *patches);
+    status_t listAudioPatches(unsigned int* num_patches,
+            struct audio_patch* patches) const final;
 
     /* Set audio port configuration */
-    virtual status_t setAudioPortConfig(const struct audio_port_config *config);
+    status_t setAudioPortConfig(const struct audio_port_config* config) final;
 
     /* Get the HW synchronization source used for an audio session */
-    virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId);
+    audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId) final;
 
     /* Indicate JAVA services are ready (scheduling, power management ...) */
-    virtual status_t systemReady();
-    virtual status_t audioPolicyReady() { mAudioPolicyReady.store(true); return NO_ERROR; }
-            bool isAudioPolicyReady() const { return mAudioPolicyReady.load(); }
+    status_t systemReady() final;
+    status_t audioPolicyReady() final { mAudioPolicyReady.store(true); return NO_ERROR; }
 
+    status_t getMicrophones(std::vector<media::MicrophoneInfoFw>* microphones) const final;
 
-    virtual status_t getMicrophones(std::vector<media::MicrophoneInfoFw> *microphones);
+    status_t setAudioHalPids(const std::vector<pid_t>& pids) final;
 
-    virtual status_t setAudioHalPids(const std::vector<pid_t>& pids);
+    status_t setVibratorInfos(const std::vector<media::AudioVibratorInfo>& vibratorInfos) final;
 
-    virtual status_t setVibratorInfos(const std::vector<media::AudioVibratorInfo>& vibratorInfos);
+    status_t updateSecondaryOutputs(
+            const TrackSecondaryOutputsMap& trackSecondaryOutputs) final;
 
-    virtual status_t updateSecondaryOutputs(
-            const TrackSecondaryOutputsMap& trackSecondaryOutputs);
-
-    virtual status_t getMmapPolicyInfos(
+    status_t getMmapPolicyInfos(
             media::audio::common::AudioMMapPolicyType policyType,
-            std::vector<media::audio::common::AudioMMapPolicyInfo> *policyInfos);
+            std::vector<media::audio::common::AudioMMapPolicyInfo>* policyInfos) final;
 
-    virtual int32_t getAAudioMixerBurstCount();
+    int32_t getAAudioMixerBurstCount() const final;
 
-    virtual int32_t getAAudioHardwareBurstMinUsec();
+    int32_t getAAudioHardwareBurstMinUsec() const final;
 
-    virtual status_t setDeviceConnectedState(const struct audio_port_v7 *port,
-                                             media::DeviceConnectedState state);
+    status_t setDeviceConnectedState(const struct audio_port_v7* port,
+            media::DeviceConnectedState state) final;
 
-    virtual status_t setSimulateDeviceConnections(bool enabled);
+    status_t setSimulateDeviceConnections(bool enabled) final;
 
-    virtual status_t setRequestedLatencyMode(
-            audio_io_handle_t output, audio_latency_mode_t mode);
+    status_t setRequestedLatencyMode(
+            audio_io_handle_t output, audio_latency_mode_t mode) final;
 
-    virtual status_t getSupportedLatencyModes(audio_io_handle_t output,
-            std::vector<audio_latency_mode_t>* modes);
+    status_t getSupportedLatencyModes(audio_io_handle_t output,
+            std::vector<audio_latency_mode_t>* modes) const final;
 
-    virtual status_t setBluetoothVariableLatencyEnabled(bool enabled);
+    status_t setBluetoothVariableLatencyEnabled(bool enabled) final;
 
-    virtual status_t isBluetoothVariableLatencyEnabled(bool* enabled);
+    status_t isBluetoothVariableLatencyEnabled(bool* enabled) const final;
 
-    virtual status_t supportsBluetoothVariableLatency(bool* support);
+    status_t supportsBluetoothVariableLatency(bool* support) const final;
 
-    virtual status_t getSoundDoseInterface(const sp<media::ISoundDoseCallback>& callback,
-                                           sp<media::ISoundDose>* soundDose);
+    status_t getSoundDoseInterface(const sp<media::ISoundDoseCallback>& callback,
+            sp<media::ISoundDose>* soundDose) const final;
 
-    status_t invalidateTracks(const std::vector<audio_port_handle_t>& portIds) override;
+    status_t invalidateTracks(const std::vector<audio_port_handle_t>& portIds) final;
 
-    virtual status_t getAudioPolicyConfig(media::AudioPolicyConfig* config);
+    status_t getAudioPolicyConfig(media::AudioPolicyConfig* config) final;
 
     status_t onTransactWrapper(TransactionCode code, const Parcel& data, uint32_t flags,
-        const std::function<status_t()>& delegate) override;
+            const std::function<status_t()>& delegate) final;
 
-    // end of IAudioFlinger interface
+    // ---- end of IAudioFlinger interface
 
-    sp<NBLog::Writer>   newWriter_l(size_t size, const char *name);
-    void                unregisterWriter(const sp<NBLog::Writer>& writer);
+    // ---- begin IAfClientCallback interface
+
+    Mutex& clientMutex() const final { return mClientLock; }
+    void removeClient_l(pid_t pid) final;
+    void removeNotificationClient(pid_t pid) final;
+    status_t moveAuxEffectToIo(
+            int effectId,
+            const sp<IAfPlaybackThread>& dstThread,
+            sp<IAfPlaybackThread>* srcThread) final;
+
+    // ---- end of IAfClientCallback interface
+
+    // ---- begin IAfDeviceEffectManagerCallback interface
+
+    // also used by IAfThreadCallback
+    bool isAudioPolicyReady() const final { return mAudioPolicyReady.load(); }
+    // below also used by IAfMelReporterCallback, IAfPatchPanelCallback
+    const sp<PatchCommandThread>& getPatchCommandThread() final { return mPatchCommandThread; }
+    status_t addEffectToHal(
+            const struct audio_port_config* device, const sp<EffectHalInterface>& effect) final;
+    status_t removeEffectFromHal(
+            const struct audio_port_config* device, const sp<EffectHalInterface>& effect) final;
+
+    // ---- end of IAfDeviceEffectManagerCallback interface
+
+    // ---- begin IAfMelReporterCallback interface
+
+    // below also used by IAfThreadCallback
+    Mutex& mutex() const final { return mLock; }
+    sp<IAfThreadBase> checkOutputThread_l(audio_io_handle_t ioHandle) const final REQUIRES(mLock);
+
+    // ---- end of IAfMelReporterCallback interface
+
+    // ---- begin IAfPatchPanelCallback interface
+
+    void closeThreadInternal_l(const sp<IAfPlaybackThread>& thread) final;
+    void closeThreadInternal_l(const sp<IAfRecordThread>& thread) final;
+    // return thread associated with primary hardware device, or NULL
+    IAfPlaybackThread* primaryPlaybackThread_l() const final;
+    IAfPlaybackThread* checkPlaybackThread_l(audio_io_handle_t output) const final;
+    IAfRecordThread* checkRecordThread_l(audio_io_handle_t input) const final;
+    IAfMmapThread* checkMmapThread_l(audio_io_handle_t io) const final;
+    void lock() const final ACQUIRE(mLock) { mLock.lock(); }
+    void unlock() const final RELEASE(mLock) { mLock.unlock(); }
+    sp<IAfThreadBase> openInput_l(audio_module_handle_t module,
+            audio_io_handle_t* input,
+            audio_config_t* config,
+            audio_devices_t device,
+            const char* address,
+            audio_source_t source,
+            audio_input_flags_t flags,
+            audio_devices_t outputDevice,
+            const String8& outputDeviceAddress) final;
+    sp<IAfThreadBase> openOutput_l(audio_module_handle_t module,
+            audio_io_handle_t* output,
+            audio_config_t* halConfig,
+            audio_config_base_t* mixerConfig,
+            audio_devices_t deviceType,
+            const String8& address,
+            audio_output_flags_t flags) final;
+    const DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*>&
+            getAudioHwDevs_l() const final { return mAudioHwDevs; }
+    void updateDownStreamPatches_l(const struct audio_patch* patch,
+            const std::set<audio_io_handle_t>& streams) final;
+    void updateOutDevicesForRecordThreads_l(const DeviceDescriptorBaseVector& devices) final;
+
+    // ---- end of IAfPatchPanelCallback interface
+
+    // ----- begin IAfThreadCallback interface
+
+    bool isNonOffloadableGlobalEffectEnabled_l() const final;
+    bool btNrecIsOff() const final { return mBtNrecIsOff.load(); }
+    float masterVolume_l() const final;
+    bool masterMute_l() const final;
+    float getMasterBalance_l() const;
+    // no range check, AudioFlinger::mLock held
+    bool streamMute_l(audio_stream_type_t stream) const final { return mStreamTypes[stream].mute; }
+    audio_mode_t getMode() const final { return mMode; }
+    bool isLowRamDevice() const final { return mIsLowRamDevice; }
+    uint32_t getScreenState() const final { return mScreenState; }
+
+    std::optional<media::AudioVibratorInfo> getDefaultVibratorInfo_l() const final;
+    const sp<IAfPatchPanel>& getPatchPanel() const final { return mPatchPanel; }
+    const sp<MelReporter>& getMelReporter() const final { return mMelReporter; }
+    const sp<EffectsFactoryHalInterface>& getEffectsFactoryHal() const final {
+        return mEffectsFactoryHal;
+    }
+    sp<IAudioManager> getOrCreateAudioManager() final;
+
+    // Called when the last effect handle on an effect instance is removed. If this
+    // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated
+    // and removed from mOrphanEffectChains if it does not contain any effect.
+    // Return true if the effect was found in mOrphanEffectChains, false otherwise.
+    bool updateOrphanEffectChains(const sp<IAfEffectModule>& effect) final;
+
+    status_t moveEffectChain_l(audio_session_t sessionId,
+            IAfPlaybackThread* srcThread, IAfPlaybackThread* dstThread) final;
+
+    // This is a helper that is called during incoming binder calls.
+    // Requests media.log to start merging log buffers
+    void requestLogMerge() final;
+    sp<NBLog::Writer> newWriter_l(size_t size, const char *name) final;
+    void unregisterWriter(const sp<NBLog::Writer>& writer) final;
+
+    sp<audioflinger::SyncEvent> createSyncEvent(AudioSystem::sync_event_t type,
+            audio_session_t triggerSession,
+            audio_session_t listenerSession,
+            const audioflinger::SyncEventCallback& callBack,
+            const wp<IAfTrackBase>& cookie) final;
+
+    void ioConfigChanged(audio_io_config_event_t event,
+            const sp<AudioIoDescriptor>& ioDesc,
+            pid_t pid = 0) final;
+    void onNonOffloadableGlobalEffectEnable() final;
+    void onSupportedLatencyModesChanged(
+            audio_io_handle_t output, const std::vector<audio_latency_mode_t>& modes) final;
+
+    // ---- end of IAfThreadCallback interface
+
+    /* List available audio ports and their attributes */
+    status_t listAudioPorts(unsigned int* num_ports, struct audio_port* ports) const;
+
     sp<EffectsFactoryHalInterface> getEffectsFactory();
 
+public:
     status_t openMmapStream(MmapStreamInterface::stream_direction_t direction,
                             const audio_attributes_t *attr,
                             audio_config_base_t *config,
@@ -354,16 +486,6 @@
         const sp<os::ExternalVibration>& externalVibration);
     static void onExternalVibrationStop(const sp<os::ExternalVibration>& externalVibration);
 
-    status_t addEffectToHal(
-            const struct audio_port_config *device, const sp<EffectHalInterface>& effect);
-    status_t removeEffectFromHal(
-            const struct audio_port_config *device, const sp<EffectHalInterface>& effect);
-
-    void updateDownStreamPatches_l(const struct audio_patch *patch,
-                                   const std::set<audio_io_handle_t>& streams);
-
-    std::optional<media::AudioVibratorInfo> getDefaultVibratorInfo_l();
-
 private:
     // FIXME The 400 is temporarily too high until a leak of writers in media.log is fixed.
     static const size_t kLogMemorySize = 400 * 1024;
@@ -373,50 +495,15 @@
     Vector< sp<NBLog::Writer> > mUnregisteredWriters;
     Mutex               mUnregisteredWritersLock;
 
-public:
-    // Life cycle of gAudioFlinger and AudioFlinger:
-    //
-    // AudioFlinger is created once and survives until audioserver crashes
-    // irrespective of sp<> and wp<> as it is refcounted by ServiceManager and we
-    // don't issue a ServiceManager::tryUnregisterService().
-    //
-    // gAudioFlinger is an atomic pointer set on AudioFlinger::onFirstRef().
-    // After this is set, it is safe to obtain a wp<> or sp<> from it as the
-    // underlying object does not go away.
-    //
-    // Note: For most inner classes, it is acceptable to hold a reference to the outer
-    // AudioFlinger instance as creation requires AudioFlinger to exist in the first place.
-    //
-    // An atomic here ensures underlying writes have completed before setting
-    // the pointer. Access by memory_order_seq_cst.
-    //
-
-    static inline std::atomic<AudioFlinger *> gAudioFlinger = nullptr;
-
-    sp<audioflinger::SyncEvent> createSyncEvent(AudioSystem::sync_event_t type,
-                                        audio_session_t triggerSession,
-                                        audio_session_t listenerSession,
-                                        const audioflinger::SyncEventCallback& callBack,
-                                        const wp<RefBase>& cookie);
-
-    bool        btNrecIsOff() const { return mBtNrecIsOff.load(); }
-
-    void             lock() ACQUIRE(mLock) { mLock.lock(); }
-    void             unlock() RELEASE(mLock) { mLock.unlock(); }
-
-private:
-
-               audio_mode_t getMode() const { return mMode; }
-
                             AudioFlinger() ANDROID_API;
-    virtual                 ~AudioFlinger();
+    ~AudioFlinger() override;
 
     // call in any IAudioFlinger method that accesses mPrimaryHardwareDev
     status_t                initCheck() const { return mPrimaryHardwareDev == NULL ?
                                                         NO_INIT : NO_ERROR; }
 
     // RefBase
-    virtual     void        onFirstRef();
+    void onFirstRef() override;
 
     AudioHwDevice*          findSuitableHwDev_l(audio_module_handle_t module,
                                                 audio_devices_t deviceType);
@@ -426,6 +513,10 @@
     // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS.
     static const bool kEnableExtendedChannels = true;
 
+public:
+    // Remove this when Oboeservice is updated to obtain handle directly.
+    static inline std::atomic<AudioFlinger*> gAudioFlinger = nullptr;
+
     // Returns true if channel mask is permitted for the PCM sink in the MixerThread
     static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) {
         switch (audio_channel_mask_get_representation(channelMask)) {
@@ -478,27 +569,19 @@
     // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs
     static nsecs_t          mStandbyTimeInNsecs;
 
-    // incremented by 2 when screen state changes, bit 0 == 1 means "off"
-    // AudioFlinger::setParameters() updates, other threads read w/o lock
-    static uint32_t         mScreenState;
-
-    // Internal dump utilities.
-    static const int kDumpLockTimeoutNs = 1 * NANOS_PER_SECOND;
-public:
-    // TODO(b/288339104) extract to afutils
-    static bool dumpTryLock(Mutex& mutex);
 private:
+
+    // incremented by 2 when screen state changes, bit 0 == 1 means "off"
+    // AudioFlinger::setParameters() updates with mLock.
+    std::atomic_uint32_t mScreenState{};
+
     void dumpPermissionDenial(int fd, const Vector<String16>& args);
     void dumpClients(int fd, const Vector<String16>& args);
     void dumpInternals(int fd, const Vector<String16>& args);
 
     SimpleLog mThreadLog{16}; // 16 Thread history limit
 
-public:
-    // TODO(b/288339104)
-    class ThreadBase;
-private:
-    void dumpToThreadLog_l(const sp<ThreadBase> &thread);
+    void dumpToThreadLog_l(const sp<IAfThreadBase>& thread);
 
     // --- Notification Client ---
     class NotificationClient : public IBinder::DeathRecipient {
@@ -554,66 +637,6 @@
 
     const sp<MediaLogNotifier> mMediaLogNotifier;
 
-    // This is a helper that is called during incoming binder calls.
-    // Requests media.log to start merging log buffers
-    void requestLogMerge();
-
-    // TODO(b/288339104) replace these forward declaration classes with interfaces.
-public:
-    class RecordThread;
-    class PlaybackThread;
-    class MixerThread;
-    class DirectOutputThread;
-    class OffloadThread;
-    class DuplicatingThread;
-    class AsyncCallbackThread;
-    class BitPerfectThread;
-private:
-    class Track;
-    class RecordTrack;
-    class DeviceEffectManager;
-    // TODO(b/288339104) these should be separate files
-public:
-    class PatchPanel;
-    class DeviceEffectManagerCallback;
-private:
-    struct AudioStreamIn;
-    struct TeePatch;
-    using TeePatches = std::vector<TeePatch>;
-
-
-    struct  stream_type_t {
-        stream_type_t()
-            :   volume(1.0f),
-                mute(false)
-        {
-        }
-        float       volume;
-        bool        mute;
-    };
-
-    // Abstraction for the Audio Source for the RecordThread (HAL or PassthruPatchRecord).
-    struct Source
-    {
-        virtual ~Source() = default;
-        // The following methods have the same signatures as in StreamHalInterface.
-        virtual status_t read(void *buffer, size_t bytes, size_t *read) = 0;
-        virtual status_t getCapturePosition(int64_t *frames, int64_t *time) = 0;
-        virtual status_t standby() = 0;
-    };
-
-    // --- PlaybackThread ---
-
-#include "Threads.h"
-
-#include "PatchPanel.h"
-
-#include "PatchCommandThread.h"
-
-#include "DeviceEffectManager.h"
-
-#include "MelReporter.h"
-
     // Find io handle by session id.
     // Preference is given to an io handle with a matching effect chain to session id.
     // If none found, AUDIO_IO_HANDLE_NONE is returned.
@@ -626,7 +649,7 @@
             const uint32_t sessionType = threads.valueAt(i)->hasAudioSession(sessionId);
             if (sessionType != 0) {
                 io = threads.keyAt(i);
-                if ((sessionType & AudioFlinger::ThreadBase::EFFECT_SESSION) != 0) {
+                if ((sessionType & IAfThreadBase::EFFECT_SESSION) != 0) {
                     break; // effect chain here.
                 }
             }
@@ -634,67 +657,15 @@
         return io;
     }
 
-    // Mmap stream control interface implementation. Each MmapThreadHandle controls one
-    // MmapPlaybackThread or MmapCaptureThread instance.
-    class MmapThreadHandle : public MmapStreamInterface {
-    public:
-        explicit            MmapThreadHandle(const sp<MmapThread>& thread);
-        virtual             ~MmapThreadHandle();
+    IAfThreadBase* checkThread_l(audio_io_handle_t ioHandle) const;
+    IAfPlaybackThread* checkMixerThread_l(audio_io_handle_t output) const;
 
-        // MmapStreamInterface virtuals
-        virtual status_t createMmapBuffer(int32_t minSizeFrames,
-                                          struct audio_mmap_buffer_info *info);
-        virtual status_t getMmapPosition(struct audio_mmap_position *position);
-        virtual status_t getExternalPosition(uint64_t *position, int64_t *timeNanos);
-        virtual status_t start(const AudioClient& client,
-                               const audio_attributes_t *attr,
-                               audio_port_handle_t *handle);
-        virtual status_t stop(audio_port_handle_t handle);
-        virtual status_t standby();
-                status_t reportData(const void* buffer, size_t frameCount) override;
+              sp<VolumeInterface> getVolumeInterface_l(audio_io_handle_t output) const;
+              std::vector<sp<VolumeInterface>> getAllVolumeInterfaces_l() const;
 
-    private:
-        const sp<MmapThread> mThread;
-    };
 
-              ThreadBase *checkThread_l(audio_io_handle_t ioHandle) const;
-              sp<AudioFlinger::ThreadBase> checkOutputThread_l(audio_io_handle_t ioHandle) const
-                      REQUIRES(mLock);
-              PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const;
-              MixerThread *checkMixerThread_l(audio_io_handle_t output) const;
-              RecordThread *checkRecordThread_l(audio_io_handle_t input) const;
-              MmapThread *checkMmapThread_l(audio_io_handle_t io) const;
-              VolumeInterface *getVolumeInterface_l(audio_io_handle_t output) const;
-              Vector <VolumeInterface *> getAllVolumeInterfaces_l() const;
-
-              sp<ThreadBase> openInput_l(audio_module_handle_t module,
-                                           audio_io_handle_t *input,
-                                           audio_config_t *config,
-                                           audio_devices_t device,
-                                           const char* address,
-                                           audio_source_t source,
-                                           audio_input_flags_t flags,
-                                           audio_devices_t outputDevice,
-                                           const String8& outputDeviceAddress);
-              sp<ThreadBase> openOutput_l(audio_module_handle_t module,
-                                          audio_io_handle_t *output,
-                                          audio_config_t *halConfig,
-                                          audio_config_base_t *mixerConfig,
-                                          audio_devices_t deviceType,
-                                          const String8& address,
-                                          audio_output_flags_t flags);
-
-              void closeOutputFinish(const sp<PlaybackThread>& thread);
-              void closeInputFinish(const sp<RecordThread>& thread);
-
-              // no range check, AudioFlinger::mLock held
-              bool streamMute_l(audio_stream_type_t stream) const
-                                { return mStreamTypes[stream].mute; }
-              void ioConfigChanged(audio_io_config_event_t event,
-                                   const sp<AudioIoDescriptor>& ioDesc,
-                                   pid_t pid = 0);
-              void onSupportedLatencyModesChanged(
-                    audio_io_handle_t output, const std::vector<audio_latency_mode_t>& modes);
+    void closeOutputFinish(const sp<IAfPlaybackThread>& thread);
+    void closeInputFinish(const sp<IAfRecordThread>& thread);
 
               // Allocate an audio_unique_id_t.
               // Specific types are audio_io_handle_t, audio_session_t, effect ID (int),
@@ -707,37 +678,24 @@
               //       Thus it may fail by returning an ID of the wrong sign,
               //       or by returning a non-unique ID.
               // This is the internal API.  For the binder API see newAudioUniqueId().
-              audio_unique_id_t nextUniqueId(audio_unique_id_use_t use);
-
-              status_t moveEffectChain_l(audio_session_t sessionId,
-                                     PlaybackThread *srcThread,
-                                     PlaybackThread *dstThread);
-
-              status_t moveAuxEffectToIo(int EffectId,
-                                         const sp<PlaybackThread>& dstThread,
-                                         sp<PlaybackThread> *srcThread);
+    // used by IAfDeviceEffectManagerCallback, IAfPatchPanelCallback, IAfThreadCallback
+    audio_unique_id_t nextUniqueId(audio_unique_id_use_t use) final;
 
               // return thread associated with primary hardware device, or NULL
-              PlaybackThread *primaryPlaybackThread_l() const;
               DeviceTypeSet primaryOutputDevice_l() const;
 
               // return the playback thread with smallest HAL buffer size, and prefer fast
-              PlaybackThread *fastPlaybackThread_l() const;
+              IAfPlaybackThread* fastPlaybackThread_l() const;
 
-              sp<ThreadBase> getEffectThread_l(audio_session_t sessionId, int effectId);
+              sp<IAfThreadBase> getEffectThread_l(audio_session_t sessionId, int effectId);
 
-              ThreadBase *hapticPlaybackThread_l() const;
+              IAfThreadBase* hapticPlaybackThread_l() const;
 
               void updateSecondaryOutputsForTrack_l(
-                      PlaybackThread::Track* track,
-                      PlaybackThread* thread,
+                      IAfTrack* track,
+                      IAfPlaybackThread* thread,
                       const std::vector<audio_io_handle_t>& secondaryOutputs) const;
 
-
-                void        removeClient_l(pid_t pid);
-                void        removeNotificationClient(pid_t pid);
-                bool isNonOffloadableGlobalEffectEnabled_l();
-                void onNonOffloadableGlobalEffectEnable();
                 bool isSessionAcquired_l(audio_session_t audioSession);
 
                 // Store an effect chain to mOrphanEffectChains keyed vector.
@@ -751,49 +709,13 @@
                 // Get an effect chain for the specified session in mOrphanEffectChains and remove
                 // it if found. Returns 0 if not found (this is the most common case).
                 sp<IAfEffectChain> getOrphanEffectChain_l(audio_session_t session);
-                // Called when the last effect handle on an effect instance is removed. If this
-                // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated
-                // and removed from mOrphanEffectChains if it does not contain any effect.
-                // Return true if the effect was found in mOrphanEffectChains, false otherwise.
-public:
-// TODO(b/288339104) suggest better grouping
-                bool updateOrphanEffectChains(const sp<IAfEffectModule>& effect);
-private:
+
                 std::vector< sp<IAfEffectModule> > purgeStaleEffects_l();
 
                 void broadcastParametersToRecordThreads_l(const String8& keyValuePairs);
-                void updateOutDevicesForRecordThreads_l(const DeviceDescriptorBaseVector& devices);
                 void forwardParametersToDownstreamPatches_l(
                         audio_io_handle_t upStream, const String8& keyValuePairs,
-                        const std::function<bool(const sp<PlaybackThread>&)>& useThread = nullptr);
-
-    // AudioStreamIn is immutable, so their fields are const.
-    // For emphasis, we could also make all pointers to them be "const *",
-    // but that would clutter the code unnecessarily.
-
-    struct AudioStreamIn : public Source {
-        AudioHwDevice* const audioHwDev;
-        sp<StreamInHalInterface> stream;
-        audio_input_flags_t flags;
-
-        sp<DeviceHalInterface> hwDev() const { return audioHwDev->hwDevice(); }
-
-        AudioStreamIn(AudioHwDevice *dev, const sp<StreamInHalInterface>& in,
-                audio_input_flags_t flags) :
-            audioHwDev(dev), stream(in), flags(flags) {}
-        status_t read(void *buffer, size_t bytes, size_t *read) override {
-            return stream->read(buffer, bytes, read);
-        }
-        status_t getCapturePosition(int64_t *frames, int64_t *time) override {
-            return stream->getCapturePosition(frames, time);
-        }
-        status_t standby() override { return stream->standby(); }
-    };
-
-    struct TeePatch {
-        sp<RecordThread::PatchRecord> patchRecord;
-        sp<PlaybackThread::PatchTrack> patchTrack;
-    };
+            const std::function<bool(const sp<IAfPlaybackThread>&)>& useThread = nullptr);
 
     // for mAudioSessionRefs only
     struct AudioSessionRef {
@@ -809,10 +731,9 @@
                 // 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:
+
+    mutable Mutex mClientLock;
+
                 // protected by mClientLock
                 DefaultKeyedVector< pid_t, wp<Client> >     mClients;   // see ~Client()
 
@@ -859,7 +780,7 @@
     mutable     hardware_call_state                 mHardwareStatus;    // for dump only
 
 
-                DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> >  mPlaybackThreads;
+    DefaultKeyedVector<audio_io_handle_t, sp<IAfPlaybackThread>> mPlaybackThreads;
                 stream_type_t                       mStreamTypes[AUDIO_STREAM_CNT];
 
                 // member variables below are protected by mLock
@@ -868,7 +789,7 @@
                 float                               mMasterBalance = 0.f;
                 // end of variables protected by mLock
 
-                DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> >    mRecordThreads;
+    DefaultKeyedVector<audio_io_handle_t, sp<IAfRecordThread>> mRecordThreads;
 
                 // protected by mClientLock
                 DefaultKeyedVector< pid_t, sp<NotificationClient> >    mNotificationClients;
@@ -882,9 +803,6 @@
                 // protected by mLock
                 Vector<AudioSessionRef*> mAudioSessionRefs;
 
-                float       masterVolume_l() const;
-                float       getMasterBalance_l() const;
-                bool        masterMute_l() const;
                 AudioHwDevice* loadHwModule_l(const char *name);
 
                 // sync events awaiting for a session to be created.
@@ -899,17 +817,14 @@
                 // list of MMAP stream control threads. Those threads allow for wake lock, routing
                 // and volume control for activity on the associated MMAP stream at the HAL.
                 // Audio data transfer is directly handled by the client creating the MMAP stream
-                DefaultKeyedVector< audio_io_handle_t, sp<MmapThread> >  mMmapThreads;
+    DefaultKeyedVector<audio_io_handle_t, sp<IAfMmapThread>> mMmapThreads;
 
-private:
     sp<Client>  registerPid(pid_t pid);    // always returns non-0
 
     // for use from destructor
     status_t    closeOutput_nonvirtual(audio_io_handle_t output);
-    void        closeThreadInternal_l(const sp<PlaybackThread>& thread);
     status_t    closeInput_nonvirtual(audio_io_handle_t input);
-    void        closeThreadInternal_l(const sp<RecordThread>& thread);
-    void        setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId);
+    void setAudioHwSyncForSession_l(IAfPlaybackThread* thread, audio_session_t sessionId);
 
     status_t    checkStreamType(audio_stream_type_t stream) const;
 
@@ -918,15 +833,10 @@
                                       size_t rejectedKVPSize, const String8& rejectedKVPs,
                                       uid_t callingUid);
 
-    sp<IAudioManager> getOrCreateAudioManager();
-
-public:
     // These methods read variables atomically without mLock,
     // though the variables are updated with mLock.
-    bool    isLowRamDevice() const { return mIsLowRamDevice; }
     size_t getClientSharedHeapSize() const;
 
-private:
     std::atomic<bool> mIsLowRamDevice;
     bool    mIsDeviceTypeKnown;
     int64_t mTotalMemory;
@@ -935,16 +845,13 @@
 
     nsecs_t mGlobalEffectEnableTime;  // when a global effect was last enabled
 
-    // protected by mLock
-    PatchPanel mPatchPanel;
-public:
-    // TODO(b/288339104) access by getter.
+    /* const */ sp<IAfPatchPanel> mPatchPanel;
+
     sp<EffectsFactoryHalInterface> mEffectsFactoryHal;
-private:
 
     const sp<PatchCommandThread> mPatchCommandThread;
-    sp<DeviceEffectManager> mDeviceEffectManager;
-    sp<MelReporter> mMelReporter;
+    /* const */ sp<DeviceEffectManager> mDeviceEffectManager;  // set onFirstRef
+    /* const */ sp<MelReporter> mMelReporter;  // set onFirstRef
 
     bool       mSystemReady;
     std::atomic_bool mAudioPolicyReady{};
@@ -959,8 +866,10 @@
 
     static inline constexpr const char *mMetricsId = AMEDIAMETRICS_KEY_AUDIO_FLINGER;
 
+public:
     // Keep in sync with java definition in media/java/android/media/AudioRecord.java
     static constexpr int32_t kMaxSharedAudioHistoryMs = 5000;
+private:
 
     std::map<media::audio::common::AudioMMapPolicyType,
              std::vector<media::audio::common::AudioMMapPolicyInfo>> mPolicyInfos;
@@ -974,8 +883,6 @@
     std::atomic_bool mBluetoothLatencyModesEnabled;
 };
 
-#undef INCLUDING_FROM_AUDIOFLINGER_H
-
 std::string formatToString(audio_format_t format);
 std::string inputFlagsToString(audio_input_flags_t flags);
 std::string outputFlagsToString(audio_output_flags_t flags);
diff --git a/services/audioflinger/Client.cpp b/services/audioflinger/Client.cpp
new file mode 100644
index 0000000..93599ac
--- /dev/null
+++ b/services/audioflinger/Client.cpp
@@ -0,0 +1,37 @@
+/*
+ * 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.
+ */
+
+#include "Client.h"
+
+namespace android {
+
+Client::Client(const sp<IAfClientCallback>& afClientCallback, pid_t pid)
+    : mAfClientCallback(afClientCallback)
+    , mPid(pid)
+    , mClientAllocator(AllocatorFactory::getClientAllocator()) {}
+
+// Client destructor must be called with AudioFlinger::mClientLock held
+Client::~Client()
+{
+    mAfClientCallback->removeClient_l(mPid);
+}
+
+AllocatorFactory::ClientAllocator& Client::allocator()
+{
+    return mClientAllocator;
+}
+
+}   // namespace android
\ No newline at end of file
diff --git a/services/audioflinger/Client.h b/services/audioflinger/Client.h
index cb507fe..b2e3cf7 100644
--- a/services/audioflinger/Client.h
+++ b/services/audioflinger/Client.h
@@ -16,26 +16,42 @@
 
 #pragma once
 
-// TODO(b/288339104) Move to nested namespace
+#include <afutils/AllocatorFactory.h>
+#include <android-base/macros.h>  // DISALLOW_COPY_AND_ASSIGN
+#include <utils/Mutex.h>
+#include <utils/RefBase.h>        // avoid transitive dependency
+
+// TODO(b/291318727) Move to nested namespace
 namespace android {
 
-class AudioFlinger;
+class IAfPlaybackThread;
+
+class IAfClientCallback : public virtual RefBase {
+public:
+    virtual Mutex& clientMutex() const = 0;
+    virtual void removeClient_l(pid_t pid) = 0;
+    virtual void removeNotificationClient(pid_t pid) = 0;
+    virtual status_t moveAuxEffectToIo(
+            int effectId,
+            const sp<IAfPlaybackThread>& dstThread,
+            sp<IAfPlaybackThread>* srcThread) = 0;  // used by indirectly by clients.
+};
 
 class Client : public RefBase {
 public:
-    Client(const sp<AudioFlinger>& audioFlinger, pid_t pid);
+    Client(const sp<IAfClientCallback>& audioFlinger, pid_t pid);
 
     // TODO(b/289139675) make Client container.
     // Client destructor must be called with AudioFlinger::mClientLock held
     ~Client() override;
     AllocatorFactory::ClientAllocator& allocator();
     pid_t pid() const { return mPid; }
-    sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; }
+    const auto& afClientCallback() const { return mAfClientCallback; }
 
 private:
     DISALLOW_COPY_AND_ASSIGN(Client);
 
-    const sp<AudioFlinger> mAudioFlinger;
+    const sp<IAfClientCallback> mAfClientCallback;
     const pid_t mPid;
     AllocatorFactory::ClientAllocator mClientAllocator;
 };
diff --git a/services/audioflinger/DeviceEffectManager.cpp b/services/audioflinger/DeviceEffectManager.cpp
index 366a7ab..ebb2428 100644
--- a/services/audioflinger/DeviceEffectManager.cpp
+++ b/services/audioflinger/DeviceEffectManager.cpp
@@ -15,16 +15,17 @@
 ** limitations under the License.
 */
 
-
-#define LOG_TAG "AudioFlinger::DeviceEffectManager"
+#define LOG_TAG "DeviceEffectManager"
 //#define LOG_NDEBUG 0
 
-#include <utils/Log.h>
-#include <audio_utils/primitives.h>
+#include "DeviceEffectManager.h"
 
-#include "AudioFlinger.h"
 #include "EffectConfiguration.h"
+
+#include <afutils/DumpTryLock.h>
+#include <audio_utils/primitives.h>
 #include <media/audiohal/EffectsFactoryHalInterface.h>
+#include <utils/Log.h>
 
 // ----------------------------------------------------------------------------
 
@@ -34,20 +35,39 @@
 using detail::AudioHalVersionInfo;
 using media::IEffectClient;
 
-void AudioFlinger::DeviceEffectManager::onCreateAudioPatch(audio_patch_handle_t handle,
-        const PatchPanel::Patch& patch) {
+DeviceEffectManager::DeviceEffectManager(
+        const sp<IAfDeviceEffectManagerCallback>& afDeviceEffectManagerCallback)
+    : mAfDeviceEffectManagerCallback(afDeviceEffectManagerCallback),
+      mMyCallback(new DeviceEffectManagerCallback(*this)) {}
+
+void DeviceEffectManager::onFirstRef() {
+    mAfDeviceEffectManagerCallback->getPatchCommandThread()->addListener(this);
+}
+
+status_t DeviceEffectManager::addEffectToHal(const struct audio_port_config* device,
+        const sp<EffectHalInterface>& effect) {
+    return mAfDeviceEffectManagerCallback->addEffectToHal(device, effect);
+};
+
+status_t DeviceEffectManager::removeEffectFromHal(const struct audio_port_config* device,
+        const sp<EffectHalInterface>& effect) {
+    return mAfDeviceEffectManagerCallback->removeEffectFromHal(device, effect);
+};
+
+void DeviceEffectManager::onCreateAudioPatch(audio_patch_handle_t handle,
+        const IAfPatchPanel::Patch& patch) {
     ALOGV("%s handle %d mHalHandle %d device sink %08x",
             __func__, handle, patch.mHalHandle,
             patch.mAudioPatch.num_sinks > 0 ? patch.mAudioPatch.sinks[0].ext.device.type : 0);
     Mutex::Autolock _l(mLock);
     for (auto& effect : mDeviceEffects) {
-        status_t status = effect.second->onCreatePatch(handle, &patch); // TODO(b/288339104) void*
+        status_t status = effect.second->onCreatePatch(handle, patch);
         ALOGV("%s Effect onCreatePatch status %d", __func__, status);
         ALOGW_IF(status == BAD_VALUE, "%s onCreatePatch error %d", __func__, status);
     }
 }
 
-void AudioFlinger::DeviceEffectManager::onReleaseAudioPatch(audio_patch_handle_t handle) {
+void DeviceEffectManager::onReleaseAudioPatch(audio_patch_handle_t handle) {
     ALOGV("%s", __func__);
     Mutex::Autolock _l(mLock);
     for (auto& effect : mDeviceEffects) {
@@ -56,12 +76,12 @@
 }
 
 // DeviceEffectManager::createEffect_l() must be called with AudioFlinger::mLock held
-sp<IAfEffectHandle> AudioFlinger::DeviceEffectManager::createEffect_l(
+sp<IAfEffectHandle> DeviceEffectManager::createEffect_l(
         effect_descriptor_t *descriptor,
         const AudioDeviceTypeAddr& device,
         const sp<Client>& client,
         const sp<IEffectClient>& effectClient,
-        const std::map<audio_patch_handle_t, PatchPanel::Patch>& patches,
+        const std::map<audio_patch_handle_t, IAfPatchPanel::Patch>& patches,
         int *enabled,
         status_t *status,
         bool probe,
@@ -83,7 +103,8 @@
             effect = iter->second;
         } else {
             effect = IAfDeviceEffectProxy::create(device, mMyCallback,
-                    descriptor, mAudioFlinger.nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT),
+                    descriptor,
+                    mAfDeviceEffectManagerCallback->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT),
                     notifyFramesProcessed);
         }
         // create effect handle and connect it to effect module
@@ -93,7 +114,7 @@
         if (lStatus == NO_ERROR) {
             lStatus = effect->addHandle(handle.get());
             if (lStatus == NO_ERROR) {
-                lStatus = effect->init(&patches); // TODO(b/288339104) void*
+                lStatus = effect->init(patches);
                 if (lStatus == NAME_NOT_FOUND) {
                     lStatus = NO_ERROR;
                 }
@@ -110,7 +131,7 @@
     return handle;
 }
 
-status_t AudioFlinger::DeviceEffectManager::checkEffectCompatibility(
+status_t DeviceEffectManager::checkEffectCompatibility(
         const effect_descriptor_t *desc) {
     const sp<EffectsFactoryHalInterface> effectsFactory =
             audioflinger::EffectConfiguration::getEffectsFactoryHal();
@@ -136,7 +157,7 @@
     return NO_ERROR;
 }
 
-status_t AudioFlinger::DeviceEffectManager::createEffectHal(
+status_t DeviceEffectManager::createEffectHal(
         const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t deviceId,
         sp<EffectHalInterface> *effect) {
     status_t status = NO_INIT;
@@ -149,10 +170,10 @@
     return status;
 }
 
-void AudioFlinger::DeviceEffectManager::dump(int fd)
+void DeviceEffectManager::dump(int fd)
 NO_THREAD_SAFETY_ANALYSIS  // conditional try lock
 {
-    const bool locked = dumpTryLock(mLock);
+    const bool locked = afutils::dumpTryLock(mLock);
     if (!locked) {
         String8 result("DeviceEffectManager may be deadlocked\n");
         write(fd, result.string(), result.size());
@@ -173,15 +194,14 @@
     }
 }
 
-
-size_t AudioFlinger::DeviceEffectManager::removeEffect(const sp<IAfDeviceEffectProxy>& effect)
+size_t DeviceEffectManager::removeEffect(const sp<IAfDeviceEffectProxy>& effect)
 {
     Mutex::Autolock _l(mLock);
     mDeviceEffects.erase(effect->device());
     return mDeviceEffects.size();
 }
 
-bool AudioFlinger::DeviceEffectManagerCallback::disconnectEffectHandle(
+bool DeviceEffectManagerCallback::disconnectEffectHandle(
         IAfEffectHandle *handle, bool unpinIfLast) {
     sp<IAfEffectBase> effectBase = handle->effect().promote();
     if (effectBase == nullptr) {
@@ -203,4 +223,12 @@
     return true;
 }
 
+bool DeviceEffectManagerCallback::isAudioPolicyReady() const {
+    return mManager.afDeviceEffectManagerCallback()->isAudioPolicyReady();
+}
+
+int DeviceEffectManagerCallback::newEffectId() const {
+    return mManager.afDeviceEffectManagerCallback()->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT);
+}
+
 } // namespace android
diff --git a/services/audioflinger/DeviceEffectManager.h b/services/audioflinger/DeviceEffectManager.h
index c589714..6111030 100644
--- a/services/audioflinger/DeviceEffectManager.h
+++ b/services/audioflinger/DeviceEffectManager.h
@@ -15,26 +15,41 @@
 ** limitations under the License.
 */
 
-#ifndef INCLUDING_FROM_AUDIOFLINGER_H
-    #error This header file should only be included from AudioFlinger.h
-#endif
+#pragma once
+
+#include "IAfEffect.h"
+#include "PatchCommandThread.h"
+
+#include <utils/Mutex.h>  // avoid transitive dependency
+
+namespace android {
+
+class IAfDeviceEffectManagerCallback : public virtual RefBase {
+public:
+    virtual bool isAudioPolicyReady() const = 0;
+    virtual audio_unique_id_t nextUniqueId(audio_unique_id_use_t use) = 0;
+    virtual const sp<PatchCommandThread>& getPatchCommandThread() = 0;
+    virtual status_t addEffectToHal(
+            const struct audio_port_config* device, const sp<EffectHalInterface>& effect) = 0;
+    virtual status_t removeEffectFromHal(
+            const struct audio_port_config* device, const sp<EffectHalInterface>& effect) = 0;
+};
+
+class DeviceEffectManagerCallback;
 
 // DeviceEffectManager is concealed within AudioFlinger, their lifetimes are the same.
 class DeviceEffectManager : public PatchCommandThread::PatchCommandListener {
 public:
-    explicit DeviceEffectManager(AudioFlinger& audioFlinger)
-        : mAudioFlinger(audioFlinger),
-          mMyCallback(new DeviceEffectManagerCallback(*this)) {}
+    explicit DeviceEffectManager(
+            const sp<IAfDeviceEffectManagerCallback>& afDeviceEffectManagerCallback);
 
-    void onFirstRef() override {
-        mAudioFlinger.mPatchCommandThread->addListener(this);
-    }
+    void onFirstRef() override;
 
     sp<IAfEffectHandle> createEffect_l(effect_descriptor_t *descriptor,
                 const AudioDeviceTypeAddr& device,
                 const sp<Client>& client,
                 const sp<media::IEffectClient>& effectClient,
-                const std::map<audio_patch_handle_t, PatchPanel::Patch>& patches,
+                const std::map<audio_patch_handle_t, IAfPatchPanel::Patch>& patches,
                 int *enabled,
                 status_t *status,
                 bool probe,
@@ -45,34 +60,29 @@
            int32_t sessionId, int32_t deviceId,
            sp<EffectHalInterface> *effect);
     status_t addEffectToHal(const struct audio_port_config *device,
-            const sp<EffectHalInterface>& effect) {
-        return mAudioFlinger.addEffectToHal(device, effect);
-    };
+            const sp<EffectHalInterface>& effect);
     status_t removeEffectFromHal(const struct audio_port_config *device,
-            const sp<EffectHalInterface>& effect) {
-        return mAudioFlinger.removeEffectFromHal(device, effect);
-    };
+            const sp<EffectHalInterface>& effect);
 
-    AudioFlinger& audioFlinger() const { return mAudioFlinger; }
+    const auto& afDeviceEffectManagerCallback() const { return mAfDeviceEffectManagerCallback; }
 
     void dump(int fd);
 
     // PatchCommandThread::PatchCommandListener implementation
 
     void onCreateAudioPatch(audio_patch_handle_t handle,
-                            const PatchPanel::Patch& patch) override;
-    void onReleaseAudioPatch(audio_patch_handle_t handle) override;
+            const IAfPatchPanel::Patch& patch) final;
+    void onReleaseAudioPatch(audio_patch_handle_t handle) final;
 
 private:
     status_t checkEffectCompatibility(const effect_descriptor_t *desc);
 
     Mutex mLock;
-    AudioFlinger &mAudioFlinger;
+    const sp<IAfDeviceEffectManagerCallback> mAfDeviceEffectManagerCallback;
     const sp<DeviceEffectManagerCallback> mMyCallback;
     std::map<AudioDeviceTypeAddr, sp<IAfDeviceEffectProxy>> mDeviceEffects;
 };
 
-public: // TODO(b/288339104) extract inner class.
 class DeviceEffectManagerCallback : public EffectCallbackInterface {
 public:
     explicit DeviceEffectManagerCallback(DeviceEffectManager& manager)
@@ -129,11 +139,9 @@
 
     wp<IAfEffectChain> chain() const override { return nullptr; }
 
-    bool isAudioPolicyReady() const override {
-        return mManager.audioFlinger().isAudioPolicyReady();
-    }
+    bool isAudioPolicyReady() const final;
 
-    int newEffectId() { return mManager.audioFlinger().nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT); }
+    int newEffectId() const;
 
     status_t addEffectToHal(const struct audio_port_config *device,
             const sp<EffectHalInterface>& effect) {
@@ -146,4 +154,5 @@
 private:
     DeviceEffectManager& mManager;
 };
-private:
+
+}  // namespace android
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index 1f26cb0..7590afd 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -19,10 +19,25 @@
 #define LOG_TAG "AudioFlinger"
 //#define LOG_NDEBUG 0
 
-#include <algorithm>
+#include "Effects.h"
 
-#include "Configuration.h"
-#include <utils/Log.h>
+#include "Client.h"
+#include "EffectConfiguration.h"
+
+#include <afutils/DumpTryLock.h>
+#include <audio_utils/channels.h>
+#include <audio_utils/primitives.h>
+#include <media/AudioCommonTypes.h>
+#include <media/AudioContainers.h>
+#include <media/AudioDeviceTypeAddr.h>
+#include <media/AudioEffect.h>
+#include <media/ShmemCompat.h>
+#include <media/TypeConverter.h>
+#include <media/audiohal/EffectHalInterface.h>
+#include <media/audiohal/EffectsFactoryHalInterface.h>
+#include <mediautils/MethodStatistics.h>
+#include <mediautils/ServiceUtilities.h>
+#include <mediautils/TimeCheck.h>
 #include <system/audio_effects/effect_aec.h>
 #include <system/audio_effects/effect_downmix.h>
 #include <system/audio_effects/effect_dynamicsprocessing.h>
@@ -30,22 +45,9 @@
 #include <system/audio_effects/effect_ns.h>
 #include <system/audio_effects/effect_spatializer.h>
 #include <system/audio_effects/effect_visualizer.h>
-#include <audio_utils/channels.h>
-#include <audio_utils/primitives.h>
-#include <media/AudioCommonTypes.h>
-#include <media/AudioContainers.h>
-#include <media/AudioEffect.h>
-#include <media/AudioDeviceTypeAddr.h>
-#include <media/ShmemCompat.h>
-#include <media/audiohal/EffectHalInterface.h>
-#include <media/audiohal/EffectsFactoryHalInterface.h>
-#include <mediautils/MethodStatistics.h>
-#include <mediautils/ServiceUtilities.h>
-#include <mediautils/TimeCheck.h>
+#include <utils/Log.h>
 
-#include "AudioFlinger.h"
-#include "EffectConfiguration.h"
-#include "Effects.h"
+#include <algorithm>
 
 // ----------------------------------------------------------------------------
 
@@ -507,7 +509,7 @@
 
     result.appendFormat("\tEffect ID %d:\n", mId);
 
-    bool locked = AudioFlinger::dumpTryLock(mLock);
+    const bool locked = afutils::dumpTryLock(mLock);
     // failed to lock - AudioFlinger is probably deadlocked
     if (!locked) {
         result.append("\t\tCould not lock Fx mutex:\n");
@@ -1621,7 +1623,7 @@
     EffectBase::dump(fd, args);
 
     String8 result;
-    bool locked = AudioFlinger::dumpTryLock(mLock);
+    const bool locked = afutils::dumpTryLock(mLock);
 
     result.append("\t\tStatus Engine:\n");
     result.appendFormat("\t\t%03d    %p\n",
@@ -1637,7 +1639,7 @@
             mConfig.inputCfg.samplingRate,
             mConfig.inputCfg.channels,
             mConfig.inputCfg.format,
-            formatToString((audio_format_t)mConfig.inputCfg.format).c_str());
+            toString(static_cast<audio_format_t>(mConfig.inputCfg.format)).c_str());
 
     result.append("\t\t- Output configuration:\n");
     result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
@@ -1647,7 +1649,7 @@
             mConfig.outputCfg.samplingRate,
             mConfig.outputCfg.channels,
             mConfig.outputCfg.format,
-            formatToString((audio_format_t)mConfig.outputCfg.format).c_str());
+            toString(static_cast<audio_format_t>(mConfig.outputCfg.format)).c_str());
 
     result.appendFormat("\t\t- HAL buffers:\n"
             "\t\t\tIn(%s) InConversion(%s) Out(%s) OutConversion(%s)\n",
@@ -1886,7 +1888,7 @@
         }
         mCblkMemory.clear();    // free the shared memory before releasing the heap it belongs to
         // Client destructor must run with AudioFlinger client mutex locked
-        Mutex::Autolock _l2(mClient->audioFlinger()->mClientLock);
+        Mutex::Autolock _l2(mClient->afClientCallback()->clientMutex());
         mClient.clear();
     }
 }
@@ -2095,7 +2097,7 @@
 void EffectHandle::dumpToBuffer(char* buffer, size_t size) const
 NO_THREAD_SAFETY_ANALYSIS  // conditional try lock
 {
-    bool locked = mCblk != NULL && AudioFlinger::dumpTryLock(mCblk->lock);
+    const bool locked = mCblk != nullptr && afutils::dumpTryLock(mCblk->lock);
 
     snprintf(buffer, size, "\t\t\t%5d    %5d  %3s    %3s  %5u  %5u\n",
             (mClient == 0) ? getpid() : mClient->pid(),
@@ -2116,34 +2118,25 @@
 
 /* static */
 sp<IAfEffectChain> IAfEffectChain::create(
-        const wp<Thread /*ThreadBase*/>& wThread,  // TODO(b/288339104) update type
+        const sp<IAfThreadBase>& thread,
         audio_session_t sessionId)
 {
-    // TODO(b/288339104) no weak pointer cast.
-    return sp<EffectChain>::make(sp<AudioFlinger::ThreadBase>::cast(wThread.promote()), sessionId);
+    return sp<EffectChain>::make(thread, sessionId);
 }
 
-EffectChain::EffectChain(const wp<AudioFlinger::ThreadBase>& thread,
+EffectChain::EffectChain(const sp<IAfThreadBase>& thread,
                                        audio_session_t sessionId)
     : mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
       mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
       mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX),
       mEffectCallback(new EffectCallback(wp<EffectChain>(this), thread))
 {
-    sp<AudioFlinger::ThreadBase> p = thread.promote();
-    if (p == nullptr) {
-        return;
-    }
-    mStrategy = p->getStrategyForStream(AUDIO_STREAM_MUSIC);
-    mMaxTailBuffers = ((kProcessTailDurationMs * p->sampleRate()) / 1000) /
-                                    p->frameCount();
+    mStrategy = thread->getStrategyForStream(AUDIO_STREAM_MUSIC);
+    mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) /
+                                    thread->frameCount();
 }
 
-EffectChain::~EffectChain()
-{
-}
-
-// getEffectFromDesc_l() must be called with AudioFlinger::ThreadBase::mLock held
+// getEffectFromDesc_l() must be called with IAfThreadBase::mutex() held
 sp<IAfEffectModule> EffectChain::getEffectFromDesc_l(
         effect_descriptor_t *descriptor) const
 {
@@ -2157,7 +2150,7 @@
     return 0;
 }
 
-// getEffectFromId_l() must be called with AudioFlinger::ThreadBase::mLock held
+// getEffectFromId_l() must be called with IAfThreadBase::mutex() held
 sp<IAfEffectModule> EffectChain::getEffectFromId_l(int id) const
 {
     size_t size = mEffects.size();
@@ -2171,7 +2164,7 @@
     return 0;
 }
 
-// getEffectFromType_l() must be called with AudioFlinger::ThreadBase::mLock held
+// getEffectFromType_l() must be called with IAfThreadBase::mutex() held
 sp<IAfEffectModule> EffectChain::getEffectFromType_l(
         const effect_uuid_t *type) const
 {
@@ -2266,7 +2259,7 @@
     }
 }
 
-// createEffect_l() must be called with AudioFlinger::ThreadBase::mLock held
+// createEffect_l() must be called with IAfThreadBase::mutex() held
 status_t EffectChain::createEffect_l(sp<IAfEffectModule>& effect,
                                                    effect_descriptor_t *desc,
                                                    int id,
@@ -2285,13 +2278,13 @@
     return lStatus;
 }
 
-// addEffect_l() must be called with AudioFlinger::ThreadBase::mLock held
+// addEffect_l() must be called with IAfThreadBase::mutex() held
 status_t EffectChain::addEffect_l(const sp<IAfEffectModule>& effect)
 {
     Mutex::Autolock _l(mLock);
     return addEffect_ll(effect);
 }
-// addEffect_l() must be called with AudioFlinger::ThreadBase::mLock and EffectChain::mLock held
+// addEffect_l() must be called with IAfThreadBase::mLock and EffectChain::mutex() held
 status_t EffectChain::addEffect_ll(const sp<IAfEffectModule>& effect)
 {
     effect->setCallback(mEffectCallback);
@@ -2445,7 +2438,7 @@
     return idx_insert;
 }
 
-// removeEffect_l() must be called with AudioFlinger::ThreadBase::mLock held
+// removeEffect_l() must be called with IAfThreadBase::mutex() held
 size_t EffectChain::removeEffect_l(const sp<IAfEffectModule>& effect,
                                                  bool release)
 {
@@ -2493,7 +2486,7 @@
     return mEffects.size();
 }
 
-// setDevices_l() must be called with AudioFlinger::ThreadBase::mLock held
+// setDevices_l() must be called with IAfThreadBase::mutex() held
 void EffectChain::setDevices_l(const AudioDeviceTypeAddrVector &devices)
 {
     size_t size = mEffects.size();
@@ -2502,7 +2495,7 @@
     }
 }
 
-// setInputDevice_l() must be called with AudioFlinger::ThreadBase::mLock held
+// setInputDevice_l() must be called with IAfThreadBase::mutex() held
 void EffectChain::setInputDevice_l(const AudioDeviceTypeAddr &device)
 {
     size_t size = mEffects.size();
@@ -2511,7 +2504,7 @@
     }
 }
 
-// setMode_l() must be called with AudioFlinger::ThreadBase::mLock held
+// setMode_l() must be called with IAfThreadBase::mutex() held
 void EffectChain::setMode_l(audio_mode_t mode)
 {
     size_t size = mEffects.size();
@@ -2520,7 +2513,7 @@
     }
 }
 
-// setAudioSource_l() must be called with AudioFlinger::ThreadBase::mLock held
+// setAudioSource_l() must be called with IAfThreadBase::mutex() held
 void EffectChain::setAudioSource_l(audio_source_t source)
 {
     size_t size = mEffects.size();
@@ -2536,7 +2529,7 @@
     return false;
 }
 
-// setVolume_l() must be called with AudioFlinger::ThreadBase::mLock or EffectChain::mLock held
+// setVolume_l() must be called with IAfThreadBase::mLock or EffectChain::mLock held
 bool EffectChain::setVolume_l(uint32_t *left, uint32_t *right, bool force)
 {
     uint32_t newLeft = *left;
@@ -2603,7 +2596,7 @@
     return hasControl;
 }
 
-// resetVolume_l() must be called with AudioFlinger::ThreadBase::mLock or EffectChain::mLock held
+// resetVolume_l() must be called with IAfThreadBase::mutex() or EffectChain::mLock held
 void EffectChain::resetVolume_l()
 {
     if ((mLeftVolume != UINT_MAX) && (mRightVolume != UINT_MAX)) {
@@ -2614,7 +2607,7 @@
 }
 
 // containsHapticGeneratingEffect_l must be called with
-// AudioFlinger::ThreadBase::mLock or EffectChain::mLock held
+// IAfThreadBase::mutex() or EffectChain::mLock held
 bool EffectChain::containsHapticGeneratingEffect_l()
 {
     for (size_t i = 0; i < mEffects.size(); ++i) {
@@ -2653,7 +2646,7 @@
     result.appendFormat("    %zu effects for session %d\n", numEffects, mSessionId);
 
     if (numEffects) {
-        bool locked = AudioFlinger::dumpTryLock(mLock);
+        const bool locked = afutils::dumpTryLock(mLock);
         // failed to lock - AudioFlinger is probably deadlocked
         if (!locked) {
             result.append("\tCould not lock mutex:\n");
@@ -2683,7 +2676,7 @@
     }
 }
 
-// must be called with AudioFlinger::ThreadBase::mLock held
+// must be called with IAfThreadBase::mutex() held
 void EffectChain::setEffectSuspended_l(
         const effect_uuid_t *type, bool suspend)
 {
@@ -2739,7 +2732,7 @@
     }
 }
 
-// must be called with AudioFlinger::ThreadBase::mLock held
+// must be called with IAfThreadBase::mutex() held
 void EffectChain::setEffectSuspendedAll_l(bool suspend)
 {
     sp<SuspendedEffectDesc> desc;
@@ -2895,7 +2888,7 @@
     return false;
 }
 
-void EffectChain::setThread(const sp<AudioFlinger::ThreadBase>& thread)
+void EffectChain::setThread(const sp<IAfThreadBase>& thread)
 {
     Mutex::Autolock _l(mLock);
     mEffectCallback->setThread(thread);
@@ -2962,7 +2955,7 @@
 }
 
 // isCompatibleWithThread_l() must be called with thread->mLock held
-bool EffectChain::isCompatibleWithThread_l(const sp<AudioFlinger::ThreadBase>& thread) const
+bool EffectChain::isCompatibleWithThread_l(const sp<IAfThreadBase>& thread) const
 {
     Mutex::Autolock _l(mLock);
     for (size_t i = 0; i < mEffects.size(); i++) {
@@ -2989,18 +2982,18 @@
 bool EffectChain::EffectCallback::updateOrphanEffectChains(
         const sp<IAfEffectBase>& effect) {
     // in EffectChain context, an EffectBase is always from an EffectModule so static cast is safe
-    return mAudioFlinger.updateOrphanEffectChains(effect->asEffectModule());
+    return mAfThreadCallback->updateOrphanEffectChains(effect->asEffectModule());
 }
 
 status_t EffectChain::EffectCallback::allocateHalBuffer(
         size_t size, sp<EffectBufferHalInterface>* buffer) {
-    return mAudioFlinger.mEffectsFactoryHal->allocateBuffer(size, buffer);
+    return mAfThreadCallback->getEffectsFactoryHal()->allocateBuffer(size, buffer);
 }
 
 status_t EffectChain::EffectCallback::addEffectToHal(
         const sp<EffectHalInterface>& effect) {
     status_t result = NO_INIT;
-    sp<AudioFlinger::ThreadBase> t = thread().promote();
+    const sp<IAfThreadBase> t = thread().promote();
     if (t == nullptr) {
         return result;
     }
@@ -3016,7 +3009,7 @@
 status_t EffectChain::EffectCallback::removeEffectFromHal(
         const sp<EffectHalInterface>& effect) {
     status_t result = NO_INIT;
-    sp<AudioFlinger::ThreadBase> t = thread().promote();
+    const sp<IAfThreadBase> t = thread().promote();
     if (t == nullptr) {
         return result;
     }
@@ -3030,7 +3023,7 @@
 }
 
 audio_io_handle_t EffectChain::EffectCallback::io() const {
-    sp<AudioFlinger::ThreadBase> t = thread().promote();
+    const sp<IAfThreadBase> t = thread().promote();
     if (t == nullptr) {
         return AUDIO_IO_HANDLE_NONE;
     }
@@ -3038,7 +3031,7 @@
 }
 
 bool EffectChain::EffectCallback::isOutput() const {
-    sp<AudioFlinger::ThreadBase> t = thread().promote();
+    const sp<IAfThreadBase> t = thread().promote();
     if (t == nullptr) {
         return true;
     }
@@ -3046,19 +3039,19 @@
 }
 
 bool EffectChain::EffectCallback::isOffload() const {
-    return mThreadType == AudioFlinger::ThreadBase::OFFLOAD;
+    return mThreadType == IAfThreadBase::OFFLOAD;
 }
 
 bool EffectChain::EffectCallback::isOffloadOrDirect() const {
-    return mThreadType == AudioFlinger::ThreadBase::OFFLOAD
-            || mThreadType == AudioFlinger::ThreadBase::DIRECT;
+    return mThreadType == IAfThreadBase::OFFLOAD
+            || mThreadType == IAfThreadBase::DIRECT;
 }
 
 bool EffectChain::EffectCallback::isOffloadOrMmap() const {
     switch (mThreadType) {
-    case AudioFlinger::ThreadBase::OFFLOAD:
-    case AudioFlinger::ThreadBase::MMAP_PLAYBACK:
-    case AudioFlinger::ThreadBase::MMAP_CAPTURE:
+    case IAfThreadBase::OFFLOAD:
+    case IAfThreadBase::MMAP_PLAYBACK:
+    case IAfThreadBase::MMAP_CAPTURE:
         return true;
     default:
         return false;
@@ -3066,11 +3059,11 @@
 }
 
 bool EffectChain::EffectCallback::isSpatializer() const {
-    return mThreadType == AudioFlinger::ThreadBase::SPATIALIZER;
+    return mThreadType == IAfThreadBase::SPATIALIZER;
 }
 
 uint32_t EffectChain::EffectCallback::sampleRate() const {
-    sp<AudioFlinger::ThreadBase> t = thread().promote();
+    const sp<IAfThreadBase> t = thread().promote();
     if (t == nullptr) {
         return 0;
     }
@@ -3078,7 +3071,7 @@
 }
 
 audio_channel_mask_t EffectChain::EffectCallback::inChannelMask(int id) const {
-    sp<AudioFlinger::ThreadBase> t = thread().promote();
+    const sp<IAfThreadBase> t = thread().promote();
     if (t == nullptr) {
         return AUDIO_CHANNEL_NONE;
     }
@@ -3087,7 +3080,7 @@
         return AUDIO_CHANNEL_NONE;
     }
 
-    if (mThreadType == AudioFlinger::ThreadBase::SPATIALIZER) {
+    if (mThreadType == IAfThreadBase::SPATIALIZER) {
         if (c->sessionId() == AUDIO_SESSION_OUTPUT_STAGE) {
             if (c->isFirstEffect(id)) {
                 return t->mixerChannelMask();
@@ -3096,7 +3089,7 @@
             }
         } else if (!audio_is_global_session(c->sessionId())) {
             if ((t->hasAudioSession_l(c->sessionId())
-                    & AudioFlinger::ThreadBase::SPATIALIZED_SESSION) != 0) {
+                    & IAfThreadBase::SPATIALIZED_SESSION) != 0) {
                 return t->mixerChannelMask();
             } else {
                 return t->channelMask();
@@ -3114,7 +3107,7 @@
 }
 
 audio_channel_mask_t EffectChain::EffectCallback::outChannelMask() const {
-    sp<AudioFlinger::ThreadBase> t = thread().promote();
+    const sp<IAfThreadBase> t = thread().promote();
     if (t == nullptr) {
         return AUDIO_CHANNEL_NONE;
     }
@@ -3123,10 +3116,10 @@
         return AUDIO_CHANNEL_NONE;
     }
 
-    if (mThreadType == AudioFlinger::ThreadBase::SPATIALIZER) {
+    if (mThreadType == IAfThreadBase::SPATIALIZER) {
         if (!audio_is_global_session(c->sessionId())) {
             if ((t->hasAudioSession_l(c->sessionId())
-                    & AudioFlinger::ThreadBase::SPATIALIZED_SESSION) != 0) {
+                    & IAfThreadBase::SPATIALIZED_SESSION) != 0) {
                 return t->mixerChannelMask();
             } else {
                 return t->channelMask();
@@ -3144,7 +3137,7 @@
 }
 
 audio_channel_mask_t EffectChain::EffectCallback::hapticChannelMask() const {
-    sp<AudioFlinger::ThreadBase> t = thread().promote();
+    const sp<IAfThreadBase> t = thread().promote();
     if (t == nullptr) {
         return AUDIO_CHANNEL_NONE;
     }
@@ -3152,7 +3145,7 @@
 }
 
 size_t EffectChain::EffectCallback::frameCount() const {
-    sp<AudioFlinger::ThreadBase> t = thread().promote();
+    const sp<IAfThreadBase> t = thread().promote();
     if (t == nullptr) {
         return 0;
     }
@@ -3162,7 +3155,7 @@
 uint32_t EffectChain::EffectCallback::latency() const
 NO_THREAD_SAFETY_ANALYSIS  // latency_l() access
 {
-    sp<AudioFlinger::ThreadBase> t = thread().promote();
+    const sp<IAfThreadBase> t = thread().promote();
     if (t == nullptr) {
         return 0;
     }
@@ -3173,7 +3166,7 @@
 void EffectChain::EffectCallback::setVolumeForOutput(float left, float right) const
 NO_THREAD_SAFETY_ANALYSIS  // setVolumeForOutput_l() access
 {
-    sp<AudioFlinger::ThreadBase> t = thread().promote();
+    const sp<IAfThreadBase> t = thread().promote();
     if (t == nullptr) {
         return;
     }
@@ -3182,7 +3175,7 @@
 
 void EffectChain::EffectCallback::checkSuspendOnEffectEnabled(
         const sp<IAfEffectBase>& effect, bool enabled, bool threadLocked) {
-    sp<AudioFlinger::ThreadBase> t = thread().promote();
+    const sp<IAfThreadBase> t = thread().promote();
     if (t == nullptr) {
         return;
     }
@@ -3197,7 +3190,7 @@
 }
 
 void EffectChain::EffectCallback::onEffectEnable(const sp<IAfEffectBase>& effect) {
-    sp<AudioFlinger::ThreadBase> t = thread().promote();
+    const sp<IAfThreadBase> t = thread().promote();
     if (t == nullptr) {
         return;
     }
@@ -3208,7 +3201,7 @@
 void EffectChain::EffectCallback::onEffectDisable(const sp<IAfEffectBase>& effect) {
     checkSuspendOnEffectEnabled(effect, false, false /*threadLocked*/);
 
-    sp<AudioFlinger::ThreadBase> t = thread().promote();
+    const sp<IAfThreadBase> t = thread().promote();
     if (t == nullptr) {
         return;
     }
@@ -3217,7 +3210,7 @@
 
 bool EffectChain::EffectCallback::disconnectEffectHandle(IAfEffectHandle *handle,
                                                       bool unpinIfLast) {
-    sp<AudioFlinger::ThreadBase> t = thread().promote();
+    const sp<IAfThreadBase> t = thread().promote();
     if (t == nullptr) {
         return false;
     }
@@ -3257,11 +3250,11 @@
 /* static */
 sp<IAfDeviceEffectProxy> IAfDeviceEffectProxy::create(
         const AudioDeviceTypeAddr& device,
-        const sp</* DeviceEffectManagerCallback */ RefBase>& callback,  // TODO(b/288339104) type
+        const sp<DeviceEffectManagerCallback>& callback,
         effect_descriptor_t *desc, int id, bool notifyFramesProcessed)
 {
     return sp<DeviceEffectProxy>::make(device,
-            sp<AudioFlinger::DeviceEffectManagerCallback>::cast(callback),
+            callback,
             desc, id, notifyFramesProcessed);
 }
 
@@ -3287,7 +3280,7 @@
 }
 
 status_t DeviceEffectProxy::init(
-        const std::map <audio_patch_handle_t, AudioFlinger::PatchPanel::Patch>& patches) {
+        const std::map <audio_patch_handle_t, IAfPatchPanel::Patch>& patches) {
 //For all audio patches
 //If src or sink device match
 //If the effect is HW accelerated
@@ -3310,7 +3303,7 @@
 }
 
 status_t DeviceEffectProxy::onCreatePatch(
-        audio_patch_handle_t patchHandle, const AudioFlinger::PatchPanel::Patch& patch) {
+        audio_patch_handle_t patchHandle, const IAfPatchPanel::Patch& patch) {
     status_t status = NAME_NOT_FOUND;
     sp<IAfEffectHandle> handle;
     // only consider source[0] as this is the only "true" source of a patch
@@ -3330,7 +3323,7 @@
     return status;
 }
 
-status_t DeviceEffectProxy::checkPort(const AudioFlinger::PatchPanel::Patch& patch,
+status_t DeviceEffectProxy::checkPort(const IAfPatchPanel::Patch& patch,
         const struct audio_port_config *port, sp<IAfEffectHandle> *handle) {
 
     ALOGV("%s type %d device type %d address %s device ID %d patch.isSoftware() %d",
@@ -3376,7 +3369,7 @@
             mDevicePort.id = AUDIO_PORT_HANDLE_NONE;
         }
     } else if (patch.isSoftware() || patch.thread().promote() != nullptr) {
-        sp <AudioFlinger::ThreadBase> thread;
+        sp<IAfThreadBase> thread;
         if (audio_port_config_has_input_direction(port)) {
             if (patch.isSoftware()) {
                 thread = patch.mRecord.thread();
@@ -3489,7 +3482,7 @@
     const Vector<String16> args;
     EffectBase::dump(fd, args);
 
-    const bool locked = AudioFlinger::dumpTryLock(mProxyLock);
+    const bool locked = afutils::dumpTryLock(mProxyLock);
     if (!locked) {
         String8 result("DeviceEffectProxy may be deadlocked\n");
         write(fd, result.string(), result.size());
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index 07790be..82ad486 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -15,6 +15,17 @@
 ** limitations under the License.
 */
 
+#pragma once
+
+#include "DeviceEffectManager.h"
+#include "IAfEffect.h"
+
+#include <android-base/macros.h>  // DISALLOW_COPY_AND_ASSIGN
+#include <mediautils/Synchronization.h>
+#include <private/media/AudioEffectShared.h>
+
+#include <map>  // avoid transitive dependency
+
 namespace android {
 
 //--- Audio Effect Management
@@ -382,8 +393,7 @@
 // it also provide it's own input buffer used by the track as accumulation buffer.
 class EffectChain : public IAfEffectChain {
 public:
-    EffectChain(const wp<AudioFlinger::ThreadBase>& wThread, audio_session_t sessionId);
-    ~EffectChain() override;
+    EffectChain(const sp<IAfThreadBase>& thread, audio_session_t sessionId);
 
     void process_l() final;
 
@@ -479,12 +489,7 @@
     bool isBitPerfectCompatible() const final;
 
     // isCompatibleWithThread_l() must be called with thread->mLock held
-    // TODO(b/288339104) type
-    bool isCompatibleWithThread_l(const sp<Thread>& thread) const final {
-        return isCompatibleWithThread_l(sp<AudioFlinger::ThreadBase>::cast(thread));
-    }
-
-    bool isCompatibleWithThread_l(const sp<AudioFlinger::ThreadBase>& thread) const;
+    bool isCompatibleWithThread_l(const sp<IAfThreadBase>& thread) const final;
 
     bool containsHapticGeneratingEffect_l() final;
 
@@ -492,8 +497,7 @@
 
     sp<EffectCallbackInterface> effectCallback() const final { return mEffectCallback; }
 
-    // TODO(b/288339104) type
-    wp<Thread> thread() const final { return mEffectCallback->thread(); }
+    wp<IAfThreadBase> thread() const final { return mEffectCallback->thread(); }
 
     bool isFirstEffect(int id) const final {
         return !mEffects.isEmpty() && id == mEffects[0]->id();
@@ -507,12 +511,7 @@
         return mEffects[index];
     }
 
-    // TODO(b/288339104) type
-    void setThread(const sp<Thread>& thread) final {
-        setThread(sp<AudioFlinger::ThreadBase>::cast(thread));
-    }
-
-    void setThread(const sp<AudioFlinger::ThreadBase>& thread);
+    void setThread(const sp<IAfThreadBase>& thread) final;
 
 private:
 
@@ -527,16 +526,11 @@
         // Note: ctors taking a weak pointer to their owner must not promote it
         // during construction (but may keep a reference for later promotion).
         EffectCallback(const wp<EffectChain>& owner,
-                       const wp<AudioFlinger::ThreadBase>& thread)
+                const sp<IAfThreadBase>& thread)  // we take a sp<> but store a wp<>.
             : mChain(owner)
-            , mThread(thread)
-            , mAudioFlinger(*AudioFlinger::gAudioFlinger) {
-            sp<AudioFlinger::ThreadBase> base = thread.promote();
-            if (base != nullptr) {
-                mThreadType = base->type();
-            } else {
-                mThreadType = AudioFlinger::ThreadBase::MIXER;  // assure a consistent value.
-            }
+            , mThread(thread) {
+            mThreadType = thread->type();
+            mAfThreadCallback = thread->afThreadCallback();
         }
 
         status_t createEffectHal(const effect_uuid_t *pEffectUuid,
@@ -577,21 +571,22 @@
         wp<IAfEffectChain> chain() const final { return mChain; }
 
         bool isAudioPolicyReady() const final {
-            return mAudioFlinger.isAudioPolicyReady();
+            return mAfThreadCallback->isAudioPolicyReady();
         }
 
-        wp<AudioFlinger::ThreadBase> thread() const { return mThread.load(); }
+        wp<IAfThreadBase> thread() const { return mThread.load(); }
 
-        void setThread(const sp<AudioFlinger::ThreadBase>& thread) {
+        void setThread(const sp<IAfThreadBase>& thread) {
             mThread = thread;
             mThreadType = thread->type();
+            mAfThreadCallback = thread->afThreadCallback();
         }
 
     private:
         const wp<IAfEffectChain> mChain;
-        mediautils::atomic_wp<AudioFlinger::ThreadBase> mThread;
-        AudioFlinger &mAudioFlinger;  // implementation detail: outer instance always exists.
-        AudioFlinger::ThreadBase::type_t mThreadType;
+        mediautils::atomic_wp<IAfThreadBase> mThread;
+        sp<IAfThreadCallback> mAfThreadCallback;
+        IAfThreadBase::type_t mThreadType;
     };
 
     DISALLOW_COPY_AND_ASSIGN(EffectChain);
@@ -657,7 +652,7 @@
 class DeviceEffectProxy : public IAfDeviceEffectProxy, public EffectBase {
 public:
     DeviceEffectProxy(const AudioDeviceTypeAddr& device,
-                const sp<AudioFlinger::DeviceEffectManagerCallback>& callback,
+            const sp<DeviceEffectManagerCallback>& callback,
                 effect_descriptor_t *desc, int id, bool notifyFramesProcessed)
             : EffectBase(callback, desc, id, AUDIO_SESSION_DEVICE, false),
                 mDevice(device), mManagerCallback(callback),
@@ -667,22 +662,11 @@
     status_t setEnabled(bool enabled, bool fromHandle) final;
     sp<IAfDeviceEffectProxy> asDeviceEffectProxy() final { return this; }
 
-    // TODO(b/288339104) type
-    status_t init(const /* std::map<audio_patch_handle_t,
-            PatchPanel::Patch>& */ void * patches) final {
-        return init(*reinterpret_cast<const std::map<
-                audio_patch_handle_t, AudioFlinger::PatchPanel::Patch> *>(patches));
-    }
-    // TODO(b/288339104) type
-    status_t onCreatePatch(audio_patch_handle_t patchHandle,
-            /* const PatchPanel::Patch& */ const void * patch) final {
-        return onCreatePatch(patchHandle,
-                *reinterpret_cast<const AudioFlinger::PatchPanel::Patch *>(patch));
-    }
+    status_t init(const std::map<audio_patch_handle_t,
+            IAfPatchPanel::Patch>& patches) final;
 
-    status_t init(const std::map<audio_patch_handle_t, AudioFlinger::PatchPanel::Patch>& patches);
-    status_t onCreatePatch(
-            audio_patch_handle_t patchHandle, const AudioFlinger::PatchPanel::Patch& patch);
+    status_t onCreatePatch(audio_patch_handle_t patchHandle,
+            const IAfPatchPanel::Patch& patch) final;
 
     void onReleasePatch(audio_patch_handle_t patchHandle) final;
 
@@ -706,7 +690,7 @@
         // Note: ctors taking a weak pointer to their owner must not promote it
         // during construction (but may keep a reference for later promotion).
         ProxyCallback(const wp<DeviceEffectProxy>& owner,
-                const sp<AudioFlinger::DeviceEffectManagerCallback>& callback)
+                const sp<DeviceEffectManagerCallback>& callback)
             : mProxy(owner), mManagerCallback(callback) {}
 
         status_t createEffectHal(const effect_uuid_t *pEffectUuid,
@@ -757,14 +741,14 @@
 
     private:
         const wp<DeviceEffectProxy> mProxy;
-        const sp<AudioFlinger::DeviceEffectManagerCallback> mManagerCallback;
+        const sp<DeviceEffectManagerCallback> mManagerCallback;
     };
 
-    status_t checkPort(const AudioFlinger::PatchPanel::Patch& patch,
+    status_t checkPort(const IAfPatchPanel::Patch& patch,
             const struct audio_port_config *port, sp<IAfEffectHandle> *handle);
 
     const AudioDeviceTypeAddr mDevice;
-    const sp<AudioFlinger::DeviceEffectManagerCallback> mManagerCallback;
+    const sp<DeviceEffectManagerCallback> mManagerCallback;
     const sp<ProxyCallback> mMyCallback;
 
     mutable Mutex mProxyLock;
diff --git a/services/audioflinger/IAfEffect.h b/services/audioflinger/IAfEffect.h
index 75112ca..7393448 100644
--- a/services/audioflinger/IAfEffect.h
+++ b/services/audioflinger/IAfEffect.h
@@ -16,13 +16,28 @@
 
 #pragma once
 
+#include "IAfPatchPanel.h"  // full class Patch definition needed
+
+#include <android/media/AudioVibratorInfo.h>
+#include <android/media/BnEffect.h>
+#include <android/media/BnEffectClient.h>
+#include <media/AudioCommonTypes.h>  // product_strategy_t
+#include <media/AudioDeviceTypeAddr.h>
+#include <media/audiohal/EffectHalInterface.h>
+#include <utils/RefBase.h>
+#include <vibrator/ExternalVibration.h>
+
 namespace android {
 
+class Client;
+class DeviceEffectManagerCallback;
+
 class IAfDeviceEffectProxy;
 class IAfEffectBase;
 class IAfEffectChain;
 class IAfEffectHandle;
 class IAfEffectModule;
+class IAfThreadBase;
 
 // Interface implemented by the EffectModule parent or owner (e.g an EffectChain) to abstract
 // interactions between the EffectModule and the reset of the audio framework.
@@ -190,7 +205,7 @@
     // Most of these methods are accessed from AudioFlinger::Thread
 public:
     static sp<IAfEffectChain> create(
-            const wp<Thread /*ThreadBase*/>& wThread,  // TODO(b/288339104) type
+            const sp<IAfThreadBase>& thread,
             audio_session_t sessionId);
 
     // special key used for an entry in mSuspendedEffects keyed vector
@@ -279,8 +294,7 @@
     virtual bool isBitPerfectCompatible() const = 0;
 
     // isCompatibleWithThread_l() must be called with thread->mLock held
-    //  TODO(b/288339104) type
-    virtual bool isCompatibleWithThread_l(const sp<Thread>& thread) const = 0;
+    virtual bool isCompatibleWithThread_l(const sp<IAfThreadBase>& thread) const = 0;
 
     virtual bool containsHapticGeneratingEffect_l() = 0;
 
@@ -288,8 +302,8 @@
 
     virtual sp<EffectCallbackInterface> effectCallback() const = 0;
 
-    virtual wp<Thread> thread() const = 0;  // TODO(b/288339104) type
-    virtual void setThread(const sp<Thread>& thread) = 0;  // TODO(b/288339104) type
+    virtual wp<IAfThreadBase> thread() const = 0;
+    virtual void setThread(const sp<IAfThreadBase>& thread) = 0;
 
     virtual bool isFirstEffect(int id) const = 0;
 
@@ -335,22 +349,21 @@
 
 class IAfDeviceEffectProxy : public virtual IAfEffectBase {
 public:
-    // TODO(b/288339104) type
     static sp<IAfDeviceEffectProxy> create(const AudioDeviceTypeAddr& device,
-                const sp</* DeviceEffectManagerCallback */ RefBase>& callback,
+                const sp<DeviceEffectManagerCallback>& callback,
                 effect_descriptor_t *desc, int id, bool notifyFramesProcessed);
 
     virtual status_t init(
-            const /* std::map<audio_patch_handle_t,
-            PatchPanel::Patch>& */ void * patches) = 0; // TODO(b/288339104) type
+            const std::map<audio_patch_handle_t,
+            IAfPatchPanel::Patch>& patches) = 0;
     virtual const AudioDeviceTypeAddr& device() const = 0;
 
     virtual status_t onCreatePatch(
             audio_patch_handle_t patchHandle,
-            /* const PatchPanel::Patch& */ const void * patch) = 0;
+            const IAfPatchPanel::Patch& patch) = 0;
     virtual void onReleasePatch(audio_patch_handle_t patchHandle) = 0;
 
-    virtual void dump2(int fd, int spaces) const = 0; // TODO(b/288339104) naming?
+    virtual void dump2(int fd, int spaces) const = 0; // TODO(b/291319101) naming?
 
 private:
     // used by DeviceEffectProxy
@@ -364,4 +377,4 @@
     virtual status_t removeEffectFromHal(const sp<EffectHalInterface>& effect) = 0;
 };
 
-} // namespace android
+}  // namespace android
diff --git a/services/audioflinger/IAfPatchPanel.h b/services/audioflinger/IAfPatchPanel.h
new file mode 100644
index 0000000..20e092d
--- /dev/null
+++ b/services/audioflinger/IAfPatchPanel.h
@@ -0,0 +1,293 @@
+/*
+ * 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
+
+// The following includes are required because we have class definitions below
+// for EndPoint and Patch, which precludes using a forward declaration only.
+#include "IAfThread.h"  // IAfThreadBase IAfMmapThread IAfPlaybackThread IAfRecordThread
+#include "IAfTrack.h"   // IAfPatchRecord IAfPatchTrack
+
+#include <datapath/AudioHwDevice.h>
+#include <media/DeviceDescriptorBase.h>
+#include <utils/Log.h>      // ALOG used in this file
+#include <utils/RefBase.h>  // avoid transitive dependency
+#include <utils/Thread.h>
+
+namespace android {
+
+class IAfPatchPanel;
+class PatchCommandThread;
+
+class SoftwarePatch {
+public:
+    SoftwarePatch(
+            const sp<const IAfPatchPanel>& patchPanel,
+            audio_patch_handle_t patchHandle,
+            audio_io_handle_t playbackThreadHandle,
+            audio_io_handle_t recordThreadHandle)
+        : mPatchPanel(patchPanel),
+          mPatchHandle(patchHandle),
+          mPlaybackThreadHandle(playbackThreadHandle),
+          mRecordThreadHandle(recordThreadHandle) {}
+    SoftwarePatch(const SoftwarePatch&) = default;
+
+    // Must be called under AudioFlinger::mLock
+    status_t getLatencyMs_l(double* latencyMs) const;
+    audio_patch_handle_t getPatchHandle() const { return mPatchHandle; };
+    audio_io_handle_t getPlaybackThreadHandle() const { return mPlaybackThreadHandle; };
+    audio_io_handle_t getRecordThreadHandle() const { return mRecordThreadHandle; };
+
+private:
+    const sp<const IAfPatchPanel> mPatchPanel;
+    const audio_patch_handle_t mPatchHandle;
+    const audio_io_handle_t mPlaybackThreadHandle;
+    const audio_io_handle_t mRecordThreadHandle;
+};
+
+class IAfPatchPanelCallback : public virtual RefBase {
+public:
+    virtual void closeThreadInternal_l(const sp<IAfPlaybackThread>& thread) = 0;
+    virtual void closeThreadInternal_l(const sp<IAfRecordThread>& thread) = 0;
+    virtual IAfPlaybackThread* primaryPlaybackThread_l() const = 0;
+    virtual IAfPlaybackThread* checkPlaybackThread_l(audio_io_handle_t output) const = 0;
+    virtual IAfRecordThread* checkRecordThread_l(audio_io_handle_t input) const = 0;
+    virtual IAfMmapThread* checkMmapThread_l(audio_io_handle_t io) const = 0;
+    virtual sp<IAfThreadBase> openInput_l(audio_module_handle_t module,
+            audio_io_handle_t* input,
+            audio_config_t* config,
+            audio_devices_t device,
+            const char* address,
+            audio_source_t source,
+            audio_input_flags_t flags,
+            audio_devices_t outputDevice,
+            const String8& outputDeviceAddress) = 0;
+    virtual sp<IAfThreadBase> openOutput_l(audio_module_handle_t module,
+            audio_io_handle_t* output,
+            audio_config_t* halConfig,
+            audio_config_base_t* mixerConfig,
+            audio_devices_t deviceType,
+            const String8& address,
+            audio_output_flags_t flags) = 0;
+    virtual void lock() const = 0;
+    virtual void unlock() const = 0;
+    virtual const DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*>&
+            getAudioHwDevs_l() const = 0;
+    virtual audio_unique_id_t nextUniqueId(audio_unique_id_use_t use) = 0;
+    virtual const sp<PatchCommandThread>& getPatchCommandThread() = 0;
+    virtual void updateDownStreamPatches_l(
+            const struct audio_patch* patch, const std::set<audio_io_handle_t>& streams) = 0;
+    virtual void updateOutDevicesForRecordThreads_l(const DeviceDescriptorBaseVector& devices) = 0;
+};
+
+class IAfPatchPanel : public virtual RefBase {
+public:
+    static sp<IAfPatchPanel> create(const sp<IAfPatchPanelCallback>& afPatchPanelCallback);
+
+    // Extraction of inner Endpoint and Patch classes would require interfaces
+    // (in the Endpoint case a templated interface) but that seems
+    // excessive for now.  We keep them as inner classes until extraction
+    // is needed.
+    template <typename ThreadType, typename TrackType>
+    class Endpoint final {
+    public:
+        Endpoint() = default;
+        Endpoint(const Endpoint&) = delete;
+        Endpoint& operator=(const Endpoint& other) noexcept {
+            mThread = other.mThread;
+            mCloseThread = other.mCloseThread;
+            mHandle = other.mHandle;
+            mTrack = other.mTrack;
+            return *this;
+        }
+        Endpoint(Endpoint&& other) noexcept { swap(other); }
+        Endpoint& operator=(Endpoint&& other) noexcept {
+            swap(other);
+            return *this;
+        }
+        ~Endpoint() {
+            ALOGE_IF(
+                    mHandle != AUDIO_PATCH_HANDLE_NONE,
+                    "A non empty Patch Endpoint leaked, handle %d", mHandle);
+        }
+
+        status_t checkTrack(TrackType* trackOrNull) const {
+            if (trackOrNull == nullptr) return NO_MEMORY;
+            return trackOrNull->initCheck();
+        }
+        audio_patch_handle_t handle() const { return mHandle; }
+        sp<ThreadType> thread() const { return mThread; }
+        sp<TrackType> track() const { return mTrack; }
+        sp<const ThreadType> const_thread() const { return mThread; }
+        sp<const TrackType> const_track() const { return mTrack; }
+
+        void closeConnections(const sp<IAfPatchPanel>& panel) {
+            if (mHandle != AUDIO_PATCH_HANDLE_NONE) {
+                panel->releaseAudioPatch(mHandle);
+                mHandle = AUDIO_PATCH_HANDLE_NONE;
+            }
+            if (mThread != nullptr) {
+                if (mTrack != nullptr) {
+                    mThread->deletePatchTrack(mTrack);
+                }
+                if (mCloseThread) {
+                    panel->closeThreadInternal_l(mThread);
+                }
+            }
+        }
+        audio_patch_handle_t* handlePtr() { return &mHandle; }
+        void setThread(const sp<ThreadType>& thread, bool closeThread = true) {
+            mThread = thread;
+            mCloseThread = closeThread;
+        }
+        template <typename T>
+        void setTrackAndPeer(const sp<TrackType>& track, const sp<T>& peer, bool holdReference) {
+            mTrack = track;
+            mThread->addPatchTrack(mTrack);
+            mTrack->setPeerProxy(peer, holdReference);
+            mClearPeerProxy = holdReference;
+        }
+        void clearTrackPeer() {
+            if (mClearPeerProxy && mTrack) mTrack->clearPeerProxy();
+        }
+        void stopTrack() {
+            if (mTrack) mTrack->stop();
+        }
+
+        void swap(Endpoint& other) noexcept {
+            using std::swap;
+            swap(mThread, other.mThread);
+            swap(mCloseThread, other.mCloseThread);
+            swap(mClearPeerProxy, other.mClearPeerProxy);
+            swap(mHandle, other.mHandle);
+            swap(mTrack, other.mTrack);
+        }
+
+        friend void swap(Endpoint& a, Endpoint& b) noexcept { a.swap(b); }
+
+    private:
+        sp<ThreadType> mThread;
+        bool mCloseThread = true;
+        bool mClearPeerProxy = true;
+        audio_patch_handle_t mHandle = AUDIO_PATCH_HANDLE_NONE;
+        sp<TrackType> mTrack;
+    };
+
+    class Patch final {
+    public:
+        Patch(const struct audio_patch& patch, bool endpointPatch)
+            : mAudioPatch(patch), mIsEndpointPatch(endpointPatch) {}
+        Patch() = default;
+        ~Patch();
+        Patch(const Patch& other) noexcept {
+            mAudioPatch = other.mAudioPatch;
+            mHalHandle = other.mHalHandle;
+            mPlayback = other.mPlayback;
+            mRecord = other.mRecord;
+            mThread = other.mThread;
+            mIsEndpointPatch = other.mIsEndpointPatch;
+        }
+        Patch(Patch&& other) noexcept { swap(other); }
+        Patch& operator=(Patch&& other) noexcept {
+            swap(other);
+            return *this;
+        }
+
+        void swap(Patch& other) noexcept {
+            using std::swap;
+            swap(mAudioPatch, other.mAudioPatch);
+            swap(mHalHandle, other.mHalHandle);
+            swap(mPlayback, other.mPlayback);
+            swap(mRecord, other.mRecord);
+            swap(mThread, other.mThread);
+            swap(mIsEndpointPatch, other.mIsEndpointPatch);
+        }
+
+        friend void swap(Patch& a, Patch& b) noexcept { a.swap(b); }
+
+        status_t createConnections(const sp<IAfPatchPanel>& panel);
+        void clearConnections(const sp<IAfPatchPanel>& panel);
+        bool isSoftware() const {
+            return mRecord.handle() != AUDIO_PATCH_HANDLE_NONE ||
+                   mPlayback.handle() != AUDIO_PATCH_HANDLE_NONE;
+        }
+
+        void setThread(const sp<IAfThreadBase>& thread) { mThread = thread; }
+        wp<IAfThreadBase> thread() const { return mThread; }
+
+        // returns the latency of the patch (from record to playback).
+        status_t getLatencyMs(double* latencyMs) const;
+
+        String8 dump(audio_patch_handle_t myHandle) const;
+
+        // Note that audio_patch::id is only unique within a HAL module
+        struct audio_patch mAudioPatch;
+        // handle for audio HAL patch handle present only when the audio HAL version is >= 3.0
+        audio_patch_handle_t mHalHandle = AUDIO_PATCH_HANDLE_NONE;
+        // below members are used by a software audio patch connecting a source device from a
+        // given audio HW module to a sink device on an other audio HW module.
+        // 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<IAfPlaybackThread, IAfPatchTrack> mPlayback;
+        // connects source device to record thread input
+        Endpoint<IAfRecordThread, IAfPatchRecord> mRecord;
+
+        wp<IAfThreadBase> mThread;
+        bool mIsEndpointPatch;
+    };
+
+    /* List connected audio ports and their attributes */
+    virtual status_t listAudioPorts(unsigned int* num_ports, struct audio_port* ports) = 0;
+
+    /* Get supported attributes for a given audio port */
+    virtual status_t getAudioPort(struct audio_port_v7* port) = 0;
+
+    /* Create a patch between several source and sink ports */
+    virtual status_t createAudioPatch(
+            const struct audio_patch* patch,
+            audio_patch_handle_t* handle,
+            bool endpointPatch = false) = 0;
+
+    /* Release a patch */
+    virtual status_t releaseAudioPatch(audio_patch_handle_t handle) = 0;
+
+    /* List connected audio devices and they attributes */
+    virtual status_t listAudioPatches(unsigned int* num_patches, struct audio_patch* patches) = 0;
+
+    // Retrieves all currently estrablished software patches for a stream
+    // opened on an intermediate module.
+    virtual status_t getDownstreamSoftwarePatches(
+            audio_io_handle_t stream, std::vector<SoftwarePatch>* patches) const = 0;
+
+    // Notifies patch panel about all opened and closed streams.
+    virtual void notifyStreamOpened(
+            AudioHwDevice* audioHwDevice, audio_io_handle_t stream, struct audio_patch* patch) = 0;
+
+    virtual void notifyStreamClosed(audio_io_handle_t stream) = 0;
+
+    virtual void dump(int fd) const = 0;
+
+    // Must be called under AudioFlinger::mLock
+
+    virtual const std::map<audio_patch_handle_t, Patch>& patches_l() const = 0;
+
+    virtual status_t getLatencyMs_l(audio_patch_handle_t patchHandle, double* latencyMs) const = 0;
+
+    virtual void closeThreadInternal_l(const sp<IAfThreadBase>& thread) const = 0;
+};
+
+}  // namespace android
diff --git a/services/audioflinger/IAfThread.h b/services/audioflinger/IAfThread.h
new file mode 100644
index 0000000..235378b
--- /dev/null
+++ b/services/audioflinger/IAfThread.h
@@ -0,0 +1,592 @@
+/*
+ * 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
+
+#include <android/media/IAudioTrackCallback.h>
+#include <android/media/IEffectClient.h>
+#include <audiomanager/IAudioManager.h>
+#include <audio_utils/MelProcessor.h>
+#include <binder/MemoryDealer.h>
+#include <datapath/AudioStreamIn.h>
+#include <datapath/AudioStreamOut.h>
+#include <datapath/VolumeInterface.h>
+#include <fastpath/FastMixerDumpState.h>
+#include <media/DeviceDescriptorBase.h>
+#include <media/MmapStreamInterface.h>
+#include <media/audiohal/StreamHalInterface.h>
+#include <media/nblog/NBLog.h>
+#include <timing/SyncEvent.h>
+#include <utils/Mutex.h>
+#include <utils/RefBase.h>
+#include <vibrator/ExternalVibration.h>
+
+#include <optional>
+
+namespace android {
+
+class IAfDirectOutputThread;
+class IAfDuplicatingThread;
+class IAfMmapCaptureThread;
+class IAfMmapPlaybackThread;
+class IAfPlaybackThread;
+class IAfRecordThread;
+
+class IAfEffectChain;
+class IAfEffectHandle;
+class IAfEffectModule;
+class IAfPatchPanel;
+class IAfPatchRecord;
+class IAfPatchTrack;
+class IAfRecordTrack;
+class IAfTrack;
+class IAfTrackBase;
+class Client;
+class MelReporter;
+
+// Note this is exposed through IAfThreadBase::afThreadCallback()
+// and hence may be used by the Effect / Track framework.
+class IAfThreadCallback : public virtual RefBase {
+public:
+    virtual Mutex& mutex() const = 0;
+    virtual bool isNonOffloadableGlobalEffectEnabled_l() const = 0;  // Tracks
+    virtual audio_unique_id_t nextUniqueId(audio_unique_id_use_t use) = 0;
+    virtual bool btNrecIsOff() const = 0;
+    virtual float masterVolume_l() const = 0;
+    virtual bool masterMute_l() const = 0;
+    virtual float getMasterBalance_l() const = 0;
+    virtual bool streamMute_l(audio_stream_type_t stream) const = 0;
+    virtual audio_mode_t getMode() const = 0;
+    virtual bool isLowRamDevice() const = 0;
+    virtual bool isAudioPolicyReady() const = 0;  // Effects
+    virtual uint32_t getScreenState() const = 0;
+    virtual std::optional<media::AudioVibratorInfo> getDefaultVibratorInfo_l() const = 0;
+    virtual const sp<IAfPatchPanel>& getPatchPanel() const = 0;
+    virtual const sp<MelReporter>& getMelReporter() const = 0;
+    virtual const sp<EffectsFactoryHalInterface>& getEffectsFactoryHal() const = 0;
+    virtual sp<IAudioManager> getOrCreateAudioManager() = 0;  // Tracks
+
+    virtual bool updateOrphanEffectChains(const sp<IAfEffectModule>& effect) = 0;
+    virtual status_t moveEffectChain_l(audio_session_t sessionId,
+            IAfPlaybackThread* srcThread, IAfPlaybackThread* dstThread) = 0;
+
+    virtual void requestLogMerge() = 0;
+    virtual sp<NBLog::Writer> newWriter_l(size_t size, const char *name) = 0;
+    virtual void unregisterWriter(const sp<NBLog::Writer>& writer) = 0;
+
+    virtual sp<audioflinger::SyncEvent> createSyncEvent(AudioSystem::sync_event_t type,
+            audio_session_t triggerSession,
+            audio_session_t listenerSession,
+            const audioflinger::SyncEventCallback& callBack,
+            const wp<IAfTrackBase>& cookie) = 0;
+
+    virtual void ioConfigChanged(audio_io_config_event_t event,
+            const sp<AudioIoDescriptor>& ioDesc,
+            pid_t pid = 0) = 0;
+    virtual void onNonOffloadableGlobalEffectEnable() = 0;
+    virtual void onSupportedLatencyModesChanged(
+            audio_io_handle_t output, const std::vector<audio_latency_mode_t>& modes) = 0;
+};
+
+class IAfThreadBase : public virtual RefBase {
+public:
+    enum type_t {
+        MIXER,          // Thread class is MixerThread
+        DIRECT,         // Thread class is DirectOutputThread
+        DUPLICATING,    // Thread class is DuplicatingThread
+        RECORD,         // Thread class is RecordThread
+        OFFLOAD,        // Thread class is OffloadThread
+        MMAP_PLAYBACK,  // Thread class for MMAP playback stream
+        MMAP_CAPTURE,   // Thread class for MMAP capture stream
+        SPATIALIZER,    //
+        BIT_PERFECT,    // Thread class for BitPerfectThread
+        // When adding a value, also update IAfThreadBase::threadTypeToString()
+    };
+
+    static const char* threadTypeToString(type_t type);
+    virtual status_t readyToRun() = 0;
+    virtual void clearPowerManager() = 0;
+    virtual status_t initCheck() const = 0;
+    virtual type_t type() const = 0;
+    virtual bool isDuplicating() const = 0;
+    virtual audio_io_handle_t id() const = 0;
+    virtual uint32_t sampleRate() const = 0;
+    virtual audio_channel_mask_t channelMask() const = 0;
+    virtual audio_channel_mask_t mixerChannelMask() const = 0;
+    virtual audio_format_t format() const = 0;
+    virtual uint32_t channelCount() const = 0;
+
+    // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects,
+    // and returns the [normal mix] buffer's frame count.
+    virtual size_t frameCount() const = 0;
+    virtual audio_channel_mask_t hapticChannelMask() const = 0;
+    virtual uint32_t hapticChannelCount() const = 0;
+    virtual uint32_t latency_l() const = 0;
+    virtual void setVolumeForOutput_l(float left, float right) const = 0;
+
+    // Return's the HAL's frame count i.e. fast mixer buffer size.
+    virtual size_t frameCountHAL() const = 0;
+    virtual size_t frameSize() const = 0;
+    // Should be "virtual status_t requestExitAndWait()" and override same
+    // method in Thread, but Thread::requestExitAndWait() is not yet virtual.
+    virtual void exit() = 0;
+    virtual bool checkForNewParameter_l(const String8& keyValuePair, status_t& status) = 0;
+    virtual status_t setParameters(const String8& keyValuePairs) = 0;
+    virtual String8 getParameters(const String8& keys) = 0;
+    virtual void ioConfigChanged(
+            audio_io_config_event_t event, pid_t pid = 0,
+            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) = 0;
+
+    // sendConfigEvent_l() must be called with ThreadBase::mLock held
+    // Can temporarily release the lock if waiting for a reply from
+    // processConfigEvents_l().
+    // status_t sendConfigEvent_l(sp<ConfigEvent>& event);
+    virtual void sendIoConfigEvent(
+            audio_io_config_event_t event, pid_t pid = 0,
+            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) = 0;
+    virtual void sendIoConfigEvent_l(
+            audio_io_config_event_t event, pid_t pid = 0,
+            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) = 0;
+    virtual void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp) = 0;
+    virtual void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio, bool forApp) = 0;
+    virtual status_t sendSetParameterConfigEvent_l(const String8& keyValuePair) = 0;
+    virtual status_t sendCreateAudioPatchConfigEvent(
+            const struct audio_patch* patch, audio_patch_handle_t* handle) = 0;
+    virtual status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle) = 0;
+    virtual status_t sendUpdateOutDeviceConfigEvent(
+            const DeviceDescriptorBaseVector& outDevices) = 0;
+    virtual void sendResizeBufferConfigEvent_l(int32_t maxSharedAudioHistoryMs) = 0;
+    virtual void sendCheckOutputStageEffectsEvent() = 0;
+    virtual void sendCheckOutputStageEffectsEvent_l() = 0;
+    virtual void sendHalLatencyModesChangedEvent_l() = 0;
+
+    virtual void processConfigEvents_l() = 0;
+    virtual void setCheckOutputStageEffects() = 0;
+    virtual void cacheParameters_l() = 0;
+    virtual status_t createAudioPatch_l(
+            const struct audio_patch* patch, audio_patch_handle_t* handle) = 0;
+    virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0;
+    virtual void updateOutDevices(const DeviceDescriptorBaseVector& outDevices) = 0;
+    virtual void toAudioPortConfig(struct audio_port_config* config) = 0;
+    virtual void resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs) = 0;
+
+    // see note at declaration of mStandby, mOutDevice and mInDevice
+    virtual bool inStandby() const = 0;
+    virtual const DeviceTypeSet outDeviceTypes() const = 0;
+    virtual audio_devices_t inDeviceType() const = 0;
+    virtual DeviceTypeSet getDeviceTypes() const = 0;
+    virtual const AudioDeviceTypeAddrVector& outDeviceTypeAddrs() const = 0;
+    virtual const AudioDeviceTypeAddr& inDeviceTypeAddr() const = 0;
+    virtual bool isOutput() const = 0;
+    virtual bool isOffloadOrMmap() const = 0;
+    virtual sp<StreamHalInterface> stream() const = 0;
+    virtual sp<IAfEffectHandle> createEffect_l(
+            const sp<Client>& client,
+            const sp<media::IEffectClient>& effectClient,
+            int32_t priority,
+            audio_session_t sessionId,
+            effect_descriptor_t* desc,
+            int* enabled,
+            status_t* status /*non-NULL*/,
+            bool pinned,
+            bool probe,
+            bool notifyFramesProcessed) = 0;
+
+    // return values for hasAudioSession (bit field)
+    enum effect_state {
+        EFFECT_SESSION = 0x1,       // the audio session corresponds to at least one
+                                    // effect
+        TRACK_SESSION = 0x2,        // the audio session corresponds to at least one
+                                    // track
+        FAST_SESSION = 0x4,         // the audio session corresponds to at least one
+                                    // fast track
+        SPATIALIZED_SESSION = 0x8,  // the audio session corresponds to at least one
+                                    // spatialized track
+        BIT_PERFECT_SESSION = 0x10  // the audio session corresponds to at least one
+                                    // bit-perfect track
+    };
+
+    // get effect chain corresponding to session Id.
+    virtual sp<IAfEffectChain> getEffectChain(audio_session_t sessionId) const = 0;
+    // same as getEffectChain() but must be called with ThreadBase mutex locked
+    virtual sp<IAfEffectChain> getEffectChain_l(audio_session_t sessionId) const = 0;
+    virtual std::vector<int> getEffectIds_l(audio_session_t sessionId) const = 0;
+    // add an effect chain to the chain list (mEffectChains)
+    virtual status_t addEffectChain_l(const sp<IAfEffectChain>& chain) = 0;
+    // remove an effect chain from the chain list (mEffectChains)
+    virtual size_t removeEffectChain_l(const sp<IAfEffectChain>& chain) = 0;
+    // lock all effect chains Mutexes. Must be called before releasing the
+    // ThreadBase mutex before processing the mixer and effects. This guarantees the
+    // integrity of the chains during the process.
+    // Also sets the parameter 'effectChains' to current value of mEffectChains.
+    virtual void lockEffectChains_l(Vector<sp<IAfEffectChain>>& effectChains) = 0;
+    // unlock effect chains after process
+    virtual void unlockEffectChains(const Vector<sp<IAfEffectChain>>& effectChains) = 0;
+    // get a copy of mEffectChains vector
+    virtual Vector<sp<IAfEffectChain>> getEffectChains_l() const = 0;
+    // set audio mode to all effect chains
+    virtual void setMode(audio_mode_t mode) = 0;
+    // get effect module with corresponding ID on specified audio session
+    virtual sp<IAfEffectModule> getEffect(audio_session_t sessionId, int effectId) const = 0;
+    virtual sp<IAfEffectModule> getEffect_l(audio_session_t sessionId, int effectId) const = 0;
+    // add and effect module. Also creates the effect chain is none exists for
+    // the effects audio session. Only called in a context of moving an effect
+    // from one thread to another
+    virtual status_t addEffect_l(const sp<IAfEffectModule>& effect) = 0;
+    // remove and effect module. Also removes the effect chain is this was the last
+    // effect
+    virtual void removeEffect_l(const sp<IAfEffectModule>& effect, bool release = false) = 0;
+    // disconnect an effect handle from module and destroy module if last handle
+    virtual void disconnectEffectHandle(IAfEffectHandle* handle, bool unpinIfLast) = 0;
+    // detach all tracks connected to an auxiliary effect
+    virtual void detachAuxEffect_l(int effectId) = 0;
+    // returns a combination of:
+    // - EFFECT_SESSION if effects on this audio session exist in one chain
+    // - TRACK_SESSION if tracks on this audio session exist
+    // - FAST_SESSION if fast tracks on this audio session exist
+    // - SPATIALIZED_SESSION if spatialized tracks on this audio session exist
+    virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const = 0;
+    virtual uint32_t hasAudioSession(audio_session_t sessionId) const = 0;
+
+    // the value returned by default implementation is not important as the
+    // strategy is only meaningful for PlaybackThread which implements this method
+    virtual product_strategy_t getStrategyForSession_l(audio_session_t sessionId) const = 0;
+
+    // check if some effects must be suspended/restored when an effect is enabled
+    // or disabled
+    virtual void checkSuspendOnEffectEnabled(
+            bool enabled, audio_session_t sessionId, bool threadLocked) = 0;
+
+    virtual status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) = 0;
+    virtual bool isValidSyncEvent(const sp<audioflinger::SyncEvent>& event) const = 0;
+
+    // Return a reference to a per-thread heap which can be used to allocate IMemory
+    // objects that will be read-only to client processes, read/write to mediaserver,
+    // and shared by all client processes of the thread.
+    // The heap is per-thread rather than common across all threads, because
+    // clients can't be trusted not to modify the offset of the IMemory they receive.
+    // If a thread does not have such a heap, this method returns 0.
+    virtual sp<MemoryDealer> readOnlyHeap() const = 0;
+
+    virtual sp<IMemory> pipeMemory() const = 0;
+
+    virtual void systemReady() = 0;
+
+    // checkEffectCompatibility_l() must be called with ThreadBase::mLock held
+    virtual status_t checkEffectCompatibility_l(
+            const effect_descriptor_t* desc, audio_session_t sessionId) = 0;
+
+    virtual void broadcast_l() = 0;
+
+    virtual bool isTimestampCorrectionEnabled() const = 0;
+
+    virtual bool isMsdDevice() const = 0;
+
+    virtual void dump(int fd, const Vector<String16>& args) = 0;
+
+    // deliver stats to mediametrics.
+    virtual void sendStatistics(bool force) = 0;
+
+    virtual Mutex& mutex() const = 0;
+
+    virtual void onEffectEnable(const sp<IAfEffectModule>& effect) = 0;
+    virtual void onEffectDisable() = 0;
+
+    // invalidateTracksForAudioSession_l must be called with holding mLock.
+    virtual void invalidateTracksForAudioSession_l(audio_session_t sessionId) const = 0;
+    // Invalidate all the tracks with the given audio session.
+    virtual void invalidateTracksForAudioSession(audio_session_t sessionId) const = 0;
+
+    virtual bool isStreamInitialized() const = 0;
+    virtual void startMelComputation_l(const sp<audio_utils::MelProcessor>& processor) = 0;
+    virtual void stopMelComputation_l() = 0;
+
+    virtual product_strategy_t getStrategyForStream(audio_stream_type_t stream) const = 0;
+
+    virtual void setEffectSuspended_l(
+            const effect_uuid_t* type, bool suspend, audio_session_t sessionId) = 0;
+
+    // Dynamic cast to derived interface
+    virtual sp<IAfDirectOutputThread> asIAfDirectOutputThread() { return nullptr; }
+    virtual sp<IAfDuplicatingThread> asIAfDuplicatingThread() { return nullptr; }
+    virtual sp<IAfPlaybackThread> asIAfPlaybackThread() { return nullptr; }
+    virtual sp<IAfRecordThread> asIAfRecordThread() { return nullptr; }
+    virtual IAfThreadCallback* afThreadCallback() const = 0;
+};
+
+class IAfPlaybackThread : public virtual IAfThreadBase, public virtual VolumeInterface {
+public:
+    static sp<IAfPlaybackThread> createBitPerfectThread(
+            const sp<IAfThreadCallback>& afThreadCallback, AudioStreamOut* output,
+            audio_io_handle_t id, bool systemReady);
+
+    static sp<IAfPlaybackThread> createDirectOutputThread(
+            const sp<IAfThreadCallback>& afThreadCallback, AudioStreamOut* output,
+            audio_io_handle_t id, bool systemReady, const audio_offload_info_t& offloadInfo);
+
+    static sp<IAfPlaybackThread> createMixerThread(
+            const sp<IAfThreadCallback>& afThreadCallback, AudioStreamOut* output,
+            audio_io_handle_t id, bool systemReady, type_t type = MIXER,
+            audio_config_base_t* mixerConfig = nullptr);
+
+    static sp<IAfPlaybackThread> createOffloadThread(
+            const sp<IAfThreadCallback>& afThreadCallback, AudioStreamOut* output,
+            audio_io_handle_t id, bool systemReady, const audio_offload_info_t& offloadInfo);
+
+    static sp<IAfPlaybackThread> createSpatializerThread(
+            const sp<IAfThreadCallback>& afThreadCallback, AudioStreamOut* output,
+            audio_io_handle_t id, bool systemReady, audio_config_base_t* mixerConfig);
+
+    static constexpr int8_t kMaxTrackStopRetriesOffload = 2;
+
+    enum mixer_state {
+        MIXER_IDLE,            // no active tracks
+        MIXER_TRACKS_ENABLED,  // at least one active track, but no track has any data ready
+        MIXER_TRACKS_READY,    // at least one active track, and at least one track has data
+        MIXER_DRAIN_TRACK,     // drain currently playing track
+        MIXER_DRAIN_ALL,       // fully drain the hardware
+        // standby mode does not have an enum value
+        // suspend by audio policy manager is orthogonal to mixer state
+    };
+
+    // return estimated latency in milliseconds, as reported by HAL
+    virtual uint32_t latency() const = 0;  // should be in IAfThreadBase?
+
+    virtual uint32_t& fastTrackAvailMask_l() = 0;
+
+    virtual sp<IAfTrack> createTrack_l(
+            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* pFrameCount,
+            size_t* pNotificationFrameCount,
+            uint32_t notificationsPerBuffer,
+            float speed,
+            const sp<IMemory>& sharedBuffer,
+            audio_session_t sessionId,
+            audio_output_flags_t* flags,
+            pid_t creatorPid,
+            const AttributionSourceState& attributionSource,
+            pid_t tid,
+            status_t* status /*non-NULL*/,
+            audio_port_handle_t portId,
+            const sp<media::IAudioTrackCallback>& callback,
+            bool isSpatialized,
+            bool isBitPerfect) = 0;
+
+    virtual status_t addTrack_l(const sp<IAfTrack>& track) = 0;
+    virtual bool destroyTrack_l(const sp<IAfTrack>& track) = 0;
+    virtual bool isTrackActive(const sp<IAfTrack>& track) const = 0;
+    virtual void addOutputTrack_l(const sp<IAfTrack>& track) = 0;
+
+    virtual AudioStreamOut* getOutput_l() const = 0;
+    virtual AudioStreamOut* getOutput() const = 0;
+    virtual AudioStreamOut* clearOutput() = 0;
+
+    // a very large number of suspend() will eventually wraparound, but unlikely
+    virtual void suspend() = 0;
+    virtual void restore() = 0;
+    virtual bool isSuspended() const = 0;
+    virtual status_t getRenderPosition(uint32_t* halFrames, uint32_t* dspFrames) const = 0;
+    // Consider also removing and passing an explicit mMainBuffer initialization
+    // parameter to AF::IAfTrack::Track().
+    virtual float* sinkBuffer() const = 0;
+
+    virtual status_t attachAuxEffect(const sp<IAfTrack>& track, int EffectId) = 0;
+    virtual status_t attachAuxEffect_l(const sp<IAfTrack>& track, int EffectId) = 0;
+
+    // called with AudioFlinger lock held
+    virtual bool invalidateTracks_l(audio_stream_type_t streamType) = 0;
+    virtual bool invalidateTracks_l(std::set<audio_port_handle_t>& portIds) = 0;
+    virtual void invalidateTracks(audio_stream_type_t streamType) = 0;
+    // Invalidate tracks by a set of port ids. The port id will be removed from
+    // the given set if the corresponding track is found and invalidated.
+    virtual void invalidateTracks(std::set<audio_port_handle_t>& portIds) = 0;
+
+    virtual status_t getTimestamp_l(AudioTimestamp& timestamp) = 0;
+    virtual void addPatchTrack(const sp<IAfPatchTrack>& track) = 0;
+    virtual void deletePatchTrack(const sp<IAfPatchTrack>& track) = 0;
+
+    // Return the asynchronous signal wait time.
+    virtual int64_t computeWaitTimeNs_l() const = 0;
+    // returns true if the track is allowed to be added to the thread.
+    virtual bool isTrackAllowed_l(
+            audio_channel_mask_t channelMask, audio_format_t format, audio_session_t sessionId,
+            uid_t uid) const = 0;
+
+    virtual bool supportsHapticPlayback() const = 0;
+
+    virtual void setDownStreamPatch(const struct audio_patch* patch) = 0;
+
+    virtual IAfTrack* getTrackById_l(audio_port_handle_t trackId) = 0;
+
+    virtual bool hasMixer() const = 0;
+
+    virtual status_t setRequestedLatencyMode(audio_latency_mode_t mode) = 0;
+
+    virtual status_t getSupportedLatencyModes(std::vector<audio_latency_mode_t>* modes) = 0;
+
+    virtual status_t setBluetoothVariableLatencyEnabled(bool enabled) = 0;
+
+    virtual void setStandby() = 0;
+    virtual void setStandby_l() = 0;
+    virtual bool waitForHalStart() = 0;
+
+    virtual bool hasFastMixer() const = 0;
+    virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const = 0;
+    virtual const std::atomic<int64_t>& framesWritten() const = 0;
+
+    virtual bool usesHwAvSync() const = 0;
+};
+
+class IAfDirectOutputThread : public virtual IAfPlaybackThread {
+public:
+    virtual status_t selectPresentation(int presentationId, int programId) = 0;
+};
+
+class IAfDuplicatingThread : public virtual IAfPlaybackThread {
+public:
+    static sp<IAfDuplicatingThread> create(
+            const sp<IAfThreadCallback>& afThreadCallback, IAfPlaybackThread* mainThread,
+            audio_io_handle_t id, bool systemReady);
+
+    virtual void addOutputTrack(IAfPlaybackThread* thread) = 0;
+    virtual uint32_t waitTimeMs() const = 0;
+    virtual void removeOutputTrack(IAfPlaybackThread* thread) = 0;
+};
+
+class IAfRecordThread : public virtual IAfThreadBase {
+public:
+    static sp<IAfRecordThread> create(
+            const sp<IAfThreadCallback>& afThreadCallback, AudioStreamIn* input,
+            audio_io_handle_t id, bool systemReady);
+
+    virtual sp<IAfRecordTrack> createRecordTrack_l(
+            const sp<Client>& client,
+            const audio_attributes_t& attr,
+            uint32_t* pSampleRate,
+            audio_format_t format,
+            audio_channel_mask_t channelMask,
+            size_t* pFrameCount,
+            audio_session_t sessionId,
+            size_t* pNotificationFrameCount,
+            pid_t creatorPid,
+            const AttributionSourceState& attributionSource,
+            audio_input_flags_t* flags,
+            pid_t tid,
+            status_t* status /*non-NULL*/,
+            audio_port_handle_t portId,
+            int32_t maxSharedAudioHistoryMs) = 0;
+    virtual void destroyTrack_l(const sp<IAfRecordTrack>& track) = 0;
+    virtual void removeTrack_l(const sp<IAfRecordTrack>& track) = 0;
+
+    virtual status_t start(
+            IAfRecordTrack* recordTrack, AudioSystem::sync_event_t event,
+            audio_session_t triggerSession) = 0;
+
+    // ask the thread to stop the specified track, and
+    // return true if the caller should then do it's part of the stopping process
+    virtual bool stop(IAfRecordTrack* recordTrack) = 0;
+
+    virtual AudioStreamIn* getInput() const = 0;
+    virtual AudioStreamIn* clearInput() = 0;
+
+    virtual status_t getActiveMicrophones(
+            std::vector<media::MicrophoneInfoFw>* activeMicrophones) const = 0;
+    virtual status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction) = 0;
+    virtual status_t setPreferredMicrophoneFieldDimension(float zoom) = 0;
+
+    virtual void addPatchTrack(const sp<IAfPatchRecord>& record) = 0;
+    virtual void deletePatchTrack(const sp<IAfPatchRecord>& record) = 0;
+    virtual bool fastTrackAvailable() const = 0;
+    virtual void setFastTrackAvailable(bool available) = 0;
+
+    virtual void setRecordSilenced(audio_port_handle_t portId, bool silenced) = 0;
+    virtual bool hasFastCapture() const = 0;
+
+    virtual void checkBtNrec() = 0;
+    virtual uint32_t getInputFramesLost() const = 0;
+
+    virtual status_t shareAudioHistory(
+            const std::string& sharedAudioPackageName,
+            audio_session_t sharedSessionId = AUDIO_SESSION_NONE,
+            int64_t sharedAudioStartMs = -1) = 0;
+    virtual void resetAudioHistory_l() = 0;
+};
+
+class IAfMmapThread : public virtual IAfThreadBase {
+public:
+    // createIAudioTrackAdapter() is a static constructor which creates an
+    // MmapStreamInterface AIDL interface adapter from the MmapThread object that
+    // may be passed back to the client.
+    //
+    // Only one AIDL MmapStreamInterface interface adapter should be created per MmapThread.
+    static sp<MmapStreamInterface> createMmapStreamInterfaceAdapter(
+            const sp<IAfMmapThread>& mmapThread);
+
+    virtual void configure(
+            const audio_attributes_t* attr,
+            audio_stream_type_t streamType,
+            audio_session_t sessionId,
+            const sp<MmapStreamCallback>& callback,
+            audio_port_handle_t deviceId,
+            audio_port_handle_t portId) = 0;
+    virtual void disconnect() = 0;
+
+    // MmapStreamInterface handling (see adapter)
+    virtual status_t createMmapBuffer(
+            int32_t minSizeFrames, struct audio_mmap_buffer_info* info) = 0;
+    virtual status_t getMmapPosition(struct audio_mmap_position* position) const = 0;
+    virtual status_t start(
+            const AudioClient& client, const audio_attributes_t* attr,
+            audio_port_handle_t* handle) = 0;
+    virtual status_t stop(audio_port_handle_t handle) = 0;
+    virtual status_t standby() = 0;
+    virtual status_t getExternalPosition(uint64_t* position, int64_t* timeNanos) const = 0;
+    virtual status_t reportData(const void* buffer, size_t frameCount) = 0;
+
+    // TODO(b/291317898)  move to IAfThreadBase?
+    virtual void invalidateTracks(std::set<audio_port_handle_t>& portIds) = 0;
+
+    // Sets the UID records silence - TODO(b/291317898)  move to IAfMmapCaptureThread
+    virtual void setRecordSilenced(audio_port_handle_t portId, bool silenced) = 0;
+
+    virtual sp<IAfMmapPlaybackThread> asIAfMmapPlaybackThread() { return nullptr; }
+    virtual sp<IAfMmapCaptureThread> asIAfMmapCaptureThread() { return nullptr; }
+};
+
+class IAfMmapPlaybackThread : public virtual IAfMmapThread, public virtual VolumeInterface {
+public:
+    static sp<IAfMmapPlaybackThread> create(
+            const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
+            AudioHwDevice* hwDev, AudioStreamOut* output, bool systemReady);
+
+    virtual AudioStreamOut* clearOutput() = 0;
+};
+
+class IAfMmapCaptureThread : public virtual IAfMmapThread {
+public:
+    static sp<IAfMmapCaptureThread> create(
+            const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
+            AudioHwDevice* hwDev, AudioStreamIn* input, bool systemReady);
+
+    virtual AudioStreamIn* clearInput() = 0;
+};
+
+}  // namespace android
diff --git a/services/audioflinger/IAfTrack.h b/services/audioflinger/IAfTrack.h
new file mode 100644
index 0000000..2302e13
--- /dev/null
+++ b/services/audioflinger/IAfTrack.h
@@ -0,0 +1,607 @@
+/*
+ * 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
+
+#include <android/media/BnAudioRecord.h>
+#include <android/media/BnAudioTrack.h>
+#include <audiomanager/IAudioManager.h>
+#include <binder/IMemory.h>
+#include <fastpath/FastMixerDumpState.h>
+#include <media/AudioSystem.h>
+#include <media/VolumeShaper.h>
+#include <private/media/AudioTrackShared.h>
+#include <timing/SyncEvent.h>
+#include <timing/SynchronizedRecordState.h>
+#include <utils/RefBase.h>
+#include <vibrator/ExternalVibration.h>
+
+#include <vector>
+
+namespace android {
+
+class Client;
+class ResamplerBufferProvider;
+struct Source;
+
+class IAfDuplicatingThread;
+class IAfPatchRecord;
+class IAfPatchTrack;
+class IAfPlaybackThread;
+class IAfRecordThread;
+class IAfThreadBase;
+
+struct TeePatch {
+    sp<IAfPatchRecord> patchRecord;
+    sp<IAfPatchTrack> patchTrack;
+};
+
+using TeePatches = std::vector<TeePatch>;
+
+// Common interface to all Playback and Record tracks.
+class IAfTrackBase : public virtual RefBase {
+public:
+    enum track_state : int32_t {
+        IDLE,
+        FLUSHED,  // for PlaybackTracks only
+        STOPPED,
+        // next 2 states are currently used for fast tracks
+        // and offloaded tracks only
+        STOPPING_1,  // waiting for first underrun
+        STOPPING_2,  // waiting for presentation complete
+        RESUMING,    // for PlaybackTracks only
+        ACTIVE,
+        PAUSING,
+        PAUSED,
+        STARTING_1,  // for RecordTrack only
+        STARTING_2,  // for RecordTrack only
+    };
+
+    // where to allocate the data buffer
+    enum alloc_type {
+        ALLOC_CBLK,      // allocate immediately after control block
+        ALLOC_READONLY,  // allocate from a separate read-only heap per thread
+        ALLOC_PIPE,      // do not allocate; use the pipe buffer
+        ALLOC_LOCAL,     // allocate a local buffer
+        ALLOC_NONE,      // do not allocate:use the buffer passed to TrackBase constructor
+    };
+
+    enum track_type {
+        TYPE_DEFAULT,
+        TYPE_OUTPUT,
+        TYPE_PATCH,
+    };
+
+    virtual status_t initCheck() const = 0;
+    virtual status_t start(
+            AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
+            audio_session_t triggerSession = AUDIO_SESSION_NONE) = 0;
+    virtual void stop() = 0;
+    virtual sp<IMemory> getCblk() const = 0;
+    virtual audio_track_cblk_t* cblk() const = 0;
+    virtual audio_session_t sessionId() const = 0;
+    virtual uid_t uid() const = 0;
+    virtual pid_t creatorPid() const = 0;
+    virtual uint32_t sampleRate() const = 0;
+    virtual size_t frameSize() const = 0;
+    virtual audio_port_handle_t portId() const = 0;
+    virtual status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) = 0;
+    virtual track_state state() const = 0;
+    virtual void setState(track_state state) = 0;
+    virtual sp<IMemory> getBuffers() const = 0;
+    virtual void* buffer() const = 0;
+    virtual size_t bufferSize() const = 0;
+    virtual bool isFastTrack() const = 0;
+    virtual bool isDirect() const = 0;
+    virtual bool isOutputTrack() const = 0;
+    virtual bool isPatchTrack() const = 0;
+    virtual bool isExternalTrack() const = 0;
+
+    virtual void invalidate() = 0;
+    virtual bool isInvalid() const = 0;
+
+    virtual void terminate() = 0;
+    virtual bool isTerminated() const = 0;
+
+    virtual audio_attributes_t attributes() const = 0;
+    virtual bool isSpatialized() const = 0;
+    virtual bool isBitPerfect() const = 0;
+
+    // not currently implemented in TrackBase, but overridden.
+    virtual void destroy() {};  // MmapTrack doesn't implement.
+    virtual void appendDumpHeader(String8& result) const = 0;
+    virtual void appendDump(String8& result, bool active) const = 0;
+
+    // Dup with AudioBufferProvider interface
+    virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0;
+    virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer) = 0;
+
+    // Added for RecordTrack and OutputTrack
+    virtual wp<IAfThreadBase> thread() const = 0;
+    virtual const sp<ServerProxy>& serverProxy() const = 0;
+
+    // TEE_SINK
+    virtual void dumpTee(int fd __unused, const std::string& reason __unused) const {};
+
+    /** returns the buffer contents size converted to time in milliseconds
+     * for PCM Playback or Record streaming tracks. The return value is zero for
+     * PCM static tracks and not defined for non-PCM tracks.
+     *
+     * This may be called without the thread lock.
+     */
+    virtual double bufferLatencyMs() const = 0;
+
+    /** returns whether the track supports server latency computation.
+     * This is set in the constructor and constant throughout the track lifetime.
+     */
+    virtual bool isServerLatencySupported() const = 0;
+
+    /** computes the server latency for PCM Playback or Record track
+     * to the device sink/source.  This is the time for the next frame in the track buffer
+     * written or read from the server thread to the device source or sink.
+     *
+     * This may be called without the thread lock, but latencyMs and fromTrack
+     * may be not be synchronized. For example PatchPanel may not obtain the
+     * thread lock before calling.
+     *
+     * \param latencyMs on success is set to the latency in milliseconds of the
+     *        next frame written/read by the server thread to/from the track buffer
+     *        from the device source/sink.
+     * \param fromTrack on success is set to true if latency was computed directly
+     *        from the track timestamp; otherwise set to false if latency was
+     *        estimated from the server timestamp.
+     *        fromTrack may be nullptr or omitted if not required.
+     *
+     * \returns OK or INVALID_OPERATION on failure.
+     */
+    virtual status_t getServerLatencyMs(double* latencyMs, bool* fromTrack = nullptr) const = 0;
+
+    /** computes the total client latency for PCM Playback or Record tracks
+     * for the next client app access to the device sink/source; i.e. the
+     * server latency plus the buffer latency.
+     *
+     * This may be called without the thread lock, but latencyMs and fromTrack
+     * may be not be synchronized. For example PatchPanel may not obtain the
+     * thread lock before calling.
+     *
+     * \param latencyMs on success is set to the latency in milliseconds of the
+     *        next frame written/read by the client app to/from the track buffer
+     *        from the device sink/source.
+     * \param fromTrack on success is set to true if latency was computed directly
+     *        from the track timestamp; otherwise set to false if latency was
+     *        estimated from the server timestamp.
+     *        fromTrack may be nullptr or omitted if not required.
+     *
+     * \returns OK or INVALID_OPERATION on failure.
+     */
+    virtual status_t getTrackLatencyMs(double* latencyMs, bool* fromTrack = nullptr) const = 0;
+
+    // TODO: Consider making this external.
+    struct FrameTime {
+        int64_t frames;
+        int64_t timeNs;
+    };
+
+    // KernelFrameTime is updated per "mix" period even for non-pcm tracks.
+    virtual void getKernelFrameTime(FrameTime* ft) const = 0;
+
+    virtual audio_format_t format() const = 0;
+    virtual int id() const = 0;
+
+    virtual const char* getTrackStateAsString() const = 0;
+
+    // Called by the PlaybackThread to indicate that the track is becoming active
+    // and a new interval should start with a given device list.
+    virtual void logBeginInterval(const std::string& devices) = 0;
+
+    // Called by the PlaybackThread to indicate the track is no longer active.
+    virtual void logEndInterval() = 0;
+
+    // Called to tally underrun frames in playback.
+    virtual void tallyUnderrunFrames(size_t frames) = 0;
+
+    virtual audio_channel_mask_t channelMask() const = 0;
+
+    /** @return true if the track has changed (metadata or volume) since
+     *          the last time this function was called,
+     *          true if this function was never called since the track creation,
+     *          false otherwise.
+     *  Thread safe.
+     */
+    virtual bool readAndClearHasChanged() = 0;
+
+    /** Set that a metadata has changed and needs to be notified to backend. Thread safe. */
+    virtual void setMetadataHasChanged() = 0;
+
+    /**
+     * Called when a track moves to active state to record its contribution to battery usage.
+     * Track state transitions should eventually be handled within the track class.
+     */
+    virtual void beginBatteryAttribution() = 0;
+
+    /**
+     * Called when a track moves out of the active state to record its contribution
+     * to battery usage.
+     */
+    virtual void endBatteryAttribution() = 0;
+
+    /**
+     * For RecordTrack
+     * TODO(b/291317964) either use this or add asRecordTrack or asTrack etc.
+     */
+    virtual void handleSyncStartEvent(const sp<audioflinger::SyncEvent>& event __unused){};
+
+    // For Thread use, fast tracks and offloaded tracks only
+    // TODO(b/291317964) rearrange to IAfTrack.
+    virtual bool isStopped() const = 0;
+    virtual bool isStopping() const = 0;
+    virtual bool isStopping_1() const = 0;
+    virtual bool isStopping_2() const = 0;
+};
+
+// Common interface for Playback tracks.
+class IAfTrack : public virtual IAfTrackBase {
+public:
+    // 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).
+    //
+    // Only one AIDL IAudioTrack interface adapter should be created per Track.
+    static sp<media::IAudioTrack> createIAudioTrackAdapter(const sp<IAfTrack>& track);
+
+    static sp<IAfTrack> create(
+            IAfPlaybackThread* 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;
+    virtual bool isOffloaded() const = 0;
+    virtual bool isOffloadedOrDirect() const = 0;
+    virtual bool isStatic() const = 0;
+    virtual status_t setParameters(const String8& keyValuePairs) = 0;
+    virtual status_t selectPresentation(int presentationId, int programId) = 0;
+    virtual status_t attachAuxEffect(int EffectId) = 0;
+    virtual void setAuxBuffer(int EffectId, int32_t* buffer) = 0;
+    virtual int32_t* auxBuffer() const = 0;
+    virtual void setMainBuffer(float* buffer) = 0;
+    virtual float* mainBuffer() const = 0;
+    virtual int auxEffectId() const = 0;
+    virtual status_t getTimestamp(AudioTimestamp& timestamp) = 0;
+    virtual void signal() = 0;
+    virtual status_t getDualMonoMode(audio_dual_mono_mode_t* mode) const = 0;
+    virtual status_t setDualMonoMode(audio_dual_mono_mode_t mode) = 0;
+    virtual status_t getAudioDescriptionMixLevel(float* leveldB) const = 0;
+    virtual status_t setAudioDescriptionMixLevel(float leveldB) = 0;
+    virtual status_t getPlaybackRateParameters(audio_playback_rate_t* playbackRate) const = 0;
+    virtual status_t setPlaybackRateParameters(const audio_playback_rate_t& playbackRate) = 0;
+
+    // implement FastMixerState::VolumeProvider interface
+    virtual gain_minifloat_packed_t getVolumeLR() const = 0;
+
+    // implement volume handling.
+    virtual media::VolumeShaper::Status applyVolumeShaper(
+            const sp<media::VolumeShaper::Configuration>& configuration,
+            const sp<media::VolumeShaper::Operation>& operation) = 0;
+    virtual sp<media::VolumeShaper::State> getVolumeShaperState(int id) const = 0;
+    virtual sp<media::VolumeHandler> getVolumeHandler() const = 0;
+    /** Set the computed normalized final volume of the track.
+     * !masterMute * masterVolume * streamVolume * averageLRVolume */
+    virtual void setFinalVolume(float volumeLeft, float volumeRight) = 0;
+    virtual float getFinalVolume() const = 0;
+    virtual void getFinalVolume(float* left, float* right) const = 0;
+
+    using SourceMetadatas = std::vector<playback_track_metadata_v7_t>;
+    using MetadataInserter = std::back_insert_iterator<SourceMetadatas>;
+    /** Copy the track metadata in the provided iterator. Thread safe. */
+    virtual void copyMetadataTo(MetadataInserter& backInserter) const = 0;
+
+    /** Return haptic playback of the track is enabled or not, used in mixer. */
+    virtual bool getHapticPlaybackEnabled() const = 0;
+    /** Set haptic playback of the track is enabled or not, should be
+     * set after query or get callback from vibrator service */
+    virtual void setHapticPlaybackEnabled(bool hapticPlaybackEnabled) = 0;
+    /** Return at what intensity to play haptics, used in mixer. */
+    virtual os::HapticScale getHapticIntensity() const = 0;
+    /** Return the maximum amplitude allowed for haptics data, used in mixer. */
+    virtual float getHapticMaxAmplitude() const = 0;
+    /** Set intensity of haptic playback, should be set after querying vibrator service. */
+    virtual void setHapticIntensity(os::HapticScale hapticIntensity) = 0;
+    /** Set maximum amplitude allowed for haptic data, should be set after querying
+     *  vibrator service.
+     */
+    virtual void setHapticMaxAmplitude(float maxAmplitude) = 0;
+    virtual sp<os::ExternalVibration> getExternalVibration() const = 0;
+
+    // This function should be called with holding thread lock.
+    virtual void updateTeePatches_l() = 0;
+
+    // Argument teePatchesToUpdate is by value, use std::move to optimize.
+    virtual void setTeePatchesToUpdate_l(TeePatches teePatchesToUpdate) = 0;
+
+    static bool checkServerLatencySupported(audio_format_t format, audio_output_flags_t flags) {
+        return audio_is_linear_pcm(format) && (flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) == 0;
+    }
+
+    virtual audio_output_flags_t getOutputFlags() const = 0;
+    virtual float getSpeed() const = 0;
+
+    /**
+     * Updates the mute state and notifies the audio service. Call this only when holding player
+     * thread lock.
+     */
+    virtual void processMuteEvent_l(
+            const sp<IAudioManager>& audioManager, mute_state_t muteState) = 0;
+
+    virtual void triggerEvents(AudioSystem::sync_event_t type) = 0;
+
+    virtual void disable() = 0;
+    virtual int& fastIndex() = 0;
+    virtual bool isPlaybackRestricted() const = 0;
+
+    // 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/291317964) 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:
+    static sp<IAfOutputTrack> create(
+            IAfPlaybackThread* playbackThread,
+            IAfDuplicatingThread* 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;
+
+    /** Set the metadatas of the upstream tracks. Thread safe. */
+    virtual void setMetadatas(const SourceMetadatas& metadatas) = 0;
+    /** returns client timestamp to the upstream duplicating thread. */
+    virtual ExtendedTimestamp getClientProxyTimestamp() const = 0;
+};
+
+class IAfMmapTrack : public virtual IAfTrackBase {
+public:
+    static sp<IAfMmapTrack> create(IAfThreadBase* 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
+    virtual bool isSilenced_l() const = 0;
+    // protected by MMapThread::mLock
+    virtual bool getAndSetSilencedNotified_l() = 0;
+
+    /**
+     * Updates the mute state and notifies the audio service. Call this only when holding player
+     * thread lock.
+     */
+    virtual void processMuteEvent_l(  // see IAfTrack
+            const sp<IAudioManager>& audioManager, mute_state_t muteState) = 0;
+};
+
+class RecordBufferConverter;
+
+class IAfRecordTrack : public virtual IAfTrackBase {
+public:
+    // createIAudioRecordAdapter() is a static constructor which creates an
+    // IAudioRecord AIDL interface adapter from the RecordTrack object that
+    // may be passed back to the client (if needed).
+    //
+    // Only one AIDL IAudioRecord interface adapter should be created per RecordTrack.
+    static sp<media::IAudioRecord> createIAudioRecordAdapter(const sp<IAfRecordTrack>& recordTrack);
+
+    static sp<IAfRecordTrack> create(IAfRecordThread* 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
+    virtual bool setOverflow() = 0;
+
+    // TODO(b/291317964) handleSyncStartEvent in IAfTrackBase should move here.
+    virtual void clearSyncStartEvent() = 0;
+    virtual void updateTrackFrameInfo(
+            int64_t trackFramesReleased, int64_t sourceFramesRead, uint32_t halSampleRate,
+            const ExtendedTimestamp& timestamp) = 0;
+
+    virtual void setSilenced(bool silenced) = 0;
+    virtual bool isSilenced() const = 0;
+    virtual status_t getActiveMicrophones(
+            std::vector<media::MicrophoneInfoFw>* activeMicrophones) const = 0;
+
+    virtual status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction) = 0;
+    virtual status_t setPreferredMicrophoneFieldDimension(float zoom) = 0;
+    virtual status_t shareAudioHistory(
+            const std::string& sharedAudioPackageName, int64_t sharedAudioStartMs) = 0;
+    virtual int32_t startFrames() const = 0;
+
+    static bool checkServerLatencySupported(audio_format_t format, audio_input_flags_t flags) {
+        return audio_is_linear_pcm(format) && (flags & AUDIO_INPUT_FLAG_HW_AV_SYNC) == 0;
+    }
+
+    using SinkMetadatas = std::vector<record_track_metadata_v7_t>;
+    using MetadataInserter = std::back_insert_iterator<SinkMetadatas>;
+    virtual void copyMetadataTo(MetadataInserter& backInserter) const = 0; // see IAfTrack
+
+    // 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.
+// it provides buffer access methods that map those of a ClientProxy (see AudioTrackShared.h)
+class PatchProxyBufferProvider {
+public:
+    virtual ~PatchProxyBufferProvider() = default;
+    virtual bool producesBufferOnDemand() const = 0;
+    virtual status_t obtainBuffer(
+            Proxy::Buffer* buffer, const struct timespec* requested = nullptr) = 0;
+    virtual void releaseBuffer(Proxy::Buffer* buffer) = 0;
+};
+
+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 {
+public:
+    static sp<IAfPatchTrack> create(
+            IAfPlaybackThread* playbackThread,
+            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. */);
+};
+
+class IAfPatchRecord : public virtual IAfRecordTrack, public virtual IAfPatchTrackBase {
+public:
+    static sp<IAfPatchRecord> create(
+            IAfRecordThread* recordThread,
+            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(
+            IAfRecordThread* recordThread,
+            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;
+};
+
+}  // namespace android
diff --git a/services/audioflinger/MelReporter.cpp b/services/audioflinger/MelReporter.cpp
index 39f772b..64a5843 100644
--- a/services/audioflinger/MelReporter.cpp
+++ b/services/audioflinger/MelReporter.cpp
@@ -16,9 +16,9 @@
 */
 
 // #define LOG_NDEBUG 0
-#define LOG_TAG "AudioFlinger::MelReporter"
+#define LOG_TAG "MelReporter"
 
-#include "AudioFlinger.h"
+#include "MelReporter.h"
 
 #include <android/media/ISoundDoseCallback.h>
 #include <audio_utils/power.h>
@@ -28,7 +28,7 @@
 
 namespace android {
 
-bool AudioFlinger::MelReporter::activateHalSoundDoseComputation(const std::string& module,
+bool MelReporter::activateHalSoundDoseComputation(const std::string& module,
         const sp<DeviceHalInterface>& device) {
     if (mSoundDoseManager->forceUseFrameworkMel()) {
         ALOGD("%s: Forcing use of internal MEL computation.", __func__);
@@ -63,7 +63,7 @@
     return true;
 }
 
-void AudioFlinger::MelReporter::activateInternalSoundDoseComputation() {
+void MelReporter::activateInternalSoundDoseComputation() {
     {
         std::lock_guard _l(mLock);
         if (!mUseHalSoundDoseInterface) {
@@ -76,42 +76,20 @@
     mSoundDoseManager->setHalSoundDoseInterface(nullptr);
 }
 
-void AudioFlinger::MelReporter::onFirstRef() {
-    mAudioFlinger.mPatchCommandThread->addListener(this);
+void MelReporter::onFirstRef() {
+    mAfMelReporterCallback->getPatchCommandThread()->addListener(this);
+
+    mSoundDoseManager = sp<SoundDoseManager>::make(sp<IMelReporterCallback>::fromExisting(this));
 }
 
-bool AudioFlinger::MelReporter::shouldComputeMelForDeviceType(audio_devices_t device) {
-    if (!mSoundDoseManager->isCsdEnabled()) {
-        ALOGV("%s csd is disabled", __func__);
-        return false;
-    }
-    if (mSoundDoseManager->forceComputeCsdOnAllDevices()) {
-        return true;
-    }
-
-    switch (device) {
-        case AUDIO_DEVICE_OUT_WIRED_HEADSET:
-        case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
-        // TODO(b/278265907): enable A2DP when we can distinguish A2DP headsets
-        // case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
-        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
-        case AUDIO_DEVICE_OUT_USB_HEADSET:
-        case AUDIO_DEVICE_OUT_BLE_HEADSET:
-        case AUDIO_DEVICE_OUT_BLE_BROADCAST:
-            return true;
-        default:
-            return false;
-    }
-}
-
-void AudioFlinger::MelReporter::updateMetadataForCsd(audio_io_handle_t streamHandle,
+void MelReporter::updateMetadataForCsd(audio_io_handle_t streamHandle,
         const std::vector<playback_track_metadata_v7_t>& metadataVec) {
     if (!mSoundDoseManager->isCsdEnabled()) {
         ALOGV("%s csd is disabled", __func__);
         return;
     }
 
-    std::lock_guard _laf(mAudioFlinger.mLock);
+    std::lock_guard _laf(mAfMelReporterCallback->mutex());
     std::lock_guard _l(mLock);
     auto activeMelPatchId = activePatchStreamHandle_l(streamHandle);
     if (!activeMelPatchId) {
@@ -127,21 +105,22 @@
     }
 
     auto activeMelPatchIt = mActiveMelPatches.find(activeMelPatchId.value());
-    if (activeMelPatchIt != mActiveMelPatches.end()
-        && shouldActivateCsd != activeMelPatchIt->second.csdActive) {
-        if (activeMelPatchIt->second.csdActive) {
-            ALOGV("%s should not compute CSD for stream handle %d", __func__, streamHandle);
-            stopMelComputationForPatch_l(activeMelPatchIt->second);
-        } else {
-            ALOGV("%s should compute CSD for stream handle %d", __func__, streamHandle);
-            startMelComputationForActivePatch_l(activeMelPatchIt->second);
+    if (activeMelPatchIt != mActiveMelPatches.end()) {
+        if (shouldActivateCsd != activeMelPatchIt->second.csdActive) {
+            if (activeMelPatchIt->second.csdActive) {
+                ALOGV("%s should not compute CSD for stream handle %d", __func__, streamHandle);
+                stopMelComputationForPatch_l(activeMelPatchIt->second);
+            } else {
+                ALOGV("%s should compute CSD for stream handle %d", __func__, streamHandle);
+                startMelComputationForActivePatch_l(activeMelPatchIt->second);
+            }
+            activeMelPatchIt->second.csdActive = shouldActivateCsd;
         }
-        activeMelPatchIt->second.csdActive = shouldActivateCsd;
     }
 }
 
-void AudioFlinger::MelReporter::onCreateAudioPatch(audio_patch_handle_t handle,
-        const PatchPanel::Patch& patch) {
+void MelReporter::onCreateAudioPatch(audio_patch_handle_t handle,
+        const IAfPatchPanel::Patch& patch) {
     if (!mSoundDoseManager->isCsdEnabled()) {
         ALOGV("%s csd is disabled", __func__);
         return;
@@ -159,53 +138,81 @@
     audio_io_handle_t streamHandle = patch.mAudioPatch.sources[0].ext.mix.handle;
     ActiveMelPatch newPatch;
     newPatch.streamHandle = streamHandle;
+    newPatch.csdActive = false;
     for (size_t i = 0; i < patch.mAudioPatch.num_sinks; ++i) {
-        if (patch.mAudioPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE
-            && shouldComputeMelForDeviceType(patch.mAudioPatch.sinks[i].ext.device.type)) {
+        if (patch.mAudioPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE &&
+                mSoundDoseManager->shouldComputeCsdForDeviceType(
+                        patch.mAudioPatch.sinks[i].ext.device.type)) {
             audio_port_handle_t deviceId = patch.mAudioPatch.sinks[i].id;
-            newPatch.deviceHandles.push_back(deviceId);
+            bool shouldComputeCsd = mSoundDoseManager->shouldComputeCsdForDeviceWithAddress(
+                    patch.mAudioPatch.sinks[i].ext.device.type,
+                    patch.mAudioPatch.sinks[i].ext.device.address);
+            newPatch.deviceStates.push_back({deviceId, shouldComputeCsd});
+            newPatch.csdActive |= shouldComputeCsd;
             AudioDeviceTypeAddr adt{patch.mAudioPatch.sinks[i].ext.device.type,
                                     patch.mAudioPatch.sinks[i].ext.device.address};
             mSoundDoseManager->mapAddressToDeviceId(adt, deviceId);
         }
     }
 
-    if (!newPatch.deviceHandles.empty()) {
-        std::lock_guard _afl(mAudioFlinger.mLock);
+    if (!newPatch.deviceStates.empty() && newPatch.csdActive) {
+        std::lock_guard _afl(mAfMelReporterCallback->mutex());
         std::lock_guard _l(mLock);
         ALOGV("%s add patch handle %d to active devices", __func__, handle);
         startMelComputationForActivePatch_l(newPatch);
-        newPatch.csdActive = true;
         mActiveMelPatches[handle] = newPatch;
     }
 }
 
-void AudioFlinger::MelReporter::startMelComputationForActivePatch_l(const ActiveMelPatch& patch)
+void MelReporter::startMelComputationForActivePatch_l(const ActiveMelPatch& patch)
 NO_THREAD_SAFETY_ANALYSIS  // access of AudioFlinger::checkOutputThread_l
 {
-    auto outputThread = mAudioFlinger.checkOutputThread_l(patch.streamHandle);
+    auto outputThread = mAfMelReporterCallback->checkOutputThread_l(patch.streamHandle);
     if (outputThread == nullptr) {
         ALOGE("%s cannot find thread for stream handle %d", __func__, patch.streamHandle);
         return;
     }
 
-    for (const auto& deviceHandle : patch.deviceHandles) {
-        ++mActiveDevices[deviceHandle];
-        ALOGI("%s add stream %d that uses device %d for CSD, nr of streams: %d", __func__,
-              patch.streamHandle, deviceHandle, mActiveDevices[deviceHandle]);
+    for (const auto& device : patch.deviceStates) {
+        if (device.second) {
+            ++mActiveDevices[device.first];
+            ALOGI("%s add stream %d that uses device %d for CSD, nr of streams: %d", __func__,
+                  patch.streamHandle, device.first, mActiveDevices[device.first]);
 
-        if (outputThread != nullptr && !useHalSoundDoseInterface_l()) {
-            outputThread->startMelComputation_l(mSoundDoseManager->getOrCreateProcessorForDevice(
-                deviceHandle,
-                patch.streamHandle,
-                outputThread->mSampleRate,
-                outputThread->mChannelCount,
-                outputThread->mFormat));
+            if (outputThread != nullptr && !useHalSoundDoseInterface_l()) {
+                outputThread->startMelComputation_l(
+                        mSoundDoseManager->getOrCreateProcessorForDevice(
+                                device.first,
+                                patch.streamHandle,
+                                outputThread->sampleRate(),
+                                outputThread->channelCount(),
+                                outputThread->format()));
+            }
         }
     }
 }
 
-void AudioFlinger::MelReporter::onReleaseAudioPatch(audio_patch_handle_t handle) {
+void MelReporter::startMelComputationForDeviceId(audio_port_handle_t deviceId) {
+    ALOGV("%s(%d)", __func__, deviceId);
+    std::lock_guard _laf(mAfMelReporterCallback->mutex());
+    std::lock_guard _l(mLock);
+
+    for (auto& activeMelPatch : mActiveMelPatches) {
+        bool csdActive = false;
+        for (auto& device: activeMelPatch.second.deviceStates) {
+            if (device.first == deviceId && !device.second) {
+                device.second = true;
+            }
+            csdActive |= device.second;
+        }
+        if (csdActive && !activeMelPatch.second.csdActive) {
+            activeMelPatch.second.csdActive = csdActive;
+            startMelComputationForActivePatch_l(activeMelPatch.second);
+        }
+    }
+}
+
+void MelReporter::onReleaseAudioPatch(audio_patch_handle_t handle) {
     if (!mSoundDoseManager->isCsdEnabled()) {
         ALOGV("%s csd is disabled", __func__);
         return;
@@ -226,53 +233,70 @@
         mActiveMelPatches.erase(patchIt);
     }
 
-    std::lock_guard _afl(mAudioFlinger.mLock);
+    std::lock_guard _afl(mAfMelReporterCallback->mutex());
     std::lock_guard _l(mLock);
     stopMelComputationForPatch_l(melPatch);
 }
 
-sp<media::ISoundDose> AudioFlinger::MelReporter::getSoundDoseInterface(
+sp<media::ISoundDose> MelReporter::getSoundDoseInterface(
         const sp<media::ISoundDoseCallback>& callback) {
     // no need to lock since getSoundDoseInterface is synchronized
     return mSoundDoseManager->getSoundDoseInterface(callback);
 }
 
-void AudioFlinger::MelReporter::stopInternalMelComputation() {
+void MelReporter::stopInternalMelComputation() {
     ALOGV("%s", __func__);
     std::lock_guard _l(mLock);
     mActiveMelPatches.clear();
     mUseHalSoundDoseInterface = true;
 }
 
-void AudioFlinger::MelReporter::stopMelComputationForPatch_l(const ActiveMelPatch& patch)
+void MelReporter::stopMelComputationForPatch_l(const ActiveMelPatch& patch)
 NO_THREAD_SAFETY_ANALYSIS  // access of AudioFlinger::checkOutputThread_l
 {
-    if (!patch.csdActive) {
-        // no need to stop CSD inactive patches
-        return;
-    }
-
-    auto outputThread = mAudioFlinger.checkOutputThread_l(patch.streamHandle);
+    auto outputThread = mAfMelReporterCallback->checkOutputThread_l(patch.streamHandle);
 
     ALOGV("%s: stop MEL for stream id: %d", __func__, patch.streamHandle);
-    for (const auto& deviceId : patch.deviceHandles) {
-        if (mActiveDevices[deviceId] > 0) {
-            --mActiveDevices[deviceId];
-            if (mActiveDevices[deviceId] == 0) {
+    for (const auto& device : patch.deviceStates) {
+        if (mActiveDevices[device.first] > 0) {
+            --mActiveDevices[device.first];
+            if (mActiveDevices[device.first] == 0) {
                 // no stream is using deviceId anymore
-                ALOGI("%s removing device %d from active CSD devices", __func__, deviceId);
-                mSoundDoseManager->clearMapDeviceIdEntries(deviceId);
+                ALOGI("%s removing device %d from active CSD devices", __func__, device.first);
+                mSoundDoseManager->clearMapDeviceIdEntries(device.first);
             }
         }
     }
 
+    mSoundDoseManager->removeStreamProcessor(patch.streamHandle);
     if (outputThread != nullptr && !useHalSoundDoseInterface_l()) {
         outputThread->stopMelComputation_l();
     }
 }
 
+void MelReporter::stopMelComputationForDeviceId(audio_port_handle_t deviceId) {
+    ALOGV("%s(%d)", __func__, deviceId);
+    std::lock_guard _laf(mAfMelReporterCallback->mutex());
+    std::lock_guard _l(mLock);
 
-std::optional<audio_patch_handle_t> AudioFlinger::MelReporter::activePatchStreamHandle_l(
+    for (auto& activeMelPatch : mActiveMelPatches) {
+        bool csdActive = false;
+        for (auto& device: activeMelPatch.second.deviceStates) {
+            if (device.first == deviceId && device.second) {
+                device.second = false;
+            }
+            csdActive |= device.second;
+        }
+
+        if (!csdActive && activeMelPatch.second.csdActive) {
+            activeMelPatch.second.csdActive = csdActive;
+            stopMelComputationForPatch_l(activeMelPatch.second);
+        }
+    }
+
+}
+
+std::optional<audio_patch_handle_t> MelReporter::activePatchStreamHandle_l(
         audio_io_handle_t streamHandle) {
     for(const auto& patchIt : mActiveMelPatches) {
         if (patchIt.second.streamHandle == streamHandle) {
@@ -282,11 +306,11 @@
     return std::nullopt;
 }
 
-bool AudioFlinger::MelReporter::useHalSoundDoseInterface_l() {
+bool MelReporter::useHalSoundDoseInterface_l() {
     return !mSoundDoseManager->forceUseFrameworkMel() & mUseHalSoundDoseInterface;
 }
 
-std::string AudioFlinger::MelReporter::dump() {
+std::string MelReporter::dump() {
     std::lock_guard _l(mLock);
     std::string output("\nSound Dose:\n");
     output.append(mSoundDoseManager->dump());
diff --git a/services/audioflinger/MelReporter.h b/services/audioflinger/MelReporter.h
index 2bc33f2..78c6c0c 100644
--- a/services/audioflinger/MelReporter.h
+++ b/services/audioflinger/MelReporter.h
@@ -15,25 +15,36 @@
 ** limitations under the License.
 */
 
-#ifndef INCLUDING_FROM_AUDIOFLINGER_H
-    #error This header file should only be included from AudioFlinger.h
-#endif
+#pragma once
+
+#include "IAfPatchPanel.h"
+#include "PatchCommandThread.h"
+
+#include <sounddose/SoundDoseManager.h>
 
 #include <mutex>
-#include <sounddose/SoundDoseManager.h>
 #include <unordered_map>
 
+namespace android {
+
 constexpr static int kMaxTimestampDeltaInSec = 120;
 
+class IAfMelReporterCallback : public virtual RefBase {
+public:
+    virtual Mutex& mutex() const = 0;
+    virtual const sp<PatchCommandThread>& getPatchCommandThread() = 0;
+    virtual sp<IAfThreadBase> checkOutputThread_l(audio_io_handle_t ioHandle) const = 0;
+};
+
 /**
  * Class for listening to new patches and starting the MEL computation. MelReporter is
  * concealed within AudioFlinger, their lifetimes are the same.
  */
-class MelReporter : public PatchCommandThread::PatchCommandListener {
+class MelReporter : public PatchCommandThread::PatchCommandListener,
+                    public IMelReporterCallback {
 public:
-    explicit MelReporter(AudioFlinger& audioFlinger)
-        : mAudioFlinger(audioFlinger),
-          mSoundDoseManager(sp<SoundDoseManager>::make()) {}
+    explicit MelReporter(const sp<IAfMelReporterCallback>& afMelReporterCallback)
+        : mAfMelReporterCallback(afMelReporterCallback) {}
 
     void onFirstRef() override;
 
@@ -65,10 +76,14 @@
 
     std::string dump();
 
+    // IMelReporterCallback methods
+    void stopMelComputationForDeviceId(audio_port_handle_t deviceId) override;
+    void startMelComputationForDeviceId(audio_port_handle_t deviceId) override;
+
     // PatchCommandListener methods
     void onCreateAudioPatch(audio_patch_handle_t handle,
-                            const PatchPanel::Patch& patch) override;
-    void onReleaseAudioPatch(audio_patch_handle_t handle) override;
+        const IAfPatchPanel::Patch& patch) final;
+    void onReleaseAudioPatch(audio_patch_handle_t handle) final;
 
     /**
      * The new metadata can determine whether we should compute MEL for the given thread.
@@ -80,13 +95,15 @@
 private:
     struct ActiveMelPatch {
         audio_io_handle_t streamHandle{AUDIO_IO_HANDLE_NONE};
-        std::vector<audio_port_handle_t> deviceHandles;
+        /**
+         * Stores device ids and whether they are compatible for CSD calculation.
+         * The boolean value can change since BT audio device types are user-configurable
+         * to headphones/headsets or other device types.
+         */
+        std::vector<std::pair<audio_port_handle_t,bool>> deviceStates;
         bool csdActive;
     };
 
-    /** Returns true if we should compute MEL for the given device. */
-    bool shouldComputeMelForDeviceType(audio_devices_t device);
-
     void stopInternalMelComputation();
 
     /** Should be called with the following order of locks: mAudioFlinger.mLock -> mLock. */
@@ -100,7 +117,7 @@
 
     bool useHalSoundDoseInterface_l() REQUIRES(mLock);
 
-    AudioFlinger& mAudioFlinger;  // does not own the object
+    const sp<IAfMelReporterCallback> mAfMelReporterCallback;
 
     sp<SoundDoseManager> mSoundDoseManager;
 
@@ -109,9 +126,9 @@
      * Locking order AudioFlinger::mLock -> PatchCommandThread::mLock -> MelReporter::mLock.
      */
     std::mutex mLock;
-    std::unordered_map<audio_patch_handle_t, ActiveMelPatch>
-        mActiveMelPatches GUARDED_BY(AudioFlinger::MelReporter::mLock);
-    std::unordered_map<audio_port_handle_t, int>
-        mActiveDevices GUARDED_BY(AudioFlinger::MelReporter::mLock);
-    bool mUseHalSoundDoseInterface GUARDED_BY(AudioFlinger::MelReporter::mLock) = false;
+    std::unordered_map<audio_patch_handle_t, ActiveMelPatch> mActiveMelPatches GUARDED_BY(mLock);
+    std::unordered_map<audio_port_handle_t, int> mActiveDevices GUARDED_BY(mLock);
+    bool mUseHalSoundDoseInterface GUARDED_BY(mLock) = false;
 };
+
+}  // namespace android
diff --git a/services/audioflinger/MmapTracks.h b/services/audioflinger/MmapTracks.h
index cb46c52..85ce142 100644
--- a/services/audioflinger/MmapTracks.h
+++ b/services/audioflinger/MmapTracks.h
@@ -15,14 +15,18 @@
 ** limitations under the License.
 */
 
-#ifndef INCLUDING_FROM_AUDIOFLINGER_H
-    #error This header file should only be included from AudioFlinger.h
-#endif
+#pragma once
+
+#include "TrackBase.h"
+
+#include <android/content/AttributionSourceState.h>
+
+namespace android {
 
 // playback track
-class MmapTrack : public TrackBase {
+class MmapTrack : public TrackBase, public IAfMmapTrack {
 public:
-                MmapTrack(ThreadBase *thread,
+    MmapTrack(IAfThreadBase* thread,
                             const audio_attributes_t& attr,
                             uint32_t sampleRate,
                             audio_format_t format,
@@ -32,26 +36,25 @@
                             const android::content::AttributionSourceState& attributionSource,
                             pid_t creatorPid,
                             audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
-    virtual             ~MmapTrack();
+    ~MmapTrack() override;
 
-                        // TrackBase virtual
-    virtual status_t    initCheck() const;
-    virtual status_t    start(AudioSystem::sync_event_t event,
-                              audio_session_t triggerSession);
-    virtual void        stop();
-    virtual bool        isFastTrack() const { return false; }
-            bool        isDirect() const override { return true; }
+    status_t initCheck() const final;
+    status_t start(
+            AudioSystem::sync_event_t event, audio_session_t triggerSession) final;
+    void stop() final;
+    bool isFastTrack() const final { return false; }
+    bool isDirect() const final { return true; }
 
-            void        appendDumpHeader(String8& result);
-            void        appendDump(String8& result, bool active);
+    void appendDumpHeader(String8& result) const final;
+    void appendDump(String8& result, bool active) const final;
 
                         // protected by MMapThread::mLock
-            void        setSilenced_l(bool silenced) { mSilenced = silenced;
+    void setSilenced_l(bool silenced) final { mSilenced = silenced;
                                                        mSilencedNotified = false;}
                         // protected by MMapThread::mLock
-            bool        isSilenced_l() const { return mSilenced; }
+    bool isSilenced_l() const final { return mSilenced; }
                         // protected by MMapThread::mLock
-            bool        getAndSetSilencedNotified_l() { bool silencedNotified = mSilencedNotified;
+    bool getAndSetSilencedNotified_l() final { bool silencedNotified = mSilencedNotified;
                                                         mSilencedNotified = true;
                                                         return silencedNotified; }
 
@@ -61,10 +64,8 @@
      */
     void processMuteEvent_l(const sp<IAudioManager>& audioManager,
                             mute_state_t muteState)
-                            REQUIRES(AudioFlinger::MmapPlaybackThread::mLock);
+                            /* REQUIRES(MmapPlaybackThread::mLock) */ final;
 private:
-    friend class MmapThread;
-
     DISALLOW_COPY_AND_ASSIGN(MmapTrack);
 
     // AudioBufferProvider interface
@@ -72,19 +73,20 @@
     // releaseBuffer() not overridden
 
     // ExtendedAudioBufferProvider interface
-    virtual size_t framesReady() const;
-    virtual int64_t framesReleased() const;
-    virtual void onTimestamp(const ExtendedTimestamp &timestamp);
+    size_t framesReady() const final;
+    int64_t framesReleased() const final;
+    void onTimestamp(const ExtendedTimestamp &timestamp) final;
 
-    pid_t mPid;
+    const pid_t mPid;
     bool  mSilenced;            // protected by MMapThread::mLock
     bool  mSilencedNotified;    // protected by MMapThread::mLock
 
     // TODO: replace PersistableBundle with own struct
     // access these two variables only when holding player thread lock.
     std::unique_ptr<os::PersistableBundle> mMuteEventExtras
-            GUARDED_BY(AudioFlinger::MmapPlaybackThread::mLock);
+            /* GUARDED_BY(MmapPlaybackThread::mLock) */;
     mute_state_t mMuteState
-            GUARDED_BY(AudioFlinger::MmapPlaybackThread::mLock);
+            /* GUARDED_BY(MmapPlaybackThread::mLock) */;
 };  // end of Track
 
+} // namespace android
\ No newline at end of file
diff --git a/services/audioflinger/PatchCommandThread.cpp b/services/audioflinger/PatchCommandThread.cpp
index f4aab1f..c3259f1 100644
--- a/services/audioflinger/PatchCommandThread.cpp
+++ b/services/audioflinger/PatchCommandThread.cpp
@@ -18,31 +18,33 @@
 #define LOG_TAG "AudioFlinger::PatchCommandThread"
 //#define LOG_NDEBUG 0
 
-#include "AudioFlinger.h"
+#include "PatchCommandThread.h"
+
+#include <utils/Log.h>
 
 namespace android {
 
 constexpr char kPatchCommandThreadName[] = "AudioFlinger_PatchCommandThread";
 
-AudioFlinger::PatchCommandThread::~PatchCommandThread() {
+PatchCommandThread::~PatchCommandThread() {
     exit();
 
     std::lock_guard _l(mLock);
     mCommands.clear();
 }
 
-void AudioFlinger::PatchCommandThread::onFirstRef() {
+void PatchCommandThread::onFirstRef() {
     run(kPatchCommandThreadName, ANDROID_PRIORITY_AUDIO);
 }
 
-void AudioFlinger::PatchCommandThread::addListener(const sp<PatchCommandListener>& listener) {
+void PatchCommandThread::addListener(const sp<PatchCommandListener>& listener) {
     ALOGV("%s add listener %p", __func__, static_cast<void*>(listener.get()));
     std::lock_guard _l(mListenerLock);
     mListeners.emplace_back(listener);
 }
 
-void AudioFlinger::PatchCommandThread::createAudioPatch(audio_patch_handle_t handle,
-        const PatchPanel::Patch& patch) {
+void PatchCommandThread::createAudioPatch(audio_patch_handle_t handle,
+        const IAfPatchPanel::Patch& patch) {
     ALOGV("%s handle %d mHalHandle %d num sinks %d device sink %08x",
             __func__, handle, patch.mHalHandle,
             patch.mAudioPatch.num_sinks,
@@ -51,12 +53,12 @@
     createAudioPatchCommand(handle, patch);
 }
 
-void AudioFlinger::PatchCommandThread::releaseAudioPatch(audio_patch_handle_t handle) {
+void PatchCommandThread::releaseAudioPatch(audio_patch_handle_t handle) {
     ALOGV("%s", __func__);
     releaseAudioPatchCommand(handle);
 }
 
-bool AudioFlinger::PatchCommandThread::threadLoop()
+bool PatchCommandThread::threadLoop()
 NO_THREAD_SAFETY_ANALYSIS  // bug in clang compiler.
 {
     std::unique_lock _l(mLock);
@@ -119,14 +121,14 @@
     return false;
 }
 
-void AudioFlinger::PatchCommandThread::sendCommand(const sp<Command>& command) {
+void PatchCommandThread::sendCommand(const sp<Command>& command) {
     std::lock_guard _l(mLock);
     mCommands.emplace_back(command);
     mWaitWorkCV.notify_one();
 }
 
-void AudioFlinger::PatchCommandThread::createAudioPatchCommand(
-        audio_patch_handle_t handle, const PatchPanel::Patch& patch) {
+void PatchCommandThread::createAudioPatchCommand(
+        audio_patch_handle_t handle, const IAfPatchPanel::Patch& patch) {
     auto command = sp<Command>::make(CREATE_AUDIO_PATCH,
                                      new CreateAudioPatchData(handle, patch));
     ALOGV("%s adding create patch handle %d mHalHandle %d.",
@@ -136,14 +138,14 @@
     sendCommand(command);
 }
 
-void AudioFlinger::PatchCommandThread::releaseAudioPatchCommand(audio_patch_handle_t handle) {
+void PatchCommandThread::releaseAudioPatchCommand(audio_patch_handle_t handle) {
     sp<Command> command =
         sp<Command>::make(RELEASE_AUDIO_PATCH, new ReleaseAudioPatchData(handle));
     ALOGV("%s adding release patch", __func__);
     sendCommand(command);
 }
 
-void AudioFlinger::PatchCommandThread::exit() {
+void PatchCommandThread::exit() {
     ALOGV("%s", __func__);
     {
         std::lock_guard _l(mLock);
diff --git a/services/audioflinger/PatchCommandThread.h b/services/audioflinger/PatchCommandThread.h
index b52e0a9..a312fb7 100644
--- a/services/audioflinger/PatchCommandThread.h
+++ b/services/audioflinger/PatchCommandThread.h
@@ -15,14 +15,22 @@
 ** limitations under the License.
 */
 
-#ifndef INCLUDING_FROM_AUDIOFLINGER_H
-    #error This header file should only be included from AudioFlinger.h
-#endif
+#pragma once
+
+#include "IAfPatchPanel.h"
+
+#include <utils/RefBase.h>  // avoid transitive dependency
+#include <utils/Thread.h>  // avoid transitive dependency
+
+#include <deque>
+#include <mutex>  // avoid transitive dependency
+
+namespace android {
 
 class Command;
 
 // Thread to execute create and release patch commands asynchronously. This is needed because
-// PatchPanel::createAudioPatch and releaseAudioPatch are executed from audio policy service
+// IAfPatchPanel::createAudioPatch and releaseAudioPatch are executed from audio policy service
 // with mutex locked and effect management requires to call back into audio policy service
 class PatchCommandThread : public Thread {
 public:
@@ -35,7 +43,7 @@
     class PatchCommandListener : public virtual RefBase {
     public:
         virtual void onCreateAudioPatch(audio_patch_handle_t handle,
-                                        const PatchPanel::Patch& patch) = 0;
+                                        const IAfPatchPanel::Patch& patch) = 0;
         virtual void onReleaseAudioPatch(audio_patch_handle_t handle) = 0;
     };
 
@@ -44,7 +52,7 @@
 
     void addListener(const sp<PatchCommandListener>& listener);
 
-    void createAudioPatch(audio_patch_handle_t handle, const PatchPanel::Patch& patch);
+    void createAudioPatch(audio_patch_handle_t handle, const IAfPatchPanel::Patch& patch);
     void releaseAudioPatch(audio_patch_handle_t handle);
 
     // Thread virtuals
@@ -54,7 +62,7 @@
     void exit();
 
     void createAudioPatchCommand(audio_patch_handle_t handle,
-            const PatchPanel::Patch& patch);
+            const IAfPatchPanel::Patch& patch);
     void releaseAudioPatchCommand(audio_patch_handle_t handle);
 
 private:
@@ -75,11 +83,11 @@
 
     class CreateAudioPatchData : public CommandData {
     public:
-        CreateAudioPatchData(audio_patch_handle_t handle, const PatchPanel::Patch& patch)
+        CreateAudioPatchData(audio_patch_handle_t handle, const IAfPatchPanel::Patch& patch)
             :   mHandle(handle), mPatch(patch) {}
 
         const audio_patch_handle_t mHandle;
-        const PatchPanel::Patch mPatch;
+        const IAfPatchPanel::Patch mPatch;
     };
 
     class ReleaseAudioPatchData : public CommandData {
@@ -100,3 +108,5 @@
     std::mutex mListenerLock;
     std::vector<wp<PatchCommandListener>> mListeners GUARDED_BY(mListenerLock);
 };
+
+}  // namespace android
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index d0feba5..ec96de5 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -19,16 +19,16 @@
 #define LOG_TAG "AudioFlinger::PatchPanel"
 //#define LOG_NDEBUG 0
 
-#include "Configuration.h"
-#include <utils/Log.h>
-#include <audio_utils/primitives.h>
+#include "PatchPanel.h"
+#include "PatchCommandThread.h"
 
-#include "AudioFlinger.h"
+#include <audio_utils/primitives.h>
 #include <media/AudioParameter.h>
 #include <media/AudioValidator.h>
 #include <media/DeviceDescriptorBase.h>
 #include <media/PatchBuilder.h>
 #include <mediautils/ServiceUtilities.h>
+#include <utils/Log.h>
 
 // ----------------------------------------------------------------------------
 
@@ -47,65 +47,43 @@
 
 namespace android {
 
-/* List connected audio ports and their attributes */
-status_t AudioFlinger::listAudioPorts(unsigned int *num_ports,
-                                struct audio_port *ports)
-{
-    Mutex::Autolock _l(mLock);
-    return mPatchPanel.listAudioPorts(num_ports, ports);
+/* static */
+sp<IAfPatchPanel> IAfPatchPanel::create(const sp<IAfPatchPanelCallback>& afPatchPanelCallback) {
+    return sp<PatchPanel>::make(afPatchPanelCallback);
 }
 
-/* Get supported attributes for a given audio port */
-status_t AudioFlinger::getAudioPort(struct audio_port_v7 *port) {
-    status_t status = AudioValidator::validateAudioPort(*port);
-    if (status != NO_ERROR) {
-        return status;
-    }
-
-    Mutex::Autolock _l(mLock);
-    return mPatchPanel.getAudioPort(port);
+status_t SoftwarePatch::getLatencyMs_l(double* latencyMs) const {
+    return mPatchPanel->getLatencyMs_l(mPatchHandle, latencyMs);
 }
 
-/* Connect a patch between several source and sink ports */
-status_t AudioFlinger::createAudioPatch(const struct audio_patch *patch,
-                                   audio_patch_handle_t *handle)
+status_t PatchPanel::getLatencyMs_l(
+        audio_patch_handle_t patchHandle, double* latencyMs) const
 {
-    status_t status = AudioValidator::validateAudioPatch(*patch);
-    if (status != NO_ERROR) {
-        return status;
-    }
-
-    Mutex::Autolock _l(mLock);
-    return mPatchPanel.createAudioPatch(patch, handle);
-}
-
-/* Disconnect a patch */
-status_t AudioFlinger::releaseAudioPatch(audio_patch_handle_t handle)
-{
-    Mutex::Autolock _l(mLock);
-    return mPatchPanel.releaseAudioPatch(handle);
-}
-
-/* List connected audio ports and they attributes */
-status_t AudioFlinger::listAudioPatches(unsigned int *num_patches,
-                                  struct audio_patch *patches)
-{
-    Mutex::Autolock _l(mLock);
-    return mPatchPanel.listAudioPatches(num_patches, patches);
-}
-
-status_t AudioFlinger::PatchPanel::SoftwarePatch::getLatencyMs_l(double *latencyMs) const
-{
-    const auto& iter = mPatchPanel.mPatches.find(mPatchHandle);
-    if (iter != mPatchPanel.mPatches.end()) {
+    const auto& iter = mPatches.find(patchHandle);
+    if (iter != mPatches.end()) {
         return iter->second.getLatencyMs(latencyMs);
     } else {
         return BAD_VALUE;
     }
 }
 
+void PatchPanel::closeThreadInternal_l(const sp<IAfThreadBase>& thread) const
+{
+    if (const auto recordThread = thread->asIAfRecordThread();
+            recordThread) {
+        mAfPatchPanelCallback->closeThreadInternal_l(recordThread);
+    } else if (const auto playbackThread = thread->asIAfPlaybackThread();
+            playbackThread) {
+        mAfPatchPanelCallback->closeThreadInternal_l(playbackThread);
+    } else {
+        LOG_ALWAYS_FATAL("%s: Endpoints only accept IAfPlayback and IAfRecord threads, "
+                "invalid thread, id: %d  type: %d",
+                __func__, thread->id(), thread->type());
+    }
+}
+
 /* List connected audio ports and their attributes */
-status_t AudioFlinger::PatchPanel::listAudioPorts(unsigned int *num_ports __unused,
+status_t PatchPanel::listAudioPorts(unsigned int* /* num_ports */,
                                 struct audio_port *ports __unused)
 {
     ALOGV(__func__);
@@ -113,7 +91,7 @@
 }
 
 /* Get supported attributes for a given audio port */
-status_t AudioFlinger::PatchPanel::getAudioPort(struct audio_port_v7 *port)
+status_t PatchPanel::getAudioPort(struct audio_port_v7* port)
 {
     if (port->type != AUDIO_PORT_TYPE_DEVICE) {
         // Only query the HAL when the port is a device.
@@ -132,10 +110,10 @@
 }
 
 /* Connect a patch between several source and sink ports */
-status_t AudioFlinger::PatchPanel::createAudioPatch(const struct audio_patch *patch,
+status_t PatchPanel::createAudioPatch(const struct audio_patch* patch,
                                    audio_patch_handle_t *handle,
                                    bool endpointPatch)
- //unlocks AudioFlinger::mLock when calling ThreadBase::sendCreateAudioPatchConfigEvent
+ //unlocks AudioFlinger::mLock when calling IAfThreadBase::sendCreateAudioPatchConfigEvent
  //to avoid deadlocks if the thread loop needs to acquire AudioFlinger::mLock
  //before processing the create patch request.
  NO_THREAD_SAFETY_ANALYSIS
@@ -249,8 +227,8 @@
                         status = INVALID_OPERATION;
                         goto exit;
                     }
-                    sp<ThreadBase> thread =
-                            mAudioFlinger.checkPlaybackThread_l(patch->sources[1].ext.mix.handle);
+                    const sp<IAfThreadBase> thread = mAfPatchPanelCallback->checkPlaybackThread_l(
+                            patch->sources[1].ext.mix.handle);
                     if (thread == 0) {
                         ALOGW("%s() cannot get playback thread", __func__);
                         status = INVALID_OPERATION;
@@ -258,7 +236,7 @@
                     }
                     // existing playback thread is reused, so it is not closed when patch is cleared
                     newPatch.mPlayback.setThread(
-                            reinterpret_cast<PlaybackThread*>(thread.get()), false /*closeThread*/);
+                            thread->asIAfPlaybackThread().get(), false /*closeThread*/);
                 } else {
                     audio_config_t config = AUDIO_CONFIG_INITIALIZER;
                     audio_config_base_t mixerConfig = AUDIO_CONFIG_BASE_INITIALIZER;
@@ -276,7 +254,7 @@
                     if (patch->sinks[0].config_mask & AUDIO_PORT_CONFIG_FLAGS) {
                         flags = patch->sinks[0].flags.output;
                     }
-                    sp<ThreadBase> thread = mAudioFlinger.openOutput_l(
+                    const sp<IAfThreadBase> thread = mAfPatchPanelCallback->openOutput_l(
                                                             patch->sinks[0].ext.device.hw_module,
                                                             &output,
                                                             &config,
@@ -284,12 +262,12 @@
                                                             outputDevice,
                                                             outputDeviceAddress,
                                                             flags);
-                    ALOGV("mAudioFlinger.openOutput_l() returned %p", thread.get());
+                    ALOGV("mAfPatchPanelCallback->openOutput_l() returned %p", thread.get());
                     if (thread == 0) {
                         status = NO_MEMORY;
                         goto exit;
                     }
-                    newPatch.mPlayback.setThread(reinterpret_cast<PlaybackThread*>(thread.get()));
+                    newPatch.mPlayback.setThread(thread->asIAfPlaybackThread().get());
                 }
                 audio_devices_t device = patch->sources[0].ext.device.type;
                 String8 address = String8(patch->sources[0].ext.device.address);
@@ -323,7 +301,7 @@
                                 == AUDIO_STREAM_VOICE_CALL) {
                     source = AUDIO_SOURCE_VOICE_COMMUNICATION;
                 }
-                sp<ThreadBase> thread = mAudioFlinger.openInput_l(srcModule,
+                const sp<IAfThreadBase> thread = mAfPatchPanelCallback->openInput_l(srcModule,
                                                                     &input,
                                                                     &config,
                                                                     device,
@@ -332,13 +310,13 @@
                                                                     flags,
                                                                     outputDevice,
                                                                     outputDeviceAddress);
-                ALOGV("mAudioFlinger.openInput_l() returned %p inChannelMask %08x",
+                ALOGV("mAfPatchPanelCallback->openInput_l() returned %p inChannelMask %08x",
                       thread.get(), config.channel_mask);
                 if (thread == 0) {
                     status = NO_MEMORY;
                     goto exit;
                 }
-                newPatch.mRecord.setThread(reinterpret_cast<RecordThread*>(thread.get()));
+                newPatch.mRecord.setThread(thread->asIAfRecordThread().get());
                 status = newPatch.createConnections(this);
                 if (status != NO_ERROR) {
                     goto exit;
@@ -348,10 +326,11 @@
                 }
             } else {
                 if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
-                    sp<ThreadBase> thread = mAudioFlinger.checkRecordThread_l(
+                    sp<IAfThreadBase> thread = mAfPatchPanelCallback->checkRecordThread_l(
                                                               patch->sinks[0].ext.mix.handle);
                     if (thread == 0) {
-                        thread = mAudioFlinger.checkMmapThread_l(patch->sinks[0].ext.mix.handle);
+                        thread = mAfPatchPanelCallback->checkMmapThread_l(
+                                patch->sinks[0].ext.mix.handle);
                         if (thread == 0) {
                             ALOGW("%s() bad capture I/O handle %d",
                                     __func__, patch->sinks[0].ext.mix.handle);
@@ -359,9 +338,9 @@
                             goto exit;
                         }
                     }
-                    mAudioFlinger.unlock();
+                    mAfPatchPanelCallback->unlock();
                     status = thread->sendCreateAudioPatchConfigEvent(patch, &halHandle);
-                    mAudioFlinger.lock();
+                    mAfPatchPanelCallback->lock();
                     if (status == NO_ERROR) {
                         newPatch.setThread(thread);
                     }
@@ -385,7 +364,7 @@
         } break;
         case AUDIO_PORT_TYPE_MIX: {
             audio_module_handle_t srcModule =  patch->sources[0].ext.mix.hw_module;
-            ssize_t index = mAudioFlinger.mAudioHwDevs.indexOfKey(srcModule);
+            ssize_t index = mAfPatchPanelCallback->getAudioHwDevs_l().indexOfKey(srcModule);
             if (index < 0) {
                 ALOGW("%s() bad src hw module %d", __func__, srcModule);
                 status = BAD_VALUE;
@@ -411,10 +390,11 @@
                 device->applyAudioPortConfig(&patch->sinks[i]);
                 devices.push_back(device);
             }
-            sp<ThreadBase> thread =
-                            mAudioFlinger.checkPlaybackThread_l(patch->sources[0].ext.mix.handle);
+            sp<IAfThreadBase> thread = mAfPatchPanelCallback->checkPlaybackThread_l(
+                    patch->sources[0].ext.mix.handle);
             if (thread == 0) {
-                thread = mAudioFlinger.checkMmapThread_l(patch->sources[0].ext.mix.handle);
+                thread = mAfPatchPanelCallback->checkMmapThread_l(
+                        patch->sources[0].ext.mix.handle);
                 if (thread == 0) {
                     ALOGW("%s() bad playback I/O handle %d",
                             __func__, patch->sources[0].ext.mix.handle);
@@ -422,13 +402,13 @@
                     goto exit;
                 }
             }
-            if (thread == mAudioFlinger.primaryPlaybackThread_l()) {
-                mAudioFlinger.updateOutDevicesForRecordThreads_l(devices);
+            if (thread == mAfPatchPanelCallback->primaryPlaybackThread_l()) {
+                mAfPatchPanelCallback->updateOutDevicesForRecordThreads_l(devices);
             }
 
-            mAudioFlinger.unlock();
+            mAfPatchPanelCallback->unlock();
             status = thread->sendCreateAudioPatchConfigEvent(patch, &halHandle);
-            mAudioFlinger.lock();
+            mAfPatchPanelCallback->lock();
             if (status == NO_ERROR) {
                 newPatch.setThread(thread);
             }
@@ -453,9 +433,10 @@
 exit:
     ALOGV("%s() status %d", __func__, status);
     if (status == NO_ERROR) {
-        *handle = (audio_patch_handle_t) mAudioFlinger.nextUniqueId(AUDIO_UNIQUE_ID_USE_PATCH);
+        *handle = static_cast<audio_patch_handle_t>(
+                mAfPatchPanelCallback->nextUniqueId(AUDIO_UNIQUE_ID_USE_PATCH));
         newPatch.mHalHandle = halHandle;
-        mAudioFlinger.mPatchCommandThread->createAudioPatch(*handle, newPatch);
+        mAfPatchPanelCallback->getPatchCommandThread()->createAudioPatch(*handle, newPatch);
         if (insertedModule != AUDIO_MODULE_HANDLE_NONE) {
             addSoftwarePatchToInsertedModules(insertedModule, *handle, &newPatch.mAudioPatch);
         }
@@ -466,13 +447,13 @@
     return status;
 }
 
-AudioFlinger::PatchPanel::Patch::~Patch()
+PatchPanel::Patch::~Patch()
 {
     ALOGE_IF(isSoftware(), "Software patch connections leaked %d %d",
             mRecord.handle(), mPlayback.handle());
 }
 
-status_t AudioFlinger::PatchPanel::Patch::createConnections(PatchPanel *panel)
+status_t PatchPanel::Patch::createConnections(const sp<IAfPatchPanel>& panel)
 {
     // create patch from source device to record thread input
     status_t status = panel->createAudioPatch(
@@ -546,7 +527,7 @@
         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 +539,7 @@
         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 +558,7 @@
         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 +583,7 @@
     // 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,
@@ -636,7 +617,7 @@
     return status;
 }
 
-void AudioFlinger::PatchPanel::Patch::clearConnections(PatchPanel *panel)
+void PatchPanel::Patch::clearConnections(const sp<IAfPatchPanel>& panel)
 {
     ALOGV("%s() mRecord.handle %d mPlayback.handle %d",
             __func__, mRecord.handle(), mPlayback.handle());
@@ -647,7 +628,7 @@
     mPlayback.closeConnections(panel);
 }
 
-status_t AudioFlinger::PatchPanel::Patch::getLatencyMs(double *latencyMs) const
+status_t PatchPanel::Patch::getLatencyMs(double* latencyMs) const
 {
     if (!isSoftware()) return INVALID_OPERATION;
 
@@ -679,7 +660,7 @@
     // If so, do a frame diff and time difference computation to estimate
     // the total patch latency. This requires that frame counts are reported by the
     // HAL are matched properly in the case of record overruns and playback underruns.
-    ThreadBase::TrackBase::FrameTime recordFT{}, playFT{};
+    IAfTrack::FrameTime recordFT{}, playFT{};
     recordTrack->getKernelFrameTime(&recordFT);
     playbackTrack->getKernelFrameTime(&playFT);
     if (recordFT.timeNs > 0 && playFT.timeNs > 0) {
@@ -706,7 +687,7 @@
     return INVALID_OPERATION;
 }
 
-String8 AudioFlinger::PatchPanel::Patch::dump(audio_patch_handle_t myHandle) const
+String8 PatchPanel::Patch::dump(audio_patch_handle_t myHandle) const
 {
     // TODO: Consider table dump form for patches, just like tracks.
     String8 result = String8::format("Patch %d: %s (thread %p => thread %p)",
@@ -734,8 +715,8 @@
 }
 
 /* Disconnect a patch */
-status_t AudioFlinger::PatchPanel::releaseAudioPatch(audio_patch_handle_t handle)
- //unlocks AudioFlinger::mLock when calling ThreadBase::sendReleaseAudioPatchConfigEvent
+status_t PatchPanel::releaseAudioPatch(audio_patch_handle_t handle)
+ //unlocks AudioFlinger::mLock when calling IAfThreadBase::sendReleaseAudioPatchConfigEvent
  //to avoid deadlocks if the thread loop needs to acquire AudioFlinger::mLock
  //before processing the release patch request.
  NO_THREAD_SAFETY_ANALYSIS
@@ -767,18 +748,18 @@
 
             if (patch.sinks[0].type == AUDIO_PORT_TYPE_MIX) {
                 audio_io_handle_t ioHandle = patch.sinks[0].ext.mix.handle;
-                sp<ThreadBase> thread = mAudioFlinger.checkRecordThread_l(ioHandle);
+                sp<IAfThreadBase> thread = mAfPatchPanelCallback->checkRecordThread_l(ioHandle);
                 if (thread == 0) {
-                    thread = mAudioFlinger.checkMmapThread_l(ioHandle);
+                    thread = mAfPatchPanelCallback->checkMmapThread_l(ioHandle);
                     if (thread == 0) {
                         ALOGW("%s() bad capture I/O handle %d", __func__, ioHandle);
                         status = BAD_VALUE;
                         break;
                     }
                 }
-                mAudioFlinger.unlock();
+                mAfPatchPanelCallback->unlock();
                 status = thread->sendReleaseAudioPatchConfigEvent(removedPatch.mHalHandle);
-                mAudioFlinger.lock();
+                mAfPatchPanelCallback->lock();
             } else {
                 status = hwDevice->releaseAudioPatch(removedPatch.mHalHandle);
             }
@@ -790,18 +771,18 @@
                 break;
             }
             audio_io_handle_t ioHandle = src.ext.mix.handle;
-            sp<ThreadBase> thread = mAudioFlinger.checkPlaybackThread_l(ioHandle);
+            sp<IAfThreadBase> thread = mAfPatchPanelCallback->checkPlaybackThread_l(ioHandle);
             if (thread == 0) {
-                thread = mAudioFlinger.checkMmapThread_l(ioHandle);
+                thread = mAfPatchPanelCallback->checkMmapThread_l(ioHandle);
                 if (thread == 0) {
                     ALOGW("%s() bad playback I/O handle %d", __func__, ioHandle);
                     status = BAD_VALUE;
                     break;
                 }
             }
-            mAudioFlinger.unlock();
+            mAfPatchPanelCallback->unlock();
             status = thread->sendReleaseAudioPatchConfigEvent(removedPatch.mHalHandle);
-            mAudioFlinger.lock();
+            mAfPatchPanelCallback->lock();
         } break;
         default:
             status = BAD_VALUE;
@@ -811,23 +792,23 @@
     return status;
 }
 
-void AudioFlinger::PatchPanel::erasePatch(audio_patch_handle_t handle) {
+void PatchPanel::erasePatch(audio_patch_handle_t handle) {
     mPatches.erase(handle);
     removeSoftwarePatchFromInsertedModules(handle);
-    mAudioFlinger.mPatchCommandThread->releaseAudioPatch(handle);
+    mAfPatchPanelCallback->getPatchCommandThread()->releaseAudioPatch(handle);
 }
 
 /* List connected audio ports and they attributes */
-status_t AudioFlinger::PatchPanel::listAudioPatches(unsigned int *num_patches __unused,
+status_t PatchPanel::listAudioPatches(unsigned int* /* num_patches */,
                                   struct audio_patch *patches __unused)
 {
     ALOGV(__func__);
     return NO_ERROR;
 }
 
-status_t AudioFlinger::PatchPanel::getDownstreamSoftwarePatches(
+status_t PatchPanel::getDownstreamSoftwarePatches(
         audio_io_handle_t stream,
-        std::vector<AudioFlinger::PatchPanel::SoftwarePatch> *patches) const
+        std::vector<SoftwarePatch>* patches) const
 {
     for (const auto& module : mInsertedModules) {
         if (module.second.streams.count(stream)) {
@@ -835,7 +816,8 @@
                 const auto& patch_iter = mPatches.find(patchHandle);
                 if (patch_iter != mPatches.end()) {
                     const Patch &patch = patch_iter->second;
-                    patches->emplace_back(*this, patchHandle,
+                    patches->emplace_back(sp<const IAfPatchPanel>::fromExisting(this),
+                            patchHandle,
                             patch.mPlayback.const_thread()->id(),
                             patch.mRecord.const_thread()->id());
                 } else {
@@ -849,7 +831,7 @@
     return BAD_VALUE;
 }
 
-void AudioFlinger::PatchPanel::notifyStreamOpened(
+void PatchPanel::notifyStreamOpened(
         AudioHwDevice *audioHwDevice, audio_io_handle_t stream, struct audio_patch *patch)
 {
     if (audioHwDevice->isInsert()) {
@@ -867,41 +849,41 @@
     }
 }
 
-void AudioFlinger::PatchPanel::notifyStreamClosed(audio_io_handle_t stream)
+void PatchPanel::notifyStreamClosed(audio_io_handle_t stream)
 {
     for (auto& module : mInsertedModules) {
         module.second.streams.erase(stream);
     }
 }
 
-AudioHwDevice* AudioFlinger::PatchPanel::findAudioHwDeviceByModule(audio_module_handle_t module)
+AudioHwDevice* PatchPanel::findAudioHwDeviceByModule(audio_module_handle_t module)
 {
     if (module == AUDIO_MODULE_HANDLE_NONE) return nullptr;
-    ssize_t index = mAudioFlinger.mAudioHwDevs.indexOfKey(module);
+    ssize_t index = mAfPatchPanelCallback->getAudioHwDevs_l().indexOfKey(module);
     if (index < 0) {
         ALOGW("%s() bad hw module %d", __func__, module);
         return nullptr;
     }
-    return mAudioFlinger.mAudioHwDevs.valueAt(index);
+    return mAfPatchPanelCallback->getAudioHwDevs_l().valueAt(index);
 }
 
-sp<DeviceHalInterface> AudioFlinger::PatchPanel::findHwDeviceByModule(audio_module_handle_t module)
+sp<DeviceHalInterface> PatchPanel::findHwDeviceByModule(audio_module_handle_t module)
 {
     AudioHwDevice *audioHwDevice = findAudioHwDeviceByModule(module);
     return audioHwDevice ? audioHwDevice->hwDevice() : nullptr;
 }
 
-void AudioFlinger::PatchPanel::addSoftwarePatchToInsertedModules(
+void PatchPanel::addSoftwarePatchToInsertedModules(
         audio_module_handle_t module, audio_patch_handle_t handle,
         const struct audio_patch *patch)
 {
     mInsertedModules[module].sw_patches.insert(handle);
     if (!mInsertedModules[module].streams.empty()) {
-        mAudioFlinger.updateDownStreamPatches_l(patch, mInsertedModules[module].streams);
+        mAfPatchPanelCallback->updateDownStreamPatches_l(patch, mInsertedModules[module].streams);
     }
 }
 
-void AudioFlinger::PatchPanel::removeSoftwarePatchFromInsertedModules(
+void PatchPanel::removeSoftwarePatchFromInsertedModules(
         audio_patch_handle_t handle)
 {
     for (auto& module : mInsertedModules) {
@@ -909,7 +891,7 @@
     }
 }
 
-void AudioFlinger::PatchPanel::dump(int fd) const
+void PatchPanel::dump(int fd) const
 {
     String8 patchPanelDump;
     const char *indent = "  ";
diff --git a/services/audioflinger/PatchPanel.h b/services/audioflinger/PatchPanel.h
index 2c5e47c..b8b7b79 100644
--- a/services/audioflinger/PatchPanel.h
+++ b/services/audioflinger/PatchPanel.h
@@ -15,217 +15,58 @@
 ** limitations under the License.
 */
 
-#ifndef INCLUDING_FROM_AUDIOFLINGER_H
-    #error This header file should only be included from AudioFlinger.h
-#endif
+#pragma once
 
-public: // TODO(b/288339104) extract out of AudioFlinger class
-// PatchPanel is concealed within AudioFlinger, their lifetimes are the same.
-class PatchPanel {
+#include "IAfPatchPanel.h"
+
+#include <map>  // avoid transitive dependency
+#include <set>  // avoid transitive dependency
+
+namespace android {
+
+class PatchPanel : public IAfPatchPanel {
 public:
-    class SoftwarePatch {
-      public:
-        SoftwarePatch(const PatchPanel &patchPanel, audio_patch_handle_t patchHandle,
-                audio_io_handle_t playbackThreadHandle, audio_io_handle_t recordThreadHandle)
-                : mPatchPanel(patchPanel), mPatchHandle(patchHandle),
-                  mPlaybackThreadHandle(playbackThreadHandle),
-                  mRecordThreadHandle(recordThreadHandle) {}
-        SoftwarePatch(const SoftwarePatch&) = default;
-
-        // Must be called under AudioFlinger::mLock
-        status_t getLatencyMs_l(double *latencyMs) const;
-        audio_patch_handle_t getPatchHandle() const { return mPatchHandle; };
-        audio_io_handle_t getPlaybackThreadHandle() const { return mPlaybackThreadHandle; };
-        audio_io_handle_t getRecordThreadHandle() const { return mRecordThreadHandle; };
-      private:
-        const PatchPanel &mPatchPanel;
-        const audio_patch_handle_t mPatchHandle;
-        const audio_io_handle_t mPlaybackThreadHandle;
-        const audio_io_handle_t mRecordThreadHandle;
-    };
-
-    explicit PatchPanel(AudioFlinger* audioFlinger) : mAudioFlinger(*audioFlinger) {}
+    explicit PatchPanel(const sp<IAfPatchPanelCallback>& afPatchPanelCallback)
+        : mAfPatchPanelCallback(afPatchPanelCallback) {}
 
     /* List connected audio ports and their attributes */
     status_t listAudioPorts(unsigned int *num_ports,
-                                    struct audio_port *ports);
+        struct audio_port* ports) final;
 
     /* Get supported attributes for a given audio port */
-    status_t getAudioPort(struct audio_port_v7 *port);
+    status_t getAudioPort(struct audio_port_v7* port) final;
 
     /* Create a patch between several source and sink ports */
     status_t createAudioPatch(const struct audio_patch *patch,
                               audio_patch_handle_t *handle,
-                              bool endpointPatch = false);
+                              bool endpointPatch = false) final;
 
     /* Release a patch */
-    status_t releaseAudioPatch(audio_patch_handle_t handle);
+    status_t releaseAudioPatch(audio_patch_handle_t handle) final;
 
     /* List connected audio devices and they attributes */
     status_t listAudioPatches(unsigned int *num_patches,
-                                      struct audio_patch *patches);
+            struct audio_patch* patches) final;
 
     // Retrieves all currently estrablished software patches for a stream
     // opened on an intermediate module.
     status_t getDownstreamSoftwarePatches(audio_io_handle_t stream,
-            std::vector<SoftwarePatch> *patches) const;
+            std::vector<SoftwarePatch>* patches) const final;
 
     // Notifies patch panel about all opened and closed streams.
     void notifyStreamOpened(AudioHwDevice *audioHwDevice, audio_io_handle_t stream,
-                            struct audio_patch *patch);
-    void notifyStreamClosed(audio_io_handle_t stream);
+                            struct audio_patch* patch) final;
+    void notifyStreamClosed(audio_io_handle_t stream) final;
 
-    void dump(int fd) const;
-
-    template<typename ThreadType, typename TrackType>
-    class Endpoint final {
-    public:
-        Endpoint() = default;
-        Endpoint(const Endpoint&) = delete;
-        Endpoint& operator=(const Endpoint& other) noexcept {
-            mThread = other.mThread;
-            mCloseThread = other.mCloseThread;
-            mHandle = other.mHandle;
-            mTrack = other.mTrack;
-            return *this;
-        }
-        Endpoint(Endpoint&& other) noexcept { swap(other); }
-        Endpoint& operator=(Endpoint&& other) noexcept {
-            swap(other);
-            return *this;
-        }
-        ~Endpoint() {
-            ALOGE_IF(mHandle != AUDIO_PATCH_HANDLE_NONE,
-                    "A non empty Patch Endpoint leaked, handle %d", mHandle);
-        }
-
-        status_t checkTrack(TrackType *trackOrNull) const {
-            if (trackOrNull == nullptr) return NO_MEMORY;
-            return trackOrNull->initCheck();
-        }
-        audio_patch_handle_t handle() const { return mHandle; }
-        sp<ThreadType> thread() const { return mThread; }
-        sp<TrackType> track() const { return mTrack; }
-        sp<const ThreadType> const_thread() const { return mThread; }
-        sp<const TrackType> const_track() const { return mTrack; }
-
-        void closeConnections(PatchPanel *panel) {
-            if (mHandle != AUDIO_PATCH_HANDLE_NONE) {
-                panel->releaseAudioPatch(mHandle);
-                mHandle = AUDIO_PATCH_HANDLE_NONE;
-            }
-            if (mThread != 0) {
-                if (mTrack != 0) {
-                    mThread->deletePatchTrack(mTrack);
-                }
-                if (mCloseThread) {
-                    panel->mAudioFlinger.closeThreadInternal_l(mThread);
-                }
-            }
-        }
-        audio_patch_handle_t* handlePtr() { return &mHandle; }
-        void setThread(const sp<ThreadType>& thread, bool closeThread = true) {
-            mThread = thread;
-            mCloseThread = closeThread;
-        }
-        template <typename T>
-        void setTrackAndPeer(const sp<TrackType>& track, const sp<T> &peer, bool holdReference) {
-            mTrack = track;
-            mThread->addPatchTrack(mTrack);
-            mTrack->setPeerProxy(peer, holdReference);
-            mClearPeerProxy = holdReference;
-        }
-        void clearTrackPeer() { if (mClearPeerProxy && mTrack) mTrack->clearPeerProxy(); }
-        void stopTrack() { if (mTrack) mTrack->stop(); }
-
-        void swap(Endpoint &other) noexcept {
-            using std::swap;
-            swap(mThread, other.mThread);
-            swap(mCloseThread, other.mCloseThread);
-            swap(mClearPeerProxy, other.mClearPeerProxy);
-            swap(mHandle, other.mHandle);
-            swap(mTrack, other.mTrack);
-        }
-
-        friend void swap(Endpoint &a, Endpoint &b) noexcept {
-            a.swap(b);
-        }
-
-    private:
-        sp<ThreadType> mThread;
-        bool mCloseThread = true;
-        bool mClearPeerProxy = true;
-        audio_patch_handle_t mHandle = AUDIO_PATCH_HANDLE_NONE;
-        sp<TrackType> mTrack;
-    };
-
-    class Patch final {
-    public:
-        Patch(const struct audio_patch &patch, bool endpointPatch) :
-            mAudioPatch(patch), mIsEndpointPatch(endpointPatch) {}
-        Patch() = default;
-        ~Patch();
-        Patch(const Patch& other) noexcept {
-            mAudioPatch = other.mAudioPatch;
-            mHalHandle = other.mHalHandle;
-            mPlayback = other.mPlayback;
-            mRecord = other.mRecord;
-            mThread = other.mThread;
-            mIsEndpointPatch = other.mIsEndpointPatch;
-        }
-        Patch(Patch&& other) noexcept { swap(other); }
-        Patch& operator=(Patch&& other) noexcept {
-            swap(other);
-            return *this;
-        }
-
-        void swap(Patch &other) noexcept {
-            using std::swap;
-            swap(mAudioPatch, other.mAudioPatch);
-            swap(mHalHandle, other.mHalHandle);
-            swap(mPlayback, other.mPlayback);
-            swap(mRecord, other.mRecord);
-            swap(mThread, other.mThread);
-            swap(mIsEndpointPatch, other.mIsEndpointPatch);
-        }
-
-        friend void swap(Patch &a, Patch &b) noexcept {
-            a.swap(b);
-        }
-
-        status_t createConnections(PatchPanel *panel);
-        void clearConnections(PatchPanel *panel);
-        bool isSoftware() const {
-            return mRecord.handle() != AUDIO_PATCH_HANDLE_NONE ||
-                    mPlayback.handle() != AUDIO_PATCH_HANDLE_NONE; }
-
-        void setThread(const sp<ThreadBase>& thread) { mThread = thread; }
-        wp<ThreadBase> thread() const { return mThread; }
-
-        // returns the latency of the patch (from record to playback).
-        status_t getLatencyMs(double *latencyMs) const;
-
-        String8 dump(audio_patch_handle_t myHandle) const;
-
-        // Note that audio_patch::id is only unique within a HAL module
-        struct audio_patch              mAudioPatch;
-        // handle for audio HAL patch handle present only when the audio HAL version is >= 3.0
-        audio_patch_handle_t            mHalHandle = AUDIO_PATCH_HANDLE_NONE;
-        // below members are used by a software audio patch connecting a source device from a
-        // given audio HW module to a sink device on an other audio HW module.
-        // 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;
-        // connects source device to record thread input
-        Endpoint<RecordThread, RecordThread::PatchRecord> mRecord;
-
-        wp<ThreadBase> mThread;
-        bool mIsEndpointPatch;
-    };
+    void dump(int fd) const final;
 
     // Call with AudioFlinger mLock held
-    std::map<audio_patch_handle_t, Patch>& patches_l() { return mPatches; }
+    const std::map<audio_patch_handle_t, Patch>& patches_l() const final { return mPatches; }
+
+    // Must be called under AudioFlinger::mLock
+    status_t getLatencyMs_l(audio_patch_handle_t patchHandle, double* latencyMs) const final;
+
+    void closeThreadInternal_l(const sp<IAfThreadBase>& thread) const final;
 
 private:
     AudioHwDevice* findAudioHwDeviceByModule(audio_module_handle_t module);
@@ -236,7 +77,7 @@
     void removeSoftwarePatchFromInsertedModules(audio_patch_handle_t handle);
     void erasePatch(audio_patch_handle_t handle);
 
-    AudioFlinger &mAudioFlinger;
+    const sp<IAfPatchPanelCallback> mAfPatchPanelCallback;
     std::map<audio_patch_handle_t, Patch> mPatches;
 
     // This map allows going from a thread to "downstream" software patches
@@ -266,4 +107,4 @@
     std::map<audio_module_handle_t, ModuleConnections> mInsertedModules;
 };
 
-private:
+}  // namespace android
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 7ff2394..5f54e11 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -15,12 +15,15 @@
 ** 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>
+#include "TrackBase.h"
+
+#include <android/os/BnExternalVibrationController.h>
+#include <audio_utils/LinearMap.h>
+#include <binder/AppOpsManager.h>
+
+namespace android {
 
 // Checks and monitors OP_PLAY_AUDIO
 class OpPlayAudioMonitor : public RefBase {
@@ -30,13 +33,13 @@
     bool hasOpPlayAudio() const;
 
     static sp<OpPlayAudioMonitor> createIfNeeded(
-            AudioFlinger::ThreadBase* thread,
+            IAfThreadBase* thread,
             const AttributionSourceState& attributionSource,
             const audio_attributes_t& attr, int id,
             audio_stream_type_t streamType);
 
 private:
-    OpPlayAudioMonitor(AudioFlinger::ThreadBase* thread,
+    OpPlayAudioMonitor(IAfThreadBase* thread,
                        const AttributionSourceState& attributionSource,
                        audio_usage_t usage, int id, uid_t uid);
     void onFirstRef() override;
@@ -57,7 +60,7 @@
     // called by PlayAudioOpCallback when OP_PLAY_AUDIO is updated in AppOp callback
     void checkPlayAudioForUsage(bool doBroadcast);
 
-    wp<AudioFlinger::ThreadBase> mThread;
+    wp<IAfThreadBase> mThread;
     std::atomic_bool mHasOpPlayAudio;
     const AttributionSourceState mAttributionSource;
     const int32_t mUsage; // on purpose not audio_usage_t because always checked in appOps as int32_t
@@ -67,16 +70,9 @@
 };
 
 // playback track
-class Track : public TrackBase, public VolumeProvider {
+class Track : public TrackBase, public virtual IAfTrack, public VolumeProvider {
 public:
-    // createIAudioTrackAdapter() is a static constructor which creates an
-    // IAudioTrack AIDL interface adapter from the Track object that
-    // may be passed back to the client (if needed).
-    //
-    // Only one AIDL IAudioTrack interface wrapper should be created per Track.
-    static sp<media::IAudioTrack> createIAudioTrackAdapter(const sp<Track>& track);
-
-                        Track(  PlaybackThread *thread,
+    Track(IAfPlaybackThread* thread,
                                 const sp<Client>& client,
                                 audio_stream_type_t streamType,
                                 const audio_attributes_t& attr,
@@ -99,72 +95,66 @@
                                 float speed = 1.0f,
                                 bool isSpatialized = false,
                                 bool isBitPerfect = false);
-    virtual             ~Track();
-    virtual status_t    initCheck() const;
-
-            void        appendDumpHeader(String8& result);
-            void        appendDump(String8& result, bool active);
-    virtual status_t    start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
-                              audio_session_t triggerSession = AUDIO_SESSION_NONE);
-    virtual void        stop();
-            void        pause();
-
-            void        flush();
-            void        destroy();
-
-    virtual uint32_t    sampleRate() const;
-
-            audio_stream_type_t streamType() const {
+    ~Track() override;
+    status_t initCheck() const final;
+    void appendDumpHeader(String8& result) const final;
+    void appendDump(String8& result, bool active) const final;
+    status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
+            audio_session_t triggerSession = AUDIO_SESSION_NONE) override;
+    void stop() override;
+    void pause() final;
+    void flush() final;
+    void destroy() final;
+    uint32_t sampleRate() const final;
+    audio_stream_type_t streamType() const final {
                 return mStreamType;
             }
-            bool        isOffloaded() const
+    bool isOffloaded() const final
                                 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }
-            bool        isDirect() const override
+    bool isDirect() const final
                                 { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; }
-            bool        isOffloadedOrDirect() const { return (mFlags
+    bool isOffloadedOrDirect() const final { return (mFlags
                             & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD
                                     | AUDIO_OUTPUT_FLAG_DIRECT)) != 0; }
-            bool        isStatic() const { return  mSharedBuffer.get() != nullptr; }
+    bool isStatic() const final { return  mSharedBuffer.get() != nullptr; }
 
-            status_t    setParameters(const String8& keyValuePairs);
-            status_t    selectPresentation(int presentationId, int programId);
-            status_t    attachAuxEffect(int EffectId);
-            void        setAuxBuffer(int EffectId, int32_t *buffer);
-            int32_t     *auxBuffer() const { return mAuxBuffer; }
-            void        setMainBuffer(float *buffer) { mMainBuffer = buffer; }
-            float       *mainBuffer() const { return mMainBuffer; }
-            int         auxEffectId() const { return mAuxEffectId; }
-    virtual status_t    getTimestamp(AudioTimestamp& timestamp);
-            void        signal();
-            status_t    getDualMonoMode(audio_dual_mono_mode_t* mode);
-            status_t    setDualMonoMode(audio_dual_mono_mode_t mode);
-            status_t    getAudioDescriptionMixLevel(float* leveldB);
-            status_t    setAudioDescriptionMixLevel(float leveldB);
-            status_t    getPlaybackRateParameters(audio_playback_rate_t* playbackRate);
-            status_t    setPlaybackRateParameters(const audio_playback_rate_t& playbackRate);
+    status_t setParameters(const String8& keyValuePairs) final;
+    status_t selectPresentation(int presentationId, int programId) final;
+    status_t attachAuxEffect(int EffectId) final;
+    void setAuxBuffer(int EffectId, int32_t* buffer) final;
+    int32_t* auxBuffer() const final { return mAuxBuffer; }
+    void setMainBuffer(float* buffer) final { mMainBuffer = buffer; }
+    float* mainBuffer() const final { return mMainBuffer; }
+    int auxEffectId() const final { return mAuxEffectId; }
+    status_t getTimestamp(AudioTimestamp& timestamp) final;
+    void signal() final;
+    status_t getDualMonoMode(audio_dual_mono_mode_t* mode) const final;
+    status_t setDualMonoMode(audio_dual_mono_mode_t mode) final;
+    status_t getAudioDescriptionMixLevel(float* leveldB) const final;
+    status_t setAudioDescriptionMixLevel(float leveldB) final;
+    status_t getPlaybackRateParameters(audio_playback_rate_t* playbackRate) const final;
+    status_t setPlaybackRateParameters(const audio_playback_rate_t& playbackRate) final;
 
-// implement FastMixerState::VolumeProvider interface
-    virtual gain_minifloat_packed_t getVolumeLR();
+    // implement FastMixerState::VolumeProvider interface
+    gain_minifloat_packed_t getVolumeLR() const final;
 
-            status_t    setSyncEvent(const sp<audioflinger::SyncEvent>& event) override;
-
-    virtual bool        isFastTrack() const { return (mFlags & AUDIO_OUTPUT_FLAG_FAST) != 0; }
-
-            double      bufferLatencyMs() const override {
+    status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) final;
+    bool isFastTrack() const final { return (mFlags & AUDIO_OUTPUT_FLAG_FAST) != 0; }
+    double bufferLatencyMs() const final {
                             return isStatic() ? 0. : TrackBase::bufferLatencyMs();
                         }
 
-// implement volume handling.
+    // implement volume handling.
     media::VolumeShaper::Status applyVolumeShaper(
                                 const sp<media::VolumeShaper::Configuration>& configuration,
                                 const sp<media::VolumeShaper::Operation>& operation);
-    sp<media::VolumeShaper::State> getVolumeShaperState(int id);
-    sp<media::VolumeHandler>   getVolumeHandler() { return mVolumeHandler; }
+    sp<media::VolumeShaper::State> getVolumeShaperState(int id) const final;
+    sp<media::VolumeHandler> getVolumeHandler() const final{ return mVolumeHandler; }
     /** Set the computed normalized final volume of the track.
      * !masterMute * masterVolume * streamVolume * averageLRVolume */
-    void                setFinalVolume(float volumeLeft, float volumeRight);
-    float               getFinalVolume() const { return mFinalVolume; }
-    void                getFinalVolume(float* left, float* right) const {
+    void setFinalVolume(float volumeLeft, float volumeRight) final;
+    float getFinalVolume() const final { return mFinalVolume; }
+    void getFinalVolume(float* left, float* right) const final {
                             *left = mFinalVolumeLeft;
                             *right = mFinalVolumeRight;
     }
@@ -172,21 +162,22 @@
     using SourceMetadatas = std::vector<playback_track_metadata_v7_t>;
     using MetadataInserter = std::back_insert_iterator<SourceMetadatas>;
     /** Copy the track metadata in the provided iterator. Thread safe. */
-    virtual void    copyMetadataTo(MetadataInserter& backInserter) const;
+    void copyMetadataTo(MetadataInserter& backInserter) const override;
+
 
             /** Return haptic playback of the track is enabled or not, used in mixer. */
-            bool    getHapticPlaybackEnabled() const { return mHapticPlaybackEnabled; }
+    bool getHapticPlaybackEnabled() const final { return mHapticPlaybackEnabled; }
             /** Set haptic playback of the track is enabled or not, should be
              *  set after query or get callback from vibrator service */
-            void    setHapticPlaybackEnabled(bool hapticPlaybackEnabled) {
+    void setHapticPlaybackEnabled(bool hapticPlaybackEnabled) final {
                 mHapticPlaybackEnabled = hapticPlaybackEnabled;
             }
             /** Return at what intensity to play haptics, used in mixer. */
-            os::HapticScale getHapticIntensity() const { return mHapticIntensity; }
+    os::HapticScale getHapticIntensity() const final { return mHapticIntensity; }
             /** Return the maximum amplitude allowed for haptics data, used in mixer. */
-            float getHapticMaxAmplitude() const { return mHapticMaxAmplitude; }
+    float getHapticMaxAmplitude() const final { return mHapticMaxAmplitude; }
             /** Set intensity of haptic playback, should be set after querying vibrator service. */
-            void    setHapticIntensity(os::HapticScale hapticIntensity) {
+    void setHapticIntensity(os::HapticScale hapticIntensity) final {
                 if (os::isValidHapticScale(hapticIntensity)) {
                     mHapticIntensity = hapticIntensity;
                     setHapticPlaybackEnabled(mHapticIntensity != os::HapticScale::MUTE);
@@ -195,16 +186,16 @@
             /** Set maximum amplitude allowed for haptic data, should be set after querying
              *  vibrator service.
              */
-            void    setHapticMaxAmplitude(float maxAmplitude) {
+    void setHapticMaxAmplitude(float maxAmplitude) final {
                 mHapticMaxAmplitude = maxAmplitude;
             }
-            sp<os::ExternalVibration> getExternalVibration() const { return mExternalVibration; }
+    sp<os::ExternalVibration> getExternalVibration() const final { return mExternalVibration; }
 
             // This function should be called with holding thread lock.
-            void    updateTeePatches();
-            void    setTeePatchesToUpdate(TeePatches teePatchesToUpdate);
+    void updateTeePatches_l() final;
+    void setTeePatchesToUpdate_l(TeePatches teePatchesToUpdate) final;
 
-    void tallyUnderrunFrames(size_t frames) override {
+    void tallyUnderrunFrames(size_t frames) final {
        if (isOut()) { // we expect this from output tracks only
            mAudioTrackServerProxy->tallyUnderrunFrames(frames);
            // Fetch absolute numbers from AudioTrackShared as it counts
@@ -215,29 +206,18 @@
        }
     }
 
-    static bool checkServerLatencySupported(
-            audio_format_t format, audio_output_flags_t flags) {
-        return audio_is_linear_pcm(format)
-                && (flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) == 0;
-    }
-
-    audio_output_flags_t getOutputFlags() const { return mFlags; }
-    float getSpeed() const { return mSpeed; }
-    bool isSpatialized() const override { return mIsSpatialized; }
-    bool isBitPerfect() const override { return mIsBitPerfect; }
+    audio_output_flags_t getOutputFlags() const final { return mFlags; }
+    float getSpeed() const final { return mSpeed; }
+    bool isSpatialized() const final { return mIsSpatialized; }
+    bool isBitPerfect() const final { return mIsBitPerfect; }
 
     /**
      * Updates the mute state and notifies the audio service. Call this only when holding player
      * thread lock.
      */
-    void processMuteEvent_l(const sp<IAudioManager>& audioManager, mute_state_t muteState);
+    void processMuteEvent_l(const sp<IAudioManager>& audioManager, mute_state_t muteState) final;
 
 protected:
-    // for numerous
-    friend class PlaybackThread;
-    friend class MixerThread;
-    friend class DirectOutputThread;
-    friend class OffloadThread;
 
     DISALLOW_COPY_AND_ASSIGN(Track);
 
@@ -246,38 +226,39 @@
     void releaseBuffer(AudioBufferProvider::Buffer* buffer) override;
 
     // ExtendedAudioBufferProvider interface
-    virtual size_t framesReady() const;
-    virtual int64_t framesReleased() const;
-    virtual void onTimestamp(const ExtendedTimestamp &timestamp);
+    size_t framesReady() const override;
+    int64_t framesReleased() const override;
+    void onTimestamp(const ExtendedTimestamp &timestamp) override;
 
-    bool isPausing() const { return mState == PAUSING; }
-    bool isPaused() const { return mState == PAUSED; }
-    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;
     }
@@ -288,25 +269,43 @@
 
     void signalClientFlag(int32_t flag);
 
-public:
-    void triggerEvents(AudioSystem::sync_event_t type);
-    virtual void invalidate();
-    void disable();
-
-    int fastIndex() const { return mFastIndex; }
-
-    bool isPlaybackRestricted() const {
+    void triggerEvents(AudioSystem::sync_event_t type) final;
+    void invalidate() final;
+    void disable() final;
+    int& fastIndex() final { return mFastIndex; }
+    bool isPlaybackRestricted() const final {
         // The monitor is only created for tracks that can be silenced.
         return mOpPlayAudioMonitor ? !mOpPlayAudioMonitor->hasOpPlayAudio() : false; }
 
-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
+    // see comment at ~Track for why this can't be const
     sp<IMemory>         mSharedBuffer;
 
     bool                mResetDone;
@@ -349,8 +348,9 @@
 
 private:
     void                interceptBuffer(const AudioBufferProvider::Buffer& buffer);
+    // Must hold thread lock to access tee patches
     template <class F>
-    void                forEachTeePatchTrack(F f) {
+    void                forEachTeePatchTrack_l(F f) {
         for (auto& tp : mTeePatches) { f(tp.patchTrack); }
     };
 
@@ -384,7 +384,7 @@
     bool                mFlushHwPending; // track requests for thread flush
     bool                mPauseHwPending = false; // direct/offload track request for thread pause
     audio_output_flags_t mFlags;
-    TeePatches  mTeePatches;
+    TeePatches mTeePatches;
     std::optional<TeePatches> mTeePatchesToUpdate;
     const float         mSpeed;
     const bool          mIsSpatialized;
@@ -398,7 +398,7 @@
 
 
 // playback track, used by DuplicatingThread
-class OutputTrack : public Track {
+class OutputTrack : public Track, public IAfOutputTrack {
 public:
 
     class Buffer : public AudioBufferProvider::Buffer {
@@ -406,29 +406,28 @@
         void *mBuffer;
     };
 
-                        OutputTrack(PlaybackThread *thread,
-                                DuplicatingThread *sourceThread,
+    OutputTrack(IAfPlaybackThread* thread,
+            IAfDuplicatingThread* sourceThread,
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
                                 size_t frameCount,
                                 const AttributionSourceState& attributionSource);
-    virtual             ~OutputTrack();
+    ~OutputTrack() override;
 
-    virtual status_t    start(AudioSystem::sync_event_t event =
+    status_t start(AudioSystem::sync_event_t event =
                                     AudioSystem::SYNC_EVENT_NONE,
-                             audio_session_t triggerSession = AUDIO_SESSION_NONE);
-    virtual void        stop();
-            ssize_t     write(void* data, uint32_t frames);
-            bool        bufferQueueEmpty() const { return mBufferQueue.size() == 0; }
-            bool        isActive() const { return mActive; }
-    const wp<ThreadBase>& thread() const { return mThread; }
+                             audio_session_t triggerSession = AUDIO_SESSION_NONE) final;
+    void stop() final;
+    ssize_t write(void* data, uint32_t frames) final;
+    bool bufferQueueEmpty() const final { return mBufferQueue.size() == 0; }
+    bool isActive() const final { return mActive; }
 
-            void        copyMetadataTo(MetadataInserter& backInserter) const override;
+    void copyMetadataTo(MetadataInserter& backInserter) const final;
     /** Set the metadatas of the upstream tracks. Thread safe. */
-            void        setMetadatas(const SourceMetadatas& metadatas);
+    void setMetadatas(const SourceMetadatas& metadatas) final;
     /** returns client timestamp to the upstream duplicating thread. */
-    ExtendedTimestamp   getClientProxyTimestamp() const {
+    ExtendedTimestamp getClientProxyTimestamp() const final {
                             // server - kernel difference is not true latency when drained
                             // i.e. mServerProxy->isDrained().
                             ExtendedTimestamp timestamp;
@@ -439,7 +438,6 @@
                             // (with mTimeNs[] filled with -1's) is returned.
                             return timestamp;
                         }
-
 private:
     status_t            obtainBuffer(AudioBufferProvider::Buffer* buffer,
                                      uint32_t waitTimeMs);
@@ -454,7 +452,7 @@
     Vector < Buffer* >          mBufferQueue;
     AudioBufferProvider::Buffer mOutBuffer;
     bool                        mActive;
-    DuplicatingThread* const    mSourceThread; // for waitTimeMs() in write()
+    IAfDuplicatingThread* const mSourceThread; // for waitTimeMs() in write()
     sp<AudioTrackClientProxy>   mClientProxy;
 
     /** Attributes of the source tracks.
@@ -474,10 +472,9 @@
 };  // end of OutputTrack
 
 // playback track, used by PatchPanel
-class PatchTrack : public Track, public PatchTrackBase {
+class PatchTrack : public Track, public PatchTrackBase, public IAfPatchTrack {
 public:
-
-                        PatchTrack(PlaybackThread *playbackThread,
+    PatchTrack(IAfPlaybackThread* playbackThread,
                                    audio_stream_type_t streamType,
                                    uint32_t sampleRate,
                                    audio_channel_mask_t channelMask,
@@ -491,23 +488,24 @@
                                                                     *  as soon as possible to have
                                                                     *  the lowest possible latency
                                                                     *  even if it might glitch. */);
-    virtual             ~PatchTrack();
+    ~PatchTrack() override;
 
-            size_t      framesReady() const override;
+    size_t framesReady() const final;
 
-    virtual status_t    start(AudioSystem::sync_event_t event =
+    status_t start(AudioSystem::sync_event_t event =
                                     AudioSystem::SYNC_EVENT_NONE,
-                             audio_session_t triggerSession = AUDIO_SESSION_NONE);
+                             audio_session_t triggerSession = AUDIO_SESSION_NONE) final;
 
     // AudioBufferProvider interface
-    virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
-    virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
+    status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) final;
+    void releaseBuffer(AudioBufferProvider::Buffer* buffer) final;
 
     // PatchProxyBufferProvider interface
-    virtual status_t    obtainBuffer(Proxy::Buffer* buffer,
-                                     const struct timespec *timeOut = NULL);
-    virtual void        releaseBuffer(Proxy::Buffer* buffer);
+    status_t obtainBuffer(Proxy::Buffer* buffer, const struct timespec* timeOut = nullptr) final;
+    void releaseBuffer(Proxy::Buffer* buffer) final;
 
 private:
             void restartIfDisabled();
 };  // end of PatchTrack
+
+} // namespace android
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index 61f36a1..021add4 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -15,23 +15,19 @@
 ** limitations under the License.
 */
 
-#include <android/content/AttributionSourceState.h>
+#pragma once
 
-#ifndef INCLUDING_FROM_AUDIOFLINGER_H
-    #error This header file should only be included from AudioFlinger.h
-#endif
+#include "TrackBase.h"
+
+#include <android/content/AttributionSourceState.h>
+#include <datapath/AudioStreamIn.h> // struct Source
+
+namespace android {
 
 // record track
-class RecordTrack : public TrackBase {
+class RecordTrack : public TrackBase, public virtual IAfRecordTrack {
 public:
-    // createIAudioRecordAdapter() is a static constructor which creates an
-    // IAudioRecord AIDL interface wrapper from the RecordTrack object that
-    // may be passed back to the client (if needed).
-    //
-    // Only one AIDL IAudioRecord interface wrapper should be created per RecordTrack.
-    static sp<media::IAudioRecord> createIAudioRecordAdapter(const sp<RecordTrack>& recordTrack);
-
-                        RecordTrack(RecordThread *thread,
+    RecordTrack(IAfRecordThread* thread,
                                 const sp<Client>& client,
                                 const audio_attributes_t& attr,
                                 uint32_t sampleRate,
@@ -47,67 +43,70 @@
                                 track_type type,
                                 audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE,
                                 int32_t startFrames = -1);
-    virtual             ~RecordTrack();
-    virtual status_t    initCheck() const;
+    ~RecordTrack() override;
+    status_t initCheck() const final;
 
-    virtual status_t    start(AudioSystem::sync_event_t event, audio_session_t triggerSession);
-    virtual void        stop();
+    status_t start(AudioSystem::sync_event_t event, audio_session_t triggerSession) final;
+    void stop() final;
+    void destroy() final;
+    void invalidate() final;
 
-            void        destroy();
-
-    virtual void        invalidate();
             // clear the buffer overflow flag
-            void        clearOverflow() { mOverflow = false; }
+    void clearOverflow() final { mOverflow = false; }
             // set the buffer overflow flag and return previous value
-            bool        setOverflow() { bool tmp = mOverflow; mOverflow = true;
+    bool setOverflow() final { bool tmp = mOverflow; mOverflow = true;
                                                 return tmp; }
 
-            void        appendDumpHeader(String8& result);
-            void        appendDump(String8& result, bool active);
+    void appendDumpHeader(String8& result) const final;
+    void appendDump(String8& result, bool active) const final;
 
-            void        handleSyncStartEvent(const sp<audioflinger::SyncEvent>& event);
-            void        clearSyncStartEvent();
+    void handleSyncStartEvent(const sp<audioflinger::SyncEvent>& event) final;
+    void clearSyncStartEvent() final;
 
-            void        updateTrackFrameInfo(int64_t trackFramesReleased,
+    void updateTrackFrameInfo(int64_t trackFramesReleased,
                                              int64_t sourceFramesRead,
                                              uint32_t halSampleRate,
-                                             const ExtendedTimestamp &timestamp);
+                                             const ExtendedTimestamp &timestamp) final;
 
-    virtual bool        isFastTrack() const { return (mFlags & AUDIO_INPUT_FLAG_FAST) != 0; }
-            bool        isDirect() const override
+    bool isFastTrack() const final { return (mFlags & AUDIO_INPUT_FLAG_FAST) != 0; }
+    bool isDirect() const final
                                 { return (mFlags & AUDIO_INPUT_FLAG_DIRECT) != 0; }
 
-            void        setSilenced(bool silenced) { if (!isPatchTrack()) mSilenced = silenced; }
-            bool        isSilenced() const { return mSilenced; }
+    void setSilenced(bool silenced) final { if (!isPatchTrack()) mSilenced = silenced; }
+    bool isSilenced() const final { return mSilenced; }
 
-            status_t    getActiveMicrophones(
-                    std::vector<media::MicrophoneInfoFw>* activeMicrophones);
+    status_t getActiveMicrophones(
+            std::vector<media::MicrophoneInfoFw>* activeMicrophones) const final;
 
-            status_t    setPreferredMicrophoneDirection(audio_microphone_direction_t direction);
-            status_t    setPreferredMicrophoneFieldDimension(float zoom);
-            status_t    shareAudioHistory(const std::string& sharedAudioPackageName,
-                                          int64_t sharedAudioStartMs);
-            int32_t     startFrames() { return mStartFrames; }
+    status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction) final;
+    status_t setPreferredMicrophoneFieldDimension(float zoom) final;
+    status_t shareAudioHistory(const std::string& sharedAudioPackageName,
+            int64_t sharedAudioStartMs) final;
+    int32_t startFrames() const final { return mStartFrames; }
 
-    static  bool        checkServerLatencySupported(
-                                audio_format_t format, audio_input_flags_t flags) {
-                            return audio_is_linear_pcm(format)
-                                    && (flags & AUDIO_INPUT_FLAG_HW_AV_SYNC) == 0;
-                        }
+    using SinkMetadatas = std::vector<record_track_metadata_v7_t>;
+    using MetadataInserter = std::back_insert_iterator<SinkMetadatas>;
+    void copyMetadataTo(MetadataInserter& backInserter) const final;
 
-            using SinkMetadatas = std::vector<record_track_metadata_v7_t>;
-            using MetadataInserter = std::back_insert_iterator<SinkMetadatas>;
-            virtual void    copyMetadataTo(MetadataInserter& backInserter) const;
+    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
@@ -120,7 +119,7 @@
                     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;
@@ -133,10 +132,9 @@
 };
 
 // playback track, used by PatchPanel
-class PatchRecord : public RecordTrack, public PatchTrackBase {
+class PatchRecord : public RecordTrack, public PatchTrackBase, public IAfPatchRecord {
 public:
-
-    PatchRecord(RecordThread *recordThread,
+    PatchRecord(IAfRecordThread* recordThread,
                 uint32_t sampleRate,
                 audio_channel_mask_t channelMask,
                 audio_format_t format,
@@ -146,20 +144,20 @@
                 audio_input_flags_t flags,
                 const Timeout& timeout = {},
                 audio_source_t source = AUDIO_SOURCE_DEFAULT);
-    virtual             ~PatchRecord();
+    ~PatchRecord() override;
 
-    virtual Source* getSource() { return nullptr; }
+    Source* getSource() override { return nullptr; }
 
     // AudioBufferProvider interface
-    virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
-    virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
+    status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) override;
+    void releaseBuffer(AudioBufferProvider::Buffer* buffer) override;
 
     // PatchProxyBufferProvider interface
-    virtual status_t    obtainBuffer(Proxy::Buffer *buffer,
-                                     const struct timespec *timeOut = NULL);
-    virtual void        releaseBuffer(Proxy::Buffer *buffer);
+    status_t obtainBuffer(Proxy::Buffer* buffer,
+                                     const struct timespec* timeOut = nullptr) override;
+    void releaseBuffer(Proxy::Buffer* buffer) override;
 
-    size_t writeFrames(const void* src, size_t frameCount, size_t frameSize) {
+    size_t writeFrames(const void* src, size_t frameCount, size_t frameSize) final {
         return writeFrames(this, src, frameCount, frameSize);
     }
 
@@ -172,7 +170,7 @@
 
 class PassthruPatchRecord : public PatchRecord, public Source {
 public:
-    PassthruPatchRecord(RecordThread *recordThread,
+    PassthruPatchRecord(IAfRecordThread* recordThread,
                         uint32_t sampleRate,
                         audio_channel_mask_t channelMask,
                         audio_format_t format,
@@ -180,25 +178,25 @@
                         audio_input_flags_t flags,
                         audio_source_t source = AUDIO_SOURCE_DEFAULT);
 
-    Source* getSource() override { return static_cast<Source*>(this); }
+    Source* getSource() final { return static_cast<Source*>(this); }
 
     // Source interface
-    status_t read(void *buffer, size_t bytes, size_t *read) override;
-    status_t getCapturePosition(int64_t *frames, int64_t *time) override;
-    status_t standby() override;
+    status_t read(void* buffer, size_t bytes, size_t* read) final;
+    status_t getCapturePosition(int64_t* frames, int64_t* time) final;
+    status_t standby() final;
 
     // AudioBufferProvider interface
     // This interface is used by RecordThread to pass the data obtained
     // from HAL or other source to the client. PassthruPatchRecord receives
     // the data in 'obtainBuffer' so these calls are stubbed out.
-    status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) override;
-    void releaseBuffer(AudioBufferProvider::Buffer* buffer) override;
+    status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) final;
+    void releaseBuffer(AudioBufferProvider::Buffer* buffer) final;
 
     // PatchProxyBufferProvider interface
     // This interface is used from DirectOutputThread to acquire data from HAL.
-    bool producesBufferOnDemand() const override { return true; }
-    status_t obtainBuffer(Proxy::Buffer *buffer, const struct timespec *timeOut = nullptr) override;
-    void releaseBuffer(Proxy::Buffer *buffer) override;
+    bool producesBufferOnDemand() const final { return true; }
+    status_t obtainBuffer(Proxy::Buffer* buffer, const struct timespec* timeOut = nullptr) final;
+    void releaseBuffer(Proxy::Buffer* buffer) final;
 
 private:
     // This is to use with PatchRecord::writeFrames
@@ -215,7 +213,7 @@
         PassthruPatchRecord& mPassthru;
     };
 
-    sp<StreamInHalInterface> obtainStream(sp<ThreadBase>* thread);
+    sp<StreamInHalInterface> obtainStream(sp<IAfThreadBase>* thread);
 
     PatchRecordAudioBufferProvider mPatchRecordAudioBufferProvider;
     std::unique_ptr<void, decltype(free)*> mSinkBuffer;  // frame size aligned continuous buffer
@@ -227,3 +225,5 @@
     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 e1017f5..eb6b4f3 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -76,6 +76,8 @@
 #include <media/audiohal/StreamHalInterface.h>
 
 #include "AudioFlinger.h"
+#include "Threads.h"
+
 #include <mediautils/SchedulingPolicyService.h>
 #include <mediautils/ServiceUtilities.h>
 
@@ -92,6 +94,8 @@
 #include <fastpath/AutoPark.h>
 
 #include <pthread.h>
+#include <afutils/DumpTryLock.h>
+#include <afutils/Permission.h>
 #include <afutils/TypedLogger.h>
 
 // ----------------------------------------------------------------------------
@@ -120,6 +124,7 @@
 
 namespace android {
 
+using audioflinger::SyncEvent;
 using media::IEffectClient;
 using content::AttributionSourceState;
 
@@ -515,7 +520,7 @@
 // ----------------------------------------------------------------------------
 
 // static
-const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type)
+const char* ThreadBase::threadTypeToString(ThreadBase::type_t type)
 {
     switch (type) {
     case MIXER:
@@ -541,11 +546,11 @@
     }
 }
 
-AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+ThreadBase::ThreadBase(const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
         type_t type, bool systemReady, bool isOut)
     :   Thread(false /*canCallJava*/),
         mType(type),
-        mAudioFlinger(audioFlinger),
+        mAfThreadCallback(afThreadCallback),
         mThreadMetrics(std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_THREAD) + std::to_string(id),
                isOut),
         mIsOut(isOut),
@@ -564,7 +569,7 @@
     memset(&mPatch, 0, sizeof(struct audio_patch));
 }
 
-AudioFlinger::ThreadBase::~ThreadBase()
+ThreadBase::~ThreadBase()
 {
     // mConfigEvents should be empty, but just in case it isn't, free the memory it owns
     mConfigEvents.clear();
@@ -579,7 +584,7 @@
     sendStatistics(true /* force */);
 }
 
-status_t AudioFlinger::ThreadBase::readyToRun()
+status_t ThreadBase::readyToRun()
 {
     status_t status = initCheck();
     if (status == NO_ERROR) {
@@ -590,7 +595,7 @@
     return status;
 }
 
-void AudioFlinger::ThreadBase::exit()
+void ThreadBase::exit()
 {
     ALOGV("ThreadBase::exit");
     // do any cleanup required for exit to succeed
@@ -614,7 +619,7 @@
     requestExitAndWait();
 }
 
-status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
+status_t ThreadBase::setParameters(const String8& keyValuePairs)
 {
     ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
     Mutex::Autolock _l(mLock);
@@ -624,7 +629,7 @@
 
 // sendConfigEvent_l() must be called with ThreadBase::mLock held
 // Can temporarily release the lock if waiting for a reply from processConfigEvents_l().
-status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event)
+status_t ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event)
 NO_THREAD_SAFETY_ANALYSIS  // condition variable
 {
     status_t status = NO_ERROR;
@@ -652,7 +657,7 @@
     return status;
 }
 
-void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event_t event, pid_t pid,
+void ThreadBase::sendIoConfigEvent(audio_io_config_event_t event, pid_t pid,
                                                  audio_port_handle_t portId)
 {
     Mutex::Autolock _l(mLock);
@@ -660,7 +665,7 @@
 }
 
 // sendIoConfigEvent_l() must be called with ThreadBase::mLock held
-void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event_t event, pid_t pid,
+void ThreadBase::sendIoConfigEvent_l(audio_io_config_event_t event, pid_t pid,
                                                    audio_port_handle_t portId)
 {
     // The audio statistics history is exponentially weighted to forget events
@@ -677,14 +682,14 @@
     sendConfigEvent_l(configEvent);
 }
 
-void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp)
+void ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp)
 {
     Mutex::Autolock _l(mLock);
     sendPrioConfigEvent_l(pid, tid, prio, forApp);
 }
 
 // sendPrioConfigEvent_l() must be called with ThreadBase::mLock held
-void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(
+void ThreadBase::sendPrioConfigEvent_l(
         pid_t pid, pid_t tid, int32_t prio, bool forApp)
 {
     sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio, forApp);
@@ -692,7 +697,7 @@
 }
 
 // sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held
-status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair)
+status_t ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair)
 {
     sp<ConfigEvent> configEvent;
     AudioParameter param(keyValuePair);
@@ -710,7 +715,7 @@
     return sendConfigEvent_l(configEvent);
 }
 
-status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent(
+status_t ThreadBase::sendCreateAudioPatchConfigEvent(
                                                         const struct audio_patch *patch,
                                                         audio_patch_handle_t *handle)
 {
@@ -725,7 +730,7 @@
     return status;
 }
 
-status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent(
+status_t ThreadBase::sendReleaseAudioPatchConfigEvent(
                                                                 const audio_patch_handle_t handle)
 {
     Mutex::Autolock _l(mLock);
@@ -733,7 +738,7 @@
     return sendConfigEvent_l(configEvent);
 }
 
-status_t AudioFlinger::ThreadBase::sendUpdateOutDeviceConfigEvent(
+status_t ThreadBase::sendUpdateOutDeviceConfigEvent(
         const DeviceDescriptorBaseVector& outDevices)
 {
     if (type() != RECORD) {
@@ -745,7 +750,7 @@
     return sendConfigEvent_l(configEvent);
 }
 
-void AudioFlinger::ThreadBase::sendResizeBufferConfigEvent_l(int32_t maxSharedAudioHistoryMs)
+void ThreadBase::sendResizeBufferConfigEvent_l(int32_t maxSharedAudioHistoryMs)
 {
     ALOG_ASSERT(type() == RECORD, "sendResizeBufferConfigEvent_l() called on non record thread");
     sp<ConfigEvent> configEvent =
@@ -753,27 +758,27 @@
     sendConfigEvent_l(configEvent);
 }
 
-void AudioFlinger::ThreadBase::sendCheckOutputStageEffectsEvent()
+void ThreadBase::sendCheckOutputStageEffectsEvent()
 {
     Mutex::Autolock _l(mLock);
     sendCheckOutputStageEffectsEvent_l();
 }
 
-void AudioFlinger::ThreadBase::sendCheckOutputStageEffectsEvent_l()
+void ThreadBase::sendCheckOutputStageEffectsEvent_l()
 {
     sp<ConfigEvent> configEvent =
             (ConfigEvent *)new CheckOutputStageEffectsEvent();
     sendConfigEvent_l(configEvent);
 }
 
-void AudioFlinger::ThreadBase::sendHalLatencyModesChangedEvent_l()
+void ThreadBase::sendHalLatencyModesChangedEvent_l()
 {
     sp<ConfigEvent> configEvent = sp<HalLatencyModesChangedEvent>::make();
     sendConfigEvent_l(configEvent);
 }
 
 // post condition: mConfigEvents.isEmpty()
-void AudioFlinger::ThreadBase::processConfigEvents_l()
+void ThreadBase::processConfigEvents_l()
 {
     bool configChanged = false;
 
@@ -940,13 +945,13 @@
     }
 }
 
-void AudioFlinger::ThreadBase::dump(int fd, const Vector<String16>& args)
+void ThreadBase::dump(int fd, const Vector<String16>& args)
 NO_THREAD_SAFETY_ANALYSIS  // conditional try lock
 {
     dprintf(fd, "\n%s thread %p, name %s, tid %d, type %d (%s):\n", isOutput() ? "Output" : "Input",
             this, mThreadName, getTid(), type(), threadTypeToString(type()));
 
-    bool locked = AudioFlinger::dumpTryLock(mLock);
+    const bool locked = afutils::dumpTryLock(mLock);
     if (!locked) {
         dprintf(fd, "  Thread may be deadlocked\n");
     }
@@ -978,7 +983,7 @@
     }
 }
 
-void AudioFlinger::ThreadBase::dumpBase_l(int fd, const Vector<String16>& args __unused)
+void ThreadBase::dumpBase_l(int fd, const Vector<String16>& /* args */)
 {
     dprintf(fd, "  I/O handle: %d\n", mId);
     dprintf(fd, "  Standby: %s\n", mStandby ? "yes" : "no");
@@ -1051,7 +1056,7 @@
     }
 }
 
-void AudioFlinger::ThreadBase::dumpEffectChains_l(int fd, const Vector<String16>& args)
+void ThreadBase::dumpEffectChains_l(int fd, const Vector<String16>& args)
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
@@ -1068,13 +1073,13 @@
     }
 }
 
-void AudioFlinger::ThreadBase::acquireWakeLock()
+void ThreadBase::acquireWakeLock()
 {
     Mutex::Autolock _l(mLock);
     acquireWakeLock_l();
 }
 
-String16 AudioFlinger::ThreadBase::getWakeLockTag()
+String16 ThreadBase::getWakeLockTag()
 {
     switch (mType) {
     case MIXER:
@@ -1099,7 +1104,7 @@
     }
 }
 
-void AudioFlinger::ThreadBase::acquireWakeLock_l()
+void ThreadBase::acquireWakeLock_l()
 {
     getPowerManager_l();
     if (mPowerManager != 0) {
@@ -1122,13 +1127,13 @@
             gBoottime.getBoottimeOffset();
 }
 
-void AudioFlinger::ThreadBase::releaseWakeLock()
+void ThreadBase::releaseWakeLock()
 {
     Mutex::Autolock _l(mLock);
     releaseWakeLock_l();
 }
 
-void AudioFlinger::ThreadBase::releaseWakeLock_l()
+void ThreadBase::releaseWakeLock_l()
 {
     gBoottime.release(mWakeLockToken);
     if (mWakeLockToken != 0) {
@@ -1140,7 +1145,7 @@
     }
 }
 
-void AudioFlinger::ThreadBase::getPowerManager_l() {
+void ThreadBase::getPowerManager_l() {
     if (mSystemReady && mPowerManager == 0) {
         // use checkService() to avoid blocking if power service is not up yet
         sp<IBinder> binder =
@@ -1154,7 +1159,7 @@
     }
 }
 
-void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<uid_t> &uids) {
+void ThreadBase::updateWakeLockUids_l(const SortedVector<uid_t>& uids) {
     getPowerManager_l();
 
 #if !LOG_NDEBUG
@@ -1181,25 +1186,25 @@
     }
 }
 
-void AudioFlinger::ThreadBase::clearPowerManager()
+void ThreadBase::clearPowerManager()
 {
     Mutex::Autolock _l(mLock);
     releaseWakeLock_l();
     mPowerManager.clear();
 }
 
-void AudioFlinger::ThreadBase::updateOutDevices(
+void ThreadBase::updateOutDevices(
         const DeviceDescriptorBaseVector& outDevices __unused)
 {
     ALOGE("%s should only be called in RecordThread", __func__);
 }
 
-void AudioFlinger::ThreadBase::resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs __unused)
+void ThreadBase::resizeInputBuffer_l(int32_t /* maxSharedAudioHistoryMs */)
 {
     ALOGE("%s should only be called in RecordThread", __func__);
 }
 
-void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused)
+void ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& /* who */)
 {
     sp<ThreadBase> thread = mThread.promote();
     if (thread != 0) {
@@ -1208,7 +1213,7 @@
     ALOGW("power manager service died !!!");
 }
 
-void AudioFlinger::ThreadBase::setEffectSuspended_l(
+void ThreadBase::setEffectSuspended_l(
         const effect_uuid_t *type, bool suspend, audio_session_t sessionId)
 {
     sp<IAfEffectChain> chain = getEffectChain_l(sessionId);
@@ -1223,7 +1228,7 @@
     updateSuspendedSessions_l(type, suspend, sessionId);
 }
 
-void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<IAfEffectChain>& chain)
+void ThreadBase::checkSuspendOnAddEffectChain_l(const sp<IAfEffectChain>& chain)
 {
     ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId());
     if (index < 0) {
@@ -1247,7 +1252,7 @@
     }
 }
 
-void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
+void ThreadBase::updateSuspendedSessions_l(const effect_uuid_t* type,
                                                          bool suspend,
                                                          audio_session_t sessionId)
 {
@@ -1308,7 +1313,7 @@
     }
 }
 
-void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(bool enabled,
+void ThreadBase::checkSuspendOnEffectEnabled(bool enabled,
                                                            audio_session_t sessionId,
                                                            bool threadLocked)
 NO_THREAD_SAFETY_ANALYSIS  // manual locking
@@ -1334,7 +1339,7 @@
 }
 
 // checkEffectCompatibility_l() must be called with ThreadBase::mLock held
-status_t AudioFlinger::RecordThread::checkEffectCompatibility_l(
+status_t RecordThread::checkEffectCompatibility_l(
         const effect_descriptor_t *desc, audio_session_t sessionId)
 {
     // No global output effect sessions on record threads
@@ -1378,7 +1383,7 @@
 }
 
 // checkEffectCompatibility_l() must be called with ThreadBase::mLock held
-status_t AudioFlinger::PlaybackThread::checkEffectCompatibility_l(
+status_t PlaybackThread::checkEffectCompatibility_l(
         const effect_descriptor_t *desc, audio_session_t sessionId)
 {
     // no preprocessing on playback threads
@@ -1533,7 +1538,7 @@
 }
 
 // ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held
-sp<IAfEffectHandle> AudioFlinger::ThreadBase::createEffect_l(
+sp<IAfEffectHandle> ThreadBase::createEffect_l(
         const sp<Client>& client,
         const sp<IEffectClient>& effectClient,
         int32_t priority,
@@ -1585,7 +1590,7 @@
         ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get());
 
         if (effect == 0) {
-            effectId = mAudioFlinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT);
+            effectId = mAfThreadCallback->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT);
             // create a new effect module if none present in the chain
             lStatus = chain->createEffect_l(effect, desc, effectId, sessionId, pinned);
             if (lStatus != NO_ERROR) {
@@ -1596,14 +1601,14 @@
             // FIXME: use vector of device and address when effect interface is ready.
             effect->setDevices(outDeviceTypeAddrs());
             effect->setInputDevice(inDeviceTypeAddr());
-            effect->setMode(mAudioFlinger->getMode());
+            effect->setMode(mAfThreadCallback->getMode());
             effect->setAudioSource(mAudioSource);
         }
         if (effect->isHapticGenerator()) {
             // TODO(b/184194057): Use the vibrator information from the vibrator that will be used
             // for the HapticGenerator.
             const std::optional<media::AudioVibratorInfo> defaultVibratorInfo =
-                    std::move(mAudioFlinger->getDefaultVibratorInfo_l());
+                    std::move(mAfThreadCallback->getDefaultVibratorInfo_l());
             if (defaultVibratorInfo) {
                 // Only set the vibrator info when it is a valid one.
                 effect->setVibratorInfo(*defaultVibratorInfo);
@@ -1638,7 +1643,7 @@
     return handle;
 }
 
-void AudioFlinger::ThreadBase::disconnectEffectHandle(IAfEffectHandle *handle,
+void ThreadBase::disconnectEffectHandle(IAfEffectHandle* handle,
                                                       bool unpinIfLast)
 {
     bool remove = false;
@@ -1661,14 +1666,14 @@
         sendCheckOutputStageEffectsEvent_l();
     }
     if (remove) {
-        mAudioFlinger->updateOrphanEffectChains(effect);
+        mAfThreadCallback->updateOrphanEffectChains(effect);
         if (handle->enabled()) {
             effect->checkSuspendOnEffectEnabled(false, false /*threadLocked*/);
         }
     }
 }
 
-void AudioFlinger::ThreadBase::onEffectEnable(const sp<IAfEffectModule>& effect) {
+void ThreadBase::onEffectEnable(const sp<IAfEffectModule>& effect) {
     if (isOffloadOrMmap()) {
         Mutex::Autolock _l(mLock);
         broadcast_l();
@@ -1679,33 +1684,33 @@
             t->invalidateTracks(AUDIO_STREAM_MUSIC);
         }
         if (effect->sessionId() == AUDIO_SESSION_OUTPUT_MIX) {
-            mAudioFlinger->onNonOffloadableGlobalEffectEnable();
+            mAfThreadCallback->onNonOffloadableGlobalEffectEnable();
         }
     }
 }
 
-void AudioFlinger::ThreadBase::onEffectDisable() {
+void ThreadBase::onEffectDisable() {
     if (isOffloadOrMmap()) {
         Mutex::Autolock _l(mLock);
         broadcast_l();
     }
 }
 
-sp<IAfEffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId,
-        int effectId)
+sp<IAfEffectModule> ThreadBase::getEffect(audio_session_t sessionId,
+        int effectId) const
 {
     Mutex::Autolock _l(mLock);
     return getEffect_l(sessionId, effectId);
 }
 
-sp<IAfEffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t sessionId,
-        int effectId)
+sp<IAfEffectModule> ThreadBase::getEffect_l(audio_session_t sessionId,
+        int effectId) const
 {
     sp<IAfEffectChain> chain = getEffectChain_l(sessionId);
     return chain != 0 ? chain->getEffectFromId_l(effectId) : 0;
 }
 
-std::vector<int> AudioFlinger::ThreadBase::getEffectIds_l(audio_session_t sessionId)
+std::vector<int> ThreadBase::getEffectIds_l(audio_session_t sessionId) const
 {
     sp<IAfEffectChain> chain = getEffectChain_l(sessionId);
     return chain != nullptr ? chain->getEffectIds() : std::vector<int>{};
@@ -1713,7 +1718,7 @@
 
 // PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
 // PlaybackThread::mLock held
-status_t AudioFlinger::ThreadBase::addEffect_l(const sp<IAfEffectModule>& effect)
+status_t ThreadBase::addEffect_l(const sp<IAfEffectModule>& effect)
 {
     // check for existing effect chain with the requested audio session
     audio_session_t sessionId = effect->sessionId();
@@ -1752,13 +1757,13 @@
 
     effect->setDevices(outDeviceTypeAddrs());
     effect->setInputDevice(inDeviceTypeAddr());
-    effect->setMode(mAudioFlinger->getMode());
+    effect->setMode(mAfThreadCallback->getMode());
     effect->setAudioSource(mAudioSource);
 
     return NO_ERROR;
 }
 
-void AudioFlinger::ThreadBase::removeEffect_l(const sp<IAfEffectModule>& effect, bool release) {
+void ThreadBase::removeEffect_l(const sp<IAfEffectModule>& effect, bool release) {
 
     ALOGV("%s %p effect %p", __FUNCTION__, this, effect.get());
     effect_descriptor_t desc = effect->desc();
@@ -1777,7 +1782,7 @@
     }
 }
 
-void AudioFlinger::ThreadBase::lockEffectChains_l(
+void ThreadBase::lockEffectChains_l(
         Vector<sp<IAfEffectChain>>& effectChains)
 NO_THREAD_SAFETY_ANALYSIS  // calls EffectChain::lock()
 {
@@ -1787,7 +1792,7 @@
     }
 }
 
-void AudioFlinger::ThreadBase::unlockEffectChains(
+void ThreadBase::unlockEffectChains(
         const Vector<sp<IAfEffectChain>>& effectChains)
 NO_THREAD_SAFETY_ANALYSIS  // calls EffectChain::unlock()
 {
@@ -1796,13 +1801,13 @@
     }
 }
 
-sp<IAfEffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId)
+sp<IAfEffectChain> ThreadBase::getEffectChain(audio_session_t sessionId) const
 {
     Mutex::Autolock _l(mLock);
     return getEffectChain_l(sessionId);
 }
 
-sp<IAfEffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t sessionId)
+sp<IAfEffectChain> ThreadBase::getEffectChain_l(audio_session_t sessionId)
         const
 {
     size_t size = mEffectChains.size();
@@ -1814,7 +1819,7 @@
     return 0;
 }
 
-void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
+void ThreadBase::setMode(audio_mode_t mode)
 {
     Mutex::Autolock _l(mLock);
     size_t size = mEffectChains.size();
@@ -1823,7 +1828,7 @@
     }
 }
 
-void AudioFlinger::ThreadBase::toAudioPortConfig(struct audio_port_config *config)
+void ThreadBase::toAudioPortConfig(struct audio_port_config* config)
 {
     config->type = AUDIO_PORT_TYPE_MIX;
     config->ext.mix.handle = mId;
@@ -1834,7 +1839,7 @@
                             AUDIO_PORT_CONFIG_FORMAT;
 }
 
-void AudioFlinger::ThreadBase::systemReady()
+void ThreadBase::systemReady()
 {
     Mutex::Autolock _l(mLock);
     if (mSystemReady) {
@@ -1849,7 +1854,7 @@
 }
 
 template <typename T>
-ssize_t AudioFlinger::ThreadBase::ActiveTracks<T>::add(const sp<T> &track) {
+ssize_t ThreadBase::ActiveTracks<T>::add(const sp<T>& track) {
     ssize_t index = mActiveTracks.indexOf(track);
     if (index >= 0) {
         ALOGW("ActiveTracks<T>::add track %p already there", track.get());
@@ -1858,13 +1863,13 @@
     logTrack("add", track);
     mActiveTracksGeneration++;
     mLatestActiveTrack = track;
-    ++mBatteryCounter[track->uid()].second;
+    track->beginBatteryAttribution();
     mHasChanged = true;
     return mActiveTracks.add(track);
 }
 
 template <typename T>
-ssize_t AudioFlinger::ThreadBase::ActiveTracks<T>::remove(const sp<T> &track) {
+ssize_t ThreadBase::ActiveTracks<T>::remove(const sp<T>& track) {
     ssize_t index = mActiveTracks.remove(track);
     if (index < 0) {
         ALOGW("ActiveTracks<T>::remove nonexistent track %p", track.get());
@@ -1872,7 +1877,7 @@
     }
     logTrack("remove", track);
     mActiveTracksGeneration++;
-    --mBatteryCounter[track->uid()].second;
+    track->endBatteryAttribution();
     // mLatestActiveTrack is not cleared even if is the same as track.
     mHasChanged = true;
 #ifdef TEE_SINK
@@ -1883,51 +1888,29 @@
 }
 
 template <typename T>
-void AudioFlinger::ThreadBase::ActiveTracks<T>::clear() {
+void ThreadBase::ActiveTracks<T>::clear() {
     for (const sp<T> &track : mActiveTracks) {
-        BatteryNotifier::getInstance().noteStopAudio(track->uid());
+        track->endBatteryAttribution();
         logTrack("clear", track);
     }
     mLastActiveTracksGeneration = mActiveTracksGeneration;
     if (!mActiveTracks.empty()) { mHasChanged = true; }
     mActiveTracks.clear();
     mLatestActiveTrack.clear();
-    mBatteryCounter.clear();
 }
 
 template <typename T>
-void AudioFlinger::ThreadBase::ActiveTracks<T>::updatePowerState(
+void ThreadBase::ActiveTracks<T>::updatePowerState(
         const sp<ThreadBase>& thread, bool force) {
     // Updates ActiveTracks client uids to the thread wakelock.
     if (mActiveTracksGeneration != mLastActiveTracksGeneration || force) {
         thread->updateWakeLockUids_l(getWakeLockUids());
         mLastActiveTracksGeneration = mActiveTracksGeneration;
     }
-
-    // Updates BatteryNotifier uids
-    for (auto it = mBatteryCounter.begin(); it != mBatteryCounter.end();) {
-        const uid_t uid = it->first;
-        ssize_t &previous = it->second.first;
-        ssize_t &current = it->second.second;
-        if (current > 0) {
-            if (previous == 0) {
-                BatteryNotifier::getInstance().noteStartAudio(uid);
-            }
-            previous = current;
-            ++it;
-        } else if (current == 0) {
-            if (previous > 0) {
-                BatteryNotifier::getInstance().noteStopAudio(uid);
-            }
-            it = mBatteryCounter.erase(it); // std::map<> is stable on iterator erase.
-        } else /* (current < 0) */ {
-            LOG_ALWAYS_FATAL("negative battery count %zd", current);
-        }
-    }
 }
 
 template <typename T>
-bool AudioFlinger::ThreadBase::ActiveTracks<T>::readAndClearHasChanged() {
+bool ThreadBase::ActiveTracks<T>::readAndClearHasChanged() {
     bool hasChanged = mHasChanged;
     mHasChanged = false;
 
@@ -1940,7 +1923,7 @@
 }
 
 template <typename T>
-void AudioFlinger::ThreadBase::ActiveTracks<T>::logTrack(
+void ThreadBase::ActiveTracks<T>::logTrack(
         const char *funcName, const sp<T> &track) const {
     if (mLocalLog != nullptr) {
         String8 result;
@@ -1949,7 +1932,7 @@
     }
 }
 
-void AudioFlinger::ThreadBase::broadcast_l()
+void ThreadBase::broadcast_l()
 {
     // Thread could be blocked waiting for async
     // so signal it to handle state changes immediately
@@ -1961,7 +1944,7 @@
 
 // Call only from threadLoop() or when it is idle.
 // Do not call from high performance code as this may do binder rpc to the MediaMetrics service.
-void AudioFlinger::ThreadBase::sendStatistics(bool force)
+void ThreadBase::sendStatistics(bool force)
 {
     // Do not log if we have no stats.
     // We choose the timestamp verifier because it is the most likely item to be present.
@@ -2024,16 +2007,16 @@
     item->selfrecord();
 }
 
-product_strategy_t AudioFlinger::ThreadBase::getStrategyForStream(audio_stream_type_t stream) const
+product_strategy_t ThreadBase::getStrategyForStream(audio_stream_type_t stream) const
 {
-    if (!mAudioFlinger->isAudioPolicyReady()) {
+    if (!mAfThreadCallback->isAudioPolicyReady()) {
         return PRODUCT_STRATEGY_NONE;
     }
     return AudioSystem::getStrategyForStream(stream);
 }
 
 // startMelComputation_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::ThreadBase::startMelComputation_l(
+void ThreadBase::startMelComputation_l(
         const sp<audio_utils::MelProcessor>& /*processor*/)
 {
     // Do nothing
@@ -2041,7 +2024,7 @@
 }
 
 // stopMelComputation_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::ThreadBase::stopMelComputation_l()
+void ThreadBase::stopMelComputation_l()
 {
     // Do nothing
     ALOGW("%s: ThreadBase does not support CSD", __func__);
@@ -2051,13 +2034,13 @@
 //      Playback
 // ----------------------------------------------------------------------------
 
-AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
+PlaybackThread::PlaybackThread(const sp<IAfThreadCallback>& afThreadCallback,
                                              AudioStreamOut* output,
                                              audio_io_handle_t id,
                                              type_t type,
                                              bool systemReady,
                                              audio_config_base_t *mixerConfig)
-    :   ThreadBase(audioFlinger, id, type, systemReady, true /* isOut */),
+    :   ThreadBase(afThreadCallback, id, type, systemReady, true /* isOut */),
         mNormalFrameCount(0), mSinkBuffer(NULL),
         mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision || type == SPATIALIZER),
         mMixerBuffer(NULL),
@@ -2085,7 +2068,7 @@
         mUseAsyncWrite(false),
         mWriteAckSequence(0),
         mDrainSequence(0),
-        mScreenState(AudioFlinger::mScreenState),
+        mScreenState(mAfThreadCallback->getScreenState()),
         // index 0 is reserved for normal mixer's submix
         mFastTrackAvailMask(((1 << FastMixerState::sMaxFastTracks) - 1) & ~1),
         mHwSupportsPause(false), mHwPaused(false), mFlushPending(false),
@@ -2094,7 +2077,7 @@
         mIsTimestampAdvancing(kMinimumTimeBetweenTimestampChecksNs)
 {
     snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id);
-    mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName);
+    mNBLogWriter = afThreadCallback->newWriter_l(kLogSize, mThreadName);
 
     // Assumes constructor is called by AudioFlinger with it's mLock held, but
     // it would be safer to explicitly pass initial masterVolume/masterMute as
@@ -2103,8 +2086,8 @@
     // If the HAL we are using has support for master volume or master mute,
     // then do not attenuate or mute during mixing (just leave the volume at 1.0
     // and the mute set to false).
-    mMasterVolume = audioFlinger->masterVolume_l();
-    mMasterMute = audioFlinger->masterMute_l();
+    mMasterVolume = afThreadCallback->masterVolume_l();
+    mMasterMute = afThreadCallback->masterMute_l();
     if (mOutput->audioHwDev) {
         if (mOutput->audioHwDev->canSetMasterVolume()) {
             mMasterVolume = 1.0;
@@ -2142,7 +2125,7 @@
     for (int i = AUDIO_STREAM_MIN; i < AUDIO_STREAM_FOR_POLICY_CNT; ++i) {
         const audio_stream_type_t stream{static_cast<audio_stream_type_t>(i)};
         mStreamTypes[stream].volume = 0.0f;
-        mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream);
+        mStreamTypes[stream].mute = mAfThreadCallback->streamMute_l(stream);
     }
     // Audio patch and call assistant volume are always max
     mStreamTypes[AUDIO_STREAM_PATCH].volume = 1.0f;
@@ -2151,9 +2134,9 @@
     mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].mute = false;
 }
 
-AudioFlinger::PlaybackThread::~PlaybackThread()
+PlaybackThread::~PlaybackThread()
 {
-    mAudioFlinger->unregisterWriter(mNBLogWriter);
+    mAfThreadCallback->unregisterWriter(mNBLogWriter);
     free(mSinkBuffer);
     free(mMixerBuffer);
     free(mEffectBuffer);
@@ -2162,7 +2145,7 @@
 
 // Thread virtuals
 
-void AudioFlinger::PlaybackThread::onFirstRef()
+void PlaybackThread::onFirstRef()
 {
     if (!isStreamInitialized()) {
         ALOGE("The stream is not open yet"); // This should not happen.
@@ -2177,7 +2160,7 @@
         if (mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING &&
                 mOutput->stream->setCallback(this) == OK) {
             mUseAsyncWrite = true;
-            mCallbackThread = new AudioFlinger::AsyncCallbackThread(this);
+            mCallbackThread = sp<AsyncCallbackThread>::make(this);
         }
 
         if (mOutput->stream->setEventCallback(this) != OK) {
@@ -2189,14 +2172,14 @@
 }
 
 // ThreadBase virtuals
-void AudioFlinger::PlaybackThread::preExit()
+void PlaybackThread::preExit()
 {
     ALOGV("  preExit()");
     status_t result = mOutput->stream->exit();
     ALOGE_IF(result != OK, "Error when calling exit(): %d", result);
 }
 
-void AudioFlinger::PlaybackThread::dumpTracks_l(int fd, const Vector<String16>& args __unused)
+void PlaybackThread::dumpTracks_l(int fd, const Vector<String16>& /* args */)
 {
     String8 result;
 
@@ -2230,7 +2213,7 @@
         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) {
@@ -2250,7 +2233,7 @@
         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 */);
@@ -2261,7 +2244,7 @@
     write(fd, result.string(), result.size());
 }
 
-void AudioFlinger::PlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args)
+void PlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args)
 {
     dprintf(fd, "  Master volume: %f\n", mMasterVolume);
     dprintf(fd, "  Master mute: %s\n", mMasterMute ? "on" : "off");
@@ -2297,7 +2280,7 @@
 }
 
 // PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
-sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
+sp<IAfTrack> PlaybackThread::createTrack_l(
         const sp<Client>& client,
         audio_stream_type_t streamType,
         const audio_attributes_t& attr,
@@ -2322,7 +2305,7 @@
 {
     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;
@@ -2615,7 +2598,7 @@
         // 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) {
@@ -2637,11 +2620,11 @@
             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;
@@ -2682,7 +2665,7 @@
 }
 
 template<typename T>
-ssize_t AudioFlinger::PlaybackThread::Tracks<T>::remove(const sp<T> &track)
+ssize_t PlaybackThread::Tracks<T>::remove(const sp<T>& track)
 {
     const int trackId = track->id();
     const ssize_t index = mTracks.remove(track);
@@ -2697,17 +2680,17 @@
     return index;
 }
 
-uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const
+uint32_t PlaybackThread::correctLatency_l(uint32_t latency) const
 {
     return latency;
 }
 
-uint32_t AudioFlinger::PlaybackThread::latency() const
+uint32_t PlaybackThread::latency() const
 {
     Mutex::Autolock _l(mLock);
     return latency_l();
 }
-uint32_t AudioFlinger::PlaybackThread::latency_l() const
+uint32_t PlaybackThread::latency_l() const
 {
     uint32_t latency;
     if (initCheck() == NO_ERROR && mOutput->stream->getLatency(&latency) == OK) {
@@ -2716,7 +2699,7 @@
     return 0;
 }
 
-void AudioFlinger::PlaybackThread::setMasterVolume(float value)
+void PlaybackThread::setMasterVolume(float value)
 {
     Mutex::Autolock _l(mLock);
     // Don't apply master volume in SW if our HAL can do it for us.
@@ -2728,12 +2711,12 @@
     }
 }
 
-void AudioFlinger::PlaybackThread::setMasterBalance(float balance)
+void PlaybackThread::setMasterBalance(float balance)
 {
     mMasterBalance.store(balance);
 }
 
-void AudioFlinger::PlaybackThread::setMasterMute(bool muted)
+void PlaybackThread::setMasterMute(bool muted)
 {
     if (isDuplicating()) {
         return;
@@ -2748,33 +2731,33 @@
     }
 }
 
-void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
+void PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
 {
     Mutex::Autolock _l(mLock);
     mStreamTypes[stream].volume = value;
     broadcast_l();
 }
 
-void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
+void PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
 {
     Mutex::Autolock _l(mLock);
     mStreamTypes[stream].mute = muted;
     broadcast_l();
 }
 
-float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
+float PlaybackThread::streamVolume(audio_stream_type_t stream) const
 {
     Mutex::Autolock _l(mLock);
     return mStreamTypes[stream].volume;
 }
 
-void AudioFlinger::PlaybackThread::setVolumeForOutput_l(float left, float right) const
+void PlaybackThread::setVolumeForOutput_l(float left, float right) const
 {
     mOutput->stream->setVolume(left, right);
 }
 
 // addTrack_l() must be called with ThreadBase::mLock held
-status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
+status_t PlaybackThread::addTrack_l(const sp<IAfTrack>& track)
 NO_THREAD_SAFETY_ANALYSIS  // release and re-acquire mLock
 {
     status_t status = ALREADY_EXISTS;
@@ -2784,12 +2767,12 @@
         // 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());
@@ -2815,15 +2798,15 @@
         // 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());
@@ -2839,8 +2822,8 @@
             {
                 // TODO(b/184194780): Use the vibrator information from the vibrator that will be
                 // used to play this track.
-                Mutex::Autolock _l(mAudioFlinger->mLock);
-                vibratorInfo = std::move(mAudioFlinger->getDefaultVibratorInfo_l());
+                Mutex::Autolock _l(mAfThreadCallback->mutex());
+                vibratorInfo = std::move(mAfThreadCallback->getDefaultVibratorInfo_l());
             }
             mLock.lock();
             track->setHapticIntensity(intensity);
@@ -2863,7 +2846,7 @@
             }
         }
 
-        track->mResetDone = false;
+        track->setResetDone(false);
         track->resetPresentationComplete();
         mActiveTracks.add(track);
         if (chain != 0) {
@@ -2880,25 +2863,25 @@
     return status;
 }
 
-bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
+bool 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 PlaybackThread::removeTrack_l(const sp<IAfTrack>& track)
 {
     track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
 
@@ -2912,12 +2895,12 @@
         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) {
@@ -2925,7 +2908,7 @@
     }
 }
 
-String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
+String8 PlaybackThread::getParameters(const String8& keys)
 {
     Mutex::Autolock _l(mLock);
     String8 out_s8;
@@ -2935,7 +2918,7 @@
     return {};
 }
 
-status_t AudioFlinger::DirectOutputThread::selectPresentation(int presentationId, int programId) {
+status_t DirectOutputThread::selectPresentation(int presentationId, int programId) {
     Mutex::Autolock _l(mLock);
     if (!isStreamInitialized()) {
         return NO_INIT;
@@ -2943,7 +2926,7 @@
     return mOutput->stream->selectPresentation(presentationId, programId);
 }
 
-void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
+void PlaybackThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
                                                    audio_port_handle_t portId) {
     ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event);
     sp<AudioIoDescriptor> desc;
@@ -2965,30 +2948,30 @@
         desc = sp<AudioIoDescriptor>::make(mId);
         break;
     }
-    mAudioFlinger->ioConfigChanged(event, desc, pid);
+    mAfThreadCallback->ioConfigChanged(event, desc, pid);
 }
 
-void AudioFlinger::PlaybackThread::onWriteReady()
+void PlaybackThread::onWriteReady()
 {
     mCallbackThread->resetWriteBlocked();
 }
 
-void AudioFlinger::PlaybackThread::onDrainReady()
+void PlaybackThread::onDrainReady()
 {
     mCallbackThread->resetDraining();
 }
 
-void AudioFlinger::PlaybackThread::onError()
+void PlaybackThread::onError()
 {
     mCallbackThread->setAsyncError();
 }
 
-void AudioFlinger::PlaybackThread::onCodecFormatChanged(
+void PlaybackThread::onCodecFormatChanged(
         const std::basic_string<uint8_t>& metadataBs)
 {
-    wp<AudioFlinger::PlaybackThread> weakPointerThis = this;
+    const auto weakPointerThis = wp<PlaybackThread>::fromExisting(this);
     std::thread([this, metadataBs, weakPointerThis]() {
-            sp<AudioFlinger::PlaybackThread> playbackThread = weakPointerThis.promote();
+            const sp<PlaybackThread> playbackThread = weakPointerThis.promote();
             if (playbackThread == nullptr) {
                 ALOGW("PlaybackThread was destroyed, skip codec format change event");
                 return;
@@ -3013,7 +2996,7 @@
     }).detach();
 }
 
-void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence)
+void PlaybackThread::resetWriteBlocked(uint32_t sequence)
 {
     Mutex::Autolock _l(mLock);
     // reject out of sequence requests
@@ -3023,7 +3006,7 @@
     }
 }
 
-void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence)
+void PlaybackThread::resetDraining(uint32_t sequence)
 {
     Mutex::Autolock _l(mLock);
     // reject out of sequence requests
@@ -3038,7 +3021,7 @@
     }
 }
 
-void AudioFlinger::PlaybackThread::readOutputParameters_l()
+void PlaybackThread::readOutputParameters_l()
 {
     // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL
     const audio_config_base_t audioConfig = mOutput->getAudioProperties();
@@ -3047,7 +3030,7 @@
     if (!audio_is_output_channel(mChannelMask)) {
         LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask);
     }
-    if (hasMixer() && !isValidPcmSinkChannelMask(mChannelMask)) {
+    if (hasMixer() && !AudioFlinger::isValidPcmSinkChannelMask(mChannelMask)) {
         LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output",
                 mChannelMask);
     }
@@ -3070,7 +3053,7 @@
     if (!audio_is_valid_format(mFormat)) {
         LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat);
     }
-    if (hasMixer() && !isValidPcmSinkFormat(mFormat)) {
+    if (hasMixer() && !AudioFlinger::isValidPcmSinkFormat(mFormat)) {
         LOG_FATAL("HAL format %#x not supported for mixed output",
                 mFormat);
     }
@@ -3209,7 +3192,7 @@
     // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
     Vector<sp<IAfEffectChain>> effectChains = mEffectChains;
     for (size_t i = 0; i < effectChains.size(); i ++) {
-        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(),
+        mAfThreadCallback->moveEffectChain_l(effectChains[i]->sessionId(),
             this/* srcThread */, this/* dstThread */);
     }
 
@@ -3238,14 +3221,14 @@
     item.record();
 }
 
-AudioFlinger::ThreadBase::MetadataUpdate AudioFlinger::PlaybackThread::updateMetadata_l()
+ThreadBase::MetadataUpdate PlaybackThread::updateMetadata_l()
 {
     if (!isStreamInitialized() || !mActiveTracks.readAndClearHasChanged()) {
         return {}; // nothing to do
     }
     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);
     }
@@ -3255,13 +3238,14 @@
     return change;
 }
 
-void AudioFlinger::PlaybackThread::sendMetadataToBackend_l(
+void PlaybackThread::sendMetadataToBackend_l(
         const StreamOutHalInterface::SourceMetadata& metadata)
 {
     mOutput->stream->updateSourceMetadata(metadata);
 };
 
-status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
+status_t PlaybackThread::getRenderPosition(
+        uint32_t* halFrames, uint32_t* dspFrames) const
 {
     if (halFrames == NULL || dspFrames == NULL) {
         return BAD_VALUE;
@@ -3288,7 +3272,7 @@
     }
 }
 
-product_strategy_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t sessionId)
+product_strategy_t PlaybackThread::getStrategyForSession_l(audio_session_t sessionId) const
 {
     // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
     // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
@@ -3296,7 +3280,7 @@
         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());
         }
@@ -3305,13 +3289,13 @@
 }
 
 
-AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
+AudioStreamOut* PlaybackThread::getOutput() const
 {
     Mutex::Autolock _l(mLock);
     return mOutput;
 }
 
-AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
+AudioStreamOut* PlaybackThread::clearOutput()
 {
     Mutex::Autolock _l(mLock);
     AudioStreamOut *output = mOutput;
@@ -3325,7 +3309,7 @@
 }
 
 // this method must always be called either with ThreadBase mLock held or inside the thread loop
-sp<StreamHalInterface> AudioFlinger::PlaybackThread::stream() const
+sp<StreamHalInterface> PlaybackThread::stream() const
 {
     if (mOutput == NULL) {
         return NULL;
@@ -3333,12 +3317,12 @@
     return mOutput->stream;
 }
 
-uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const
+uint32_t PlaybackThread::activeSleepTimeUs() const
 {
     return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000);
 }
 
-status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<audioflinger::SyncEvent>& event)
+status_t PlaybackThread::setSyncEvent(const sp<SyncEvent>& event)
 {
     if (!isValidSyncEvent(event)) {
         return BAD_VALUE;
@@ -3347,7 +3331,7 @@
     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;
@@ -3357,14 +3341,13 @@
     return NAME_NOT_FOUND;
 }
 
-bool AudioFlinger::PlaybackThread::isValidSyncEvent(
-        const sp<audioflinger::SyncEvent>& event) const
+bool PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const
 {
     return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE;
 }
 
-void AudioFlinger::PlaybackThread::threadLoop_removeTracks(
-        [[maybe_unused]] const Vector< sp<Track> >& tracksToRemove)
+void PlaybackThread::threadLoop_removeTracks(
+        [[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.
@@ -3378,7 +3361,7 @@
 #endif
 }
 
-void AudioFlinger::PlaybackThread::checkSilentMode_l()
+void PlaybackThread::checkSilentMode_l()
 {
     if (!mMasterMute) {
         char value[PROPERTY_VALUE_MAX];
@@ -3404,7 +3387,7 @@
 }
 
 // shared by MIXER and DIRECT, overridden by DUPLICATING
-ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
+ssize_t PlaybackThread::threadLoop_write()
 {
     LOG_HIST_TS();
     mInWrite = true;
@@ -3418,7 +3401,7 @@
 
         ATRACE_BEGIN("write");
         // update the setpoint when AudioFlinger::mScreenState changes
-        uint32_t screenState = AudioFlinger::mScreenState;
+        const uint32_t screenState = mAfThreadCallback->getScreenState();
         if (screenState != mScreenState) {
             mScreenState = screenState;
             MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
@@ -3476,7 +3459,7 @@
 }
 
 // startMelComputation_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::PlaybackThread::startMelComputation_l(
+void PlaybackThread::startMelComputation_l(
         const sp<audio_utils::MelProcessor>& processor)
 {
     auto outputSink = static_cast<AudioStreamOutSink*>(mOutputSink.get());
@@ -3486,7 +3469,7 @@
 }
 
 // stopMelComputation_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::PlaybackThread::stopMelComputation_l()
+void PlaybackThread::stopMelComputation_l()
 {
     auto outputSink = static_cast<AudioStreamOutSink*>(mOutputSink.get());
     if (outputSink != nullptr) {
@@ -3494,7 +3477,7 @@
     }
 }
 
-void AudioFlinger::PlaybackThread::threadLoop_drain()
+void PlaybackThread::threadLoop_drain()
 {
     bool supportsDrain = false;
     if (mOutput->stream->supportsDrain(&supportsDrain) == OK && supportsDrain) {
@@ -3510,12 +3493,12 @@
     }
 }
 
-void AudioFlinger::PlaybackThread::threadLoop_exit()
+void PlaybackThread::threadLoop_exit()
 {
     {
         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.
@@ -3546,7 +3529,7 @@
  - idle sleep time
 */
 
-void AudioFlinger::PlaybackThread::cacheParameters_l()
+void PlaybackThread::cacheParameters_l()
 {
     mSinkBufferSize = mNormalFrameCount * mFrameSize;
     mActiveSleepTimeUs = activeSleepTimeUs();
@@ -3563,14 +3546,14 @@
     }
 }
 
-bool AudioFlinger::PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType)
+bool PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType)
 {
     ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu",
             this,  streamType, mTracks.size());
     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;
@@ -3579,22 +3562,22 @@
     return trackMatch;
 }
 
-void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType)
+void PlaybackThread::invalidateTracks(audio_stream_type_t streamType)
 {
     Mutex::Autolock _l(mLock);
     invalidateTracks_l(streamType);
 }
 
-void AudioFlinger::PlaybackThread::invalidateTracks(std::set<audio_port_handle_t>& portIds) {
+void PlaybackThread::invalidateTracks(std::set<audio_port_handle_t>& portIds) {
     Mutex::Autolock _l(mLock);
     invalidateTracks_l(portIds);
 }
 
-bool AudioFlinger::PlaybackThread::invalidateTracks_l(std::set<audio_port_handle_t>& portIds) {
+bool PlaybackThread::invalidateTracks_l(std::set<audio_port_handle_t>& portIds) {
     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());
@@ -3608,7 +3591,7 @@
 }
 
 // getTrackById_l must be called with holding thread lock
-AudioFlinger::PlaybackThread::Track* AudioFlinger::PlaybackThread::getTrackById_l(
+IAfTrack* PlaybackThread::getTrackById_l(
         audio_port_handle_t trackPortId) {
     for (size_t i = 0; i < mTracks.size(); i++) {
         if (mTracks[i]->portId() == trackPortId) {
@@ -3618,7 +3601,7 @@
     return nullptr;
 }
 
-status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<IAfEffectChain>& chain)
+status_t PlaybackThread::addEffectChain_l(const sp<IAfEffectChain>& chain)
 {
     audio_session_t session = chain->sessionId();
     sp<EffectBufferHalInterface> halInBuffer, halOutBuffer;
@@ -3639,12 +3622,12 @@
             }
             size_t numSamples = mNormalFrameCount
                     * (audio_channel_count_from_out_mask(channelMask) + mHapticChannelCount);
-            status_t result = mAudioFlinger->mEffectsFactoryHal->allocateBuffer(
+            status_t result = mAfThreadCallback->getEffectsFactoryHal()->allocateBuffer(
                     numSamples * sizeof(float),
                     &halInBuffer);
             if (result != OK) return result;
 
-            result = mAudioFlinger->mEffectsFactoryHal->mirrorBuffer(
+            result = mAfThreadCallback->getEffectsFactoryHal()->mirrorBuffer(
                     isSessionSpatialized ? mEffectBuffer : mPostSpatializerBuffer,
                     isSessionSpatialized ? mEffectBufferSize : mPostSpatializerBufferSize,
                     &halOutBuffer);
@@ -3659,10 +3642,10 @@
             // - OUTPUT_STAGE session uses the mEffectBuffer as input buffer and
             // mPostSpatializerBuffer as output buffer
             // - DEVICE session uses the mPostSpatializerBuffer as input and output buffer.
-            status_t result = mAudioFlinger->mEffectsFactoryHal->mirrorBuffer(
+            status_t result = mAfThreadCallback->getEffectsFactoryHal()->mirrorBuffer(
                     mEffectBuffer, mEffectBufferSize, &halInBuffer);
             if (result != OK) return result;
-            result = mAudioFlinger->mEffectsFactoryHal->mirrorBuffer(
+            result = mAfThreadCallback->getEffectsFactoryHal()->mirrorBuffer(
                     mPostSpatializerBuffer, mPostSpatializerBufferSize, &halOutBuffer);
             if (result != OK) return result;
 
@@ -3671,7 +3654,7 @@
             }
         }
     } else {
-        status_t result = mAudioFlinger->mEffectsFactoryHal->mirrorBuffer(
+        status_t result = mAfThreadCallback->getEffectsFactoryHal()->mirrorBuffer(
                 mEffectBufferEnabled ? mEffectBuffer : mSinkBuffer,
                 mEffectBufferEnabled ? mEffectBufferSize : mSinkBufferSize,
                 &halInBuffer);
@@ -3687,7 +3670,8 @@
                 size_t numSamples = mNormalFrameCount
                         * (audio_channel_count_from_out_mask(mMixerChannelMask)
                                                              + mHapticChannelCount);
-                const status_t allocateStatus = mAudioFlinger->mEffectsFactoryHal->allocateBuffer(
+                const status_t allocateStatus =
+                        mAfThreadCallback->getEffectsFactoryHal()->allocateBuffer(
                         numSamples * sizeof(float),
                         &halInBuffer);
                 if (allocateStatus != OK) return allocateStatus;
@@ -3702,7 +3686,7 @@
     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);
@@ -3712,7 +3696,7 @@
         }
 
         // 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);
@@ -3754,7 +3738,7 @@
     return NO_ERROR;
 }
 
-size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<IAfEffectChain>& chain)
+size_t PlaybackThread::removeEffectChain_l(const sp<IAfEffectChain>& chain)
 {
     audio_session_t session = chain->sessionId();
 
@@ -3764,7 +3748,7 @@
         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);
@@ -3774,7 +3758,7 @@
 
             // 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();
@@ -3786,15 +3770,15 @@
     return mEffectChains.size();
 }
 
-status_t AudioFlinger::PlaybackThread::attachAuxEffect(
-        const sp<AudioFlinger::PlaybackThread::Track>& track, int EffectId)
+status_t PlaybackThread::attachAuxEffect(
+        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)
+status_t PlaybackThread::attachAuxEffect_l(
+        const sp<IAfTrack>& track, int EffectId)
 {
     status_t status = NO_ERROR;
 
@@ -3816,22 +3800,22 @@
     return status;
 }
 
-void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
+void 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);
         }
     }
 }
 
-bool AudioFlinger::PlaybackThread::threadLoop()
+bool PlaybackThread::threadLoop()
 NO_THREAD_SAFETY_ANALYSIS  // manual locking of AudioFlinger
 {
     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.
@@ -3879,14 +3863,14 @@
     {
         // Log merge requests are performed during AudioFlinger binder transactions, but
         // that does not cover audio playback. It's requested here for that reason.
-        mAudioFlinger->requestLogMerge();
+        mAfThreadCallback->requestLogMerge();
 
         cpuStats.sample(myName);
 
         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.
         //
@@ -3894,12 +3878,13 @@
         if (isMsdDevice() && outDeviceTypes().count(AUDIO_DEVICE_OUT_BUS) != 0) {
             // Here, we try for the AF lock, but do not block on it as the latency
             // is more informational.
-            if (mAudioFlinger->mLock.tryLock() == NO_ERROR) {
-                std::vector<PatchPanel::SoftwarePatch> swPatches;
+            if (mAfThreadCallback->mutex().tryLock() == NO_ERROR) {
+                std::vector<SoftwarePatch> swPatches;
                 double latencyMs = 0.; // not required; initialized for clang-tidy
                 status_t status = INVALID_OPERATION;
                 audio_patch_handle_t downstreamPatchHandle = AUDIO_PATCH_HANDLE_NONE;
-                if (mAudioFlinger->mPatchPanel.getDownstreamSoftwarePatches(id(), &swPatches) == OK
+                if (mAfThreadCallback->getPatchPanel()->getDownstreamSoftwarePatches(
+                                id(), &swPatches) == OK
                         && swPatches.size() > 0) {
                         status = swPatches[0].getLatencyMs_l(&latencyMs);
                         downstreamPatchHandle = swPatches[0].getPatchHandle();
@@ -3920,7 +3905,7 @@
                     }
                     mDownstreamLatencyStatMs.add(latencyMs);
                 }
-                mAudioFlinger->mLock.unlock();
+                mAfThreadCallback->mutex().unlock();
             }
         } else {
             if (lastDownstreamPatchHandle != AUDIO_PATCH_HANDLE_NONE) {
@@ -4076,7 +4061,7 @@
             setHalLatencyMode_l();
 
             for (const auto &track : mActiveTracks ) {
-                track->updateTeePatches();
+                track->updateTeePatches_l();
             }
 
             // signal actual start of output stream when the render position reported by the kernel
@@ -4104,14 +4089,15 @@
 
                     // 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);
                         }
                     }
                 }
@@ -4280,7 +4266,7 @@
         unlockEffectChains(effectChains);
 
         if (!metadataUpdate.playbackMetadataUpdate.empty()) {
-            mAudioFlinger->mMelReporter->updateMetadataForCsd(id(),
+            mAfThreadCallback->getMelReporter()->updateMetadataForCsd(id(),
                     metadataUpdate.playbackMetadataUpdate);
         }
 
@@ -4481,7 +4467,7 @@
     return false;
 }
 
-void AudioFlinger::PlaybackThread::collectTimestamps_l()
+void PlaybackThread::collectTimestamps_l()
 {
     if (mStandby) {
         mTimestampVerifier.discontinuity(discontinuityForStandbyOrFlush());
@@ -4586,10 +4572,10 @@
                     ? 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);
@@ -4617,7 +4603,7 @@
 }
 
 // removeTracks_l() must be called with ThreadBase::mLock held
-void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove)
+void PlaybackThread::removeTracks_l(const Vector<sp<IAfTrack>>& tracksToRemove)
 NO_THREAD_SAFETY_ANALYSIS  // release and re-acquire mLock
 {
     for (const auto& track : tracksToRemove) {
@@ -4659,7 +4645,7 @@
     }
 }
 
-status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp)
+status_t PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp)
 {
     if (mNormalSink != 0) {
         ExtendedTimestamp ets;
@@ -4688,7 +4674,7 @@
 // All tracks attached to a mixer with flag VOIP_RX are tied to the same
 // stream type STREAM_VOICE_CALL so this will only change the HAL volume once even
 // if more than one track are active
-status_t AudioFlinger::PlaybackThread::handleVoipVolume_l(float *volume)
+status_t PlaybackThread::handleVoipVolume_l(float* volume)
 {
     status_t result = NO_ERROR;
     if ((mOutput->flags & AUDIO_OUTPUT_FLAG_VOIP_RX) != 0) {
@@ -4710,7 +4696,7 @@
     return result;
 }
 
-status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch,
+status_t MixerThread::createAudioPatch_l(const struct audio_patch* patch,
                                                           audio_patch_handle_t *handle)
 {
     status_t status;
@@ -4727,7 +4713,7 @@
     return status;
 }
 
-status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch,
+status_t PlaybackThread::createAudioPatch_l(const struct audio_patch *patch,
                                                           audio_patch_handle_t *handle)
 {
     status_t status = NO_ERROR;
@@ -4811,7 +4797,7 @@
     return status;
 }
 
-status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
+status_t MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
 {
     status_t status;
     if (property_get_bool("af.patch_park", false /* default_value */)) {
@@ -4825,7 +4811,7 @@
     return status;
 }
 
-status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
+status_t PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
 {
     status_t status = NO_ERROR;
 
@@ -4844,19 +4830,19 @@
     return status;
 }
 
-void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track)
+void PlaybackThread::addPatchTrack(const sp<IAfPatchTrack>& track)
 {
     Mutex::Autolock _l(mLock);
     mTracks.add(track);
 }
 
-void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track)
+void PlaybackThread::deletePatchTrack(const sp<IAfPatchTrack>& track)
 {
     Mutex::Autolock _l(mLock);
     destroyTrack_l(track);
 }
 
-void AudioFlinger::PlaybackThread::toAudioPortConfig(struct audio_port_config *config)
+void PlaybackThread::toAudioPortConfig(struct audio_port_config* config)
 {
     ThreadBase::toAudioPortConfig(config);
     config->role = AUDIO_PORT_ROLE_SOURCE;
@@ -4870,9 +4856,16 @@
 
 // ----------------------------------------------------------------------------
 
-AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+/* static */
+sp<IAfPlaybackThread> IAfPlaybackThread::createMixerThread(
+        const sp<IAfThreadCallback>& afThreadCallback, AudioStreamOut* output,
+        audio_io_handle_t id, bool systemReady, type_t type, audio_config_base_t* mixerConfig) {
+    return sp<MixerThread>::make(afThreadCallback, output, id, systemReady, type, mixerConfig);
+}
+
+MixerThread::MixerThread(const sp<IAfThreadCallback>& afThreadCallback, AudioStreamOut* output,
         audio_io_handle_t id, bool systemReady, type_t type, audio_config_base_t *mixerConfig)
-    :   PlaybackThread(audioFlinger, output, id, type, systemReady, mixerConfig),
+    :   PlaybackThread(afThreadCallback, output, id, type, systemReady, mixerConfig),
         // mAudioMixer below
         // mFastMixer below
         mBluetoothLatencyModesEnabled(false),
@@ -4882,7 +4875,7 @@
         // mPipeSink below
         // mNormalSink below
 {
-    setMasterBalance(audioFlinger->getMasterBalance_l());
+    setMasterBalance(afThreadCallback->getMasterBalance_l());
     ALOGV("MixerThread() id=%d type=%d", id, type);
     ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%#x, mFrameSize=%zu, "
             "mFrameCount=%zu, mNormalFrameCount=%zu",
@@ -5009,7 +5002,7 @@
         state->mColdFutexAddr = &mFastMixerFutex;
         state->mColdGen++;
         state->mDumpState = &mFastMixerDumpState;
-        mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer");
+        mFastMixerNBLogWriter = afThreadCallback->newWriter_l(kFastMixerLogSize, "FastMixer");
         state->mNBLogWriter = mFastMixerNBLogWriter.get();
         sq->end();
         sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
@@ -5055,7 +5048,7 @@
     }
 }
 
-AudioFlinger::MixerThread::~MixerThread()
+MixerThread::~MixerThread()
 {
     if (mFastMixer != 0) {
         FastMixerStateQueue *sq = mFastMixer->sq();
@@ -5088,11 +5081,11 @@
         }
 #endif
     }
-    mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter);
+    mAfThreadCallback->unregisterWriter(mFastMixerNBLogWriter);
     delete mAudioMixer;
 }
 
-void AudioFlinger::MixerThread::onFirstRef() {
+void MixerThread::onFirstRef() {
     PlaybackThread::onFirstRef();
 
     Mutex::Autolock _l(mLock);
@@ -5108,7 +5101,7 @@
     }
 }
 
-uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const
+uint32_t MixerThread::correctLatency_l(uint32_t latency) const
 {
     if (mFastMixer != 0) {
         MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
@@ -5117,7 +5110,7 @@
     return latency;
 }
 
-ssize_t AudioFlinger::MixerThread::threadLoop_write()
+ssize_t MixerThread::threadLoop_write()
 {
     // FIXME we should only do one push per cycle; confirm this is true
     // Start the fast mixer if it's not already running
@@ -5145,7 +5138,7 @@
             }
             state->mCommand = FastMixerState::MIX_WRITE;
 #ifdef FAST_THREAD_STATISTICS
-            mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
+            mFastMixerDumpState.increaseSamplingN(mAfThreadCallback->isLowRamDevice() ?
                 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN);
 #endif
             sq->end();
@@ -5160,7 +5153,7 @@
     return PlaybackThread::threadLoop_write();
 }
 
-void AudioFlinger::MixerThread::threadLoop_standby()
+void MixerThread::threadLoop_standby()
 {
     // Idle the fast mixer if it's currently running
     if (mFastMixer != 0) {
@@ -5198,24 +5191,24 @@
     PlaybackThread::threadLoop_standby();
 }
 
-bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l()
+bool PlaybackThread::waitingAsyncCallback_l()
 {
     return false;
 }
 
-bool AudioFlinger::PlaybackThread::shouldStandby_l()
+bool PlaybackThread::shouldStandby_l()
 {
     return !mStandby;
 }
 
-bool AudioFlinger::PlaybackThread::waitingAsyncCallback()
+bool PlaybackThread::waitingAsyncCallback()
 {
     Mutex::Autolock _l(mLock);
     return waitingAsyncCallback_l();
 }
 
 // shared by MIXER and DIRECT, overridden by DUPLICATING
-void AudioFlinger::PlaybackThread::threadLoop_standby()
+void PlaybackThread::threadLoop_standby()
 {
     ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended);
     mOutput->standby();
@@ -5231,20 +5224,20 @@
     setHalLatencyMode_l();
 }
 
-void AudioFlinger::PlaybackThread::onAddNewTrack_l()
+void PlaybackThread::onAddNewTrack_l()
 {
     ALOGV("signal playback thread");
     broadcast_l();
 }
 
-void AudioFlinger::PlaybackThread::onAsyncError()
+void PlaybackThread::onAsyncError()
 {
     for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) {
         invalidateTracks((audio_stream_type_t)i);
     }
 }
 
-void AudioFlinger::MixerThread::threadLoop_mix()
+void MixerThread::threadLoop_mix()
 {
     // mix buffers...
     mAudioMixer->process();
@@ -5262,7 +5255,7 @@
 
 }
 
-void AudioFlinger::MixerThread::threadLoop_sleepTime()
+void MixerThread::threadLoop_sleepTime()
 {
     // If no tracks are ready, sleep once for the duration of an output
     // buffer size, then write 0s to the output
@@ -5316,8 +5309,8 @@
 }
 
 // prepareTracks_l() must be called with ThreadBase::mLock held
-AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
-        Vector< sp<Track> > *tracksToRemove)
+PlaybackThread::mixer_state MixerThread::prepareTracks_l(
+        Vector<sp<IAfTrack>>* tracksToRemove)
 {
     // clean up deleted track ids in AudioMixer before allocating new tracks
     (void)mTracks.processDeletedTrackIds([this](int trackId) {
@@ -5393,23 +5386,23 @@
         // 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()) {
@@ -5427,7 +5420,7 @@
             // 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];
@@ -5437,13 +5430,13 @@
             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;
@@ -5453,30 +5446,30 @@
                 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
@@ -5490,7 +5483,7 @@
                         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;
@@ -5501,10 +5494,10 @@
                     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
@@ -5521,7 +5514,7 @@
                     }
                 }
                 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
@@ -5531,20 +5524,20 @@
                 }
                 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();
@@ -5553,7 +5546,7 @@
                     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;
@@ -5568,12 +5561,12 @@
                 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));
 
-                track->processMuteEvent_l(mAudioFlinger->getOrCreateAudioManager(),
+                track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
                     /*muteState=*/{masterVolume == 0.f,
                                    mStreamTypes[track->streamType()].volume == 0.f,
                                    mStreamTypes[track->streamType()].mute,
@@ -5604,13 +5597,13 @@
                     // 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();
@@ -5632,14 +5625,14 @@
         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;
@@ -5652,8 +5645,8 @@
         // 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);
@@ -5703,11 +5696,11 @@
 
 
             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;
@@ -5728,9 +5721,9 @@
             // 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;
@@ -5756,7 +5749,7 @@
                     vrf = GAIN_FLOAT_UNITY;
                 }
 
-                track->processMuteEvent_l(mAudioFlinger->getOrCreateAudioManager(),
+                track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
                     /*muteState=*/{masterVolume == 0.f,
                                    mStreamTypes[track->streamType()].volume == 0.f,
                                    mStreamTypes[track->streamType()].mute,
@@ -5792,18 +5785,18 @@
                 // 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);
@@ -5911,13 +5904,14 @@
                 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
@@ -5969,7 +5963,7 @@
             } 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);
@@ -6045,7 +6039,7 @@
         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();
     }
@@ -6111,7 +6105,7 @@
 }
 
 // trackCountForUid_l() must be called with ThreadBase::mLock held
-uint32_t AudioFlinger::PlaybackThread::trackCountForUid_l(uid_t uid) const
+uint32_t PlaybackThread::trackCountForUid_l(uid_t uid) const
 {
     uint32_t trackCount = 0;
     for (size_t i = 0; i < mTracks.size() ; i++) {
@@ -6122,7 +6116,7 @@
     return trackCount;
 }
 
-bool AudioFlinger::PlaybackThread::IsTimestampAdvancing::check(AudioStreamOut * output)
+bool PlaybackThread::IsTimestampAdvancing::check(AudioStreamOut* output)
 {
     // Check the timestamp to see if it's advancing once every 150ms. If we check too frequently, we
     // could falsely detect that the frame position has stalled due to underrun because we haven't
@@ -6146,7 +6140,7 @@
     return mLatchedValue;
 }
 
-void AudioFlinger::PlaybackThread::IsTimestampAdvancing::clear()
+void PlaybackThread::IsTimestampAdvancing::clear()
 {
     mLatchedValue = true;
     mPreviousPosition = 0;
@@ -6154,7 +6148,7 @@
 }
 
 // isTrackAllowed_l() must be called with ThreadBase::mLock held
-bool AudioFlinger::MixerThread::isTrackAllowed_l(
+bool MixerThread::isTrackAllowed_l(
         audio_channel_mask_t channelMask, audio_format_t format,
         audio_session_t sessionId, uid_t uid) const
 {
@@ -6174,7 +6168,7 @@
 }
 
 // checkForNewParameter_l() must be called with ThreadBase::mLock held
-bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair,
+bool MixerThread::checkForNewParameter_l(const String8& keyValuePair,
                                                        status_t& status)
 {
     bool reconfig = false;
@@ -6188,7 +6182,7 @@
         reconfig = true;
     }
     if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
-        if (!isValidPcmSinkFormat((audio_format_t) value)) {
+        if (!AudioFlinger::isValidPcmSinkFormat(static_cast<audio_format_t>(value))) {
             status = BAD_VALUE;
         } else {
             // no need to save value, since it's constant
@@ -6196,7 +6190,7 @@
         }
     }
     if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
-        if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) {
+        if (!AudioFlinger::isValidPcmSinkChannelMask(static_cast<audio_channel_mask_t>(value))) {
             status = BAD_VALUE;
         } else {
             // no need to save value, since it's constant
@@ -6237,14 +6231,14 @@
                 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);
         }
@@ -6254,7 +6248,7 @@
 }
 
 
-void AudioFlinger::MixerThread::dumpInternals_l(int fd, const Vector<String16>& args)
+void MixerThread::dumpInternals_l(int fd, const Vector<String16>& args)
 {
     PlaybackThread::dumpInternals_l(fd, args);
     dprintf(fd, "  Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs);
@@ -6301,17 +6295,17 @@
      dprintf(fd, "Supported latency modes: %s\n", toString(mSupportedLatencyModes).c_str());
 }
 
-uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
+uint32_t MixerThread::idleSleepTimeUs() const
 {
     return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2;
 }
 
-uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const
+uint32_t MixerThread::suspendSleepTimeUs() const
 {
     return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000);
 }
 
-void AudioFlinger::MixerThread::cacheParameters_l()
+void MixerThread::cacheParameters_l()
 {
     PlaybackThread::cacheParameters_l();
 
@@ -6322,11 +6316,11 @@
     maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15;
 }
 
-void AudioFlinger::MixerThread::onHalLatencyModesChanged_l() {
-    mAudioFlinger->onSupportedLatencyModesChanged(mId, mSupportedLatencyModes);
+void MixerThread::onHalLatencyModesChanged_l() {
+    mAfThreadCallback->onSupportedLatencyModesChanged(mId, mSupportedLatencyModes);
 }
 
-void AudioFlinger::MixerThread::setHalLatencyMode_l() {
+void MixerThread::setHalLatencyMode_l() {
     // Only handle latency mode if:
     // - mBluetoothLatencyModesEnabled is true
     // - the HAL supports latency modes
@@ -6368,7 +6362,7 @@
     }
 }
 
-void AudioFlinger::MixerThread::updateHalSupportedLatencyModes_l() {
+void MixerThread::updateHalSupportedLatencyModes_l() {
 
     if (mOutput == nullptr || mOutput->stream == nullptr) {
         return;
@@ -6386,7 +6380,7 @@
     }
 }
 
-status_t AudioFlinger::MixerThread::getSupportedLatencyModes(
+status_t MixerThread::getSupportedLatencyModes(
         std::vector<audio_latency_mode_t>* modes) {
     if (modes == nullptr) {
         return BAD_VALUE;
@@ -6396,7 +6390,7 @@
     return NO_ERROR;
 }
 
-void AudioFlinger::MixerThread::onRecommendedLatencyModeChanged(
+void MixerThread::onRecommendedLatencyModeChanged(
         std::vector<audio_latency_mode_t> modes) {
     Mutex::Autolock _l(mLock);
     if (modes != mSupportedLatencyModes) {
@@ -6407,7 +6401,7 @@
     }
 }
 
-status_t AudioFlinger::MixerThread::setBluetoothVariableLatencyEnabled(bool enabled) {
+status_t MixerThread::setBluetoothVariableLatencyEnabled(bool enabled) {
     if (mOutput == nullptr || mOutput->audioHwDev == nullptr
             || !mOutput->audioHwDev->supportsBluetoothVariableLatency()) {
         return INVALID_OPERATION;
@@ -6418,27 +6412,36 @@
 
 // ----------------------------------------------------------------------------
 
-AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
+/* static */
+sp<IAfPlaybackThread> IAfPlaybackThread::createDirectOutputThread(
+        const sp<IAfThreadCallback>& afThreadCallback,
+        AudioStreamOut* output, audio_io_handle_t id, bool systemReady,
+        const audio_offload_info_t& offloadInfo) {
+    return sp<DirectOutputThread>::make(
+            afThreadCallback, output, id, systemReady, offloadInfo);
+}
+
+DirectOutputThread::DirectOutputThread(const sp<IAfThreadCallback>& afThreadCallback,
         AudioStreamOut* output, audio_io_handle_t id, ThreadBase::type_t type, bool systemReady,
         const audio_offload_info_t& offloadInfo)
-    :   PlaybackThread(audioFlinger, output, id, type, systemReady)
+    :   PlaybackThread(afThreadCallback, output, id, type, systemReady)
     , mOffloadInfo(offloadInfo)
 {
-    setMasterBalance(audioFlinger->getMasterBalance_l());
+    setMasterBalance(afThreadCallback->getMasterBalance_l());
 }
 
-AudioFlinger::DirectOutputThread::~DirectOutputThread()
+DirectOutputThread::~DirectOutputThread()
 {
 }
 
-void AudioFlinger::DirectOutputThread::dumpInternals_l(int fd, const Vector<String16>& args)
+void DirectOutputThread::dumpInternals_l(int fd, const Vector<String16>& args)
 {
     PlaybackThread::dumpInternals_l(fd, args);
     dprintf(fd, "  Master balance: %f  Left: %f  Right: %f\n",
             mMasterBalance.load(), mMasterBalanceLeft, mMasterBalanceRight);
 }
 
-void AudioFlinger::DirectOutputThread::setMasterBalance(float balance)
+void DirectOutputThread::setMasterBalance(float balance)
 {
     Mutex::Autolock _l(mLock);
     if (mMasterBalance != balance) {
@@ -6448,12 +6451,12 @@
     }
 }
 
-void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack)
+void 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];
@@ -6488,14 +6491,14 @@
         }
         left *= v;
         right *= v;
-        if (mAudioFlinger->getMode() != AUDIO_MODE_IN_COMMUNICATION
+        if (mAfThreadCallback->getMode() != AUDIO_MODE_IN_COMMUNICATION
                 || audio_channel_count_from_out_mask(mChannelMask) > 1) {
             left *= mMasterBalanceLeft; // DirectOutputThread balance applied as track volume
             right *= mMasterBalanceRight;
         }
     }
 
-    track->processMuteEvent_l(mAudioFlinger->getOrCreateAudioManager(),
+    track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
         /*muteState=*/{mMasterMute,
                        mStreamTypes[track->streamType()].volume == 0.f,
                        mStreamTypes[track->streamType()].mute,
@@ -6527,10 +6530,10 @@
     }
 }
 
-void AudioFlinger::DirectOutputThread::onAddNewTrack_l()
+void 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) {
@@ -6552,8 +6555,8 @@
     PlaybackThread::onAddNewTrack_l();
 }
 
-AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l(
-    Vector< sp<Track> > *tracksToRemove
+PlaybackThread::mixer_state DirectOutputThread::prepareTracks_l(
+    Vector<sp<IAfTrack>>* tracksToRemove
 )
 {
     size_t count = mActiveTracks.size();
@@ -6562,14 +6565,14 @@
     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
@@ -6577,7 +6580,7 @@
         // 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()) {
@@ -6613,8 +6616,8 @@
         // 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.
@@ -6626,7 +6629,7 @@
         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;
@@ -6644,8 +6647,8 @@
         {
             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;
@@ -6658,7 +6661,7 @@
             // 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
@@ -6670,7 +6673,7 @@
                 mPreviousTrack = track;
 
                 // reset retry count
-                track->mRetryCount = targetRetryCount;
+                track->retryCount() = targetRetryCount;
                 mActiveTrack = t;
                 mixerStatus = MIXER_TRACKS_READY;
                 if (mHwPaused) {
@@ -6685,7 +6688,7 @@
                 mEffectChains[0]->clearInputBuffer();
             }
             if (track->isStopping_1()) {
-                track->mState = TrackBase::STOPPING_2;
+                track->setState(IAfTrackBase::STOPPING_2);
                 if (last && mHwPaused) {
                      doHwResume = true;
                      mHwPaused = false;
@@ -6703,7 +6706,7 @@
                         mOutput->presentationComplete();
                     }
                     if (track->isStopping_2()) {
-                        track->mState = TrackBase::STOPPED;
+                        track->setState(IAfTrackBase::STOPPED);
                     }
                     if (track->isStopped()) {
                         track->reset();
@@ -6716,9 +6719,9 @@
                 // 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);
@@ -6775,7 +6778,7 @@
     return mixerStatus;
 }
 
-void AudioFlinger::DirectOutputThread::threadLoop_mix()
+void DirectOutputThread::threadLoop_mix()
 {
     size_t frameCount = mFrameCount;
     int8_t *curBuf = (int8_t *)mSinkBuffer;
@@ -6802,7 +6805,7 @@
     mActiveTrack.clear();
 }
 
-void AudioFlinger::DirectOutputThread::threadLoop_sleepTime()
+void DirectOutputThread::threadLoop_sleepTime()
 {
     // do not write to HAL when paused
     if (mHwPaused || (usesHwAvSync() && mStandby)) {
@@ -6818,7 +6821,7 @@
     // linear or proportional PCM direct tracks in underrun.
 }
 
-void AudioFlinger::DirectOutputThread::threadLoop_exit()
+void DirectOutputThread::threadLoop_exit()
 {
     {
         Mutex::Autolock _l(mLock);
@@ -6836,7 +6839,7 @@
 }
 
 // must be called with thread mutex locked
-bool AudioFlinger::DirectOutputThread::shouldStandby_l()
+bool DirectOutputThread::shouldStandby_l()
 {
     bool trackPaused = false;
     bool trackStopped = false;
@@ -6846,14 +6849,14 @@
     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));
 }
 
 // checkForNewParameter_l() must be called with ThreadBase::mLock held
-bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair,
+bool DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair,
                                                               status_t& status)
 {
     bool reconfig = false;
@@ -6895,7 +6898,7 @@
     return reconfig;
 }
 
-uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const
+uint32_t DirectOutputThread::activeSleepTimeUs() const
 {
     uint32_t time;
     if (audio_has_proportional_frames(mFormat)) {
@@ -6906,7 +6909,7 @@
     return time;
 }
 
-uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const
+uint32_t DirectOutputThread::idleSleepTimeUs() const
 {
     uint32_t time;
     if (audio_has_proportional_frames(mFormat)) {
@@ -6917,7 +6920,7 @@
     return time;
 }
 
-uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const
+uint32_t DirectOutputThread::suspendSleepTimeUs() const
 {
     uint32_t time;
     if (audio_has_proportional_frames(mFormat)) {
@@ -6928,7 +6931,7 @@
     return time;
 }
 
-void AudioFlinger::DirectOutputThread::cacheParameters_l()
+void DirectOutputThread::cacheParameters_l()
 {
     PlaybackThread::cacheParameters_l();
 
@@ -6944,7 +6947,7 @@
     }
 }
 
-void AudioFlinger::DirectOutputThread::flushHw_l()
+void DirectOutputThread::flushHw_l()
 {
     PlaybackThread::flushHw_l();
     mOutput->flush();
@@ -6955,7 +6958,7 @@
     mMonotonicFrameCounter.onFlush();
 }
 
-int64_t AudioFlinger::DirectOutputThread::computeWaitTimeNs_l() const {
+int64_t DirectOutputThread::computeWaitTimeNs_l() const {
     // If a VolumeShaper is active, we must wake up periodically to update volume.
     const int64_t NS_PER_MS = 1000000;
     return mVolumeShaperActive ?
@@ -6964,8 +6967,8 @@
 
 // ----------------------------------------------------------------------------
 
-AudioFlinger::AsyncCallbackThread::AsyncCallbackThread(
-        const wp<AudioFlinger::PlaybackThread>& playbackThread)
+AsyncCallbackThread::AsyncCallbackThread(
+        const wp<PlaybackThread>& playbackThread)
     :   Thread(false /*canCallJava*/),
         mPlaybackThread(playbackThread),
         mWriteAckSequence(0),
@@ -6974,16 +6977,12 @@
 {
 }
 
-AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread()
-{
-}
-
-void AudioFlinger::AsyncCallbackThread::onFirstRef()
+void AsyncCallbackThread::onFirstRef()
 {
     run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO);
 }
 
-bool AudioFlinger::AsyncCallbackThread::threadLoop()
+bool AsyncCallbackThread::threadLoop()
 {
     while (!exitPending()) {
         uint32_t writeAckSequence;
@@ -7012,7 +7011,7 @@
             mAsyncError = false;
         }
         {
-            sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote();
+            const sp<PlaybackThread> playbackThread = mPlaybackThread.promote();
             if (playbackThread != 0) {
                 if (writeAckSequence & 1) {
                     playbackThread->resetWriteBlocked(writeAckSequence >> 1);
@@ -7029,7 +7028,7 @@
     return false;
 }
 
-void AudioFlinger::AsyncCallbackThread::exit()
+void AsyncCallbackThread::exit()
 {
     ALOGV("AsyncCallbackThread::exit");
     Mutex::Autolock _l(mLock);
@@ -7037,14 +7036,14 @@
     mWaitWorkCV.broadcast();
 }
 
-void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence)
+void AsyncCallbackThread::setWriteBlocked(uint32_t sequence)
 {
     Mutex::Autolock _l(mLock);
     // bit 0 is cleared
     mWriteAckSequence = sequence << 1;
 }
 
-void AudioFlinger::AsyncCallbackThread::resetWriteBlocked()
+void AsyncCallbackThread::resetWriteBlocked()
 {
     Mutex::Autolock _l(mLock);
     // ignore unexpected callbacks
@@ -7054,14 +7053,14 @@
     }
 }
 
-void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence)
+void AsyncCallbackThread::setDraining(uint32_t sequence)
 {
     Mutex::Autolock _l(mLock);
     // bit 0 is cleared
     mDrainSequence = sequence << 1;
 }
 
-void AudioFlinger::AsyncCallbackThread::resetDraining()
+void AsyncCallbackThread::resetDraining()
 {
     Mutex::Autolock _l(mLock);
     // ignore unexpected callbacks
@@ -7071,7 +7070,7 @@
     }
 }
 
-void AudioFlinger::AsyncCallbackThread::setAsyncError()
+void AsyncCallbackThread::setAsyncError()
 {
     Mutex::Autolock _l(mLock);
     mAsyncError = true;
@@ -7080,10 +7079,19 @@
 
 
 // ----------------------------------------------------------------------------
-AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger,
+
+/* static */
+sp<IAfPlaybackThread> IAfPlaybackThread::createOffloadThread(
+        const sp<IAfThreadCallback>& afThreadCallback,
+        AudioStreamOut* output, audio_io_handle_t id, bool systemReady,
+        const audio_offload_info_t& offloadInfo) {
+    return sp<OffloadThread>::make(afThreadCallback, output, id, systemReady, offloadInfo);
+}
+
+OffloadThread::OffloadThread(const sp<IAfThreadCallback>& afThreadCallback,
         AudioStreamOut* output, audio_io_handle_t id, bool systemReady,
         const audio_offload_info_t& offloadInfo)
-    :   DirectOutputThread(audioFlinger, output, id, OFFLOAD, systemReady, offloadInfo),
+    :   DirectOutputThread(afThreadCallback, output, id, OFFLOAD, systemReady, offloadInfo),
         mPausedWriteLength(0), mPausedBytesRemaining(0), mKeepWakeLock(true)
 {
     //FIXME: mStandby should be set to true by ThreadBase constructo
@@ -7091,7 +7099,7 @@
     mKeepWakeLock = property_get_bool("ro.audio.offload_wakelock", true /* default_value */);
 }
 
-void AudioFlinger::OffloadThread::threadLoop_exit()
+void OffloadThread::threadLoop_exit()
 {
     if (mFlushPending || mHwPaused) {
         // If a flush is pending or track was paused, just discard buffered data
@@ -7107,8 +7115,8 @@
     PlaybackThread::threadLoop_exit();
 }
 
-AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l(
-    Vector< sp<Track> > *tracksToRemove
+PlaybackThread::mixer_state OffloadThread::prepareTracks_l(
+    Vector<sp<IAfTrack>>* tracksToRemove
 )
 {
     size_t count = mActiveTracks.size();
@@ -7120,8 +7128,8 @@
     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
@@ -7129,7 +7137,7 @@
         // 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()) {
@@ -7138,7 +7146,7 @@
             continue;
         }
 
-        if (track->mState == TrackBase::IDLE) {
+        if (track->state() == IAfTrackBase::IDLE) {
             ALOGW("An idle track shouldn't be in active list");
             continue;
         }
@@ -7170,9 +7178,9 @@
             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) {
@@ -7204,8 +7212,8 @@
         }  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;
@@ -7213,7 +7221,7 @@
             }
 
             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
@@ -7239,9 +7247,9 @@
                 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;
@@ -7249,7 +7257,7 @@
         } 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.
@@ -7257,7 +7265,8 @@
                         // 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
@@ -7277,13 +7286,13 @@
                         }
                     }
                 } 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();
@@ -7303,9 +7312,9 @@
                 // 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());
@@ -7349,7 +7358,7 @@
 }
 
 // must be called with thread mutex locked
-bool AudioFlinger::OffloadThread::waitingAsyncCallback_l()
+bool OffloadThread::waitingAsyncCallback_l()
 {
     ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d",
           mWriteAckSequence, mDrainSequence);
@@ -7359,13 +7368,13 @@
     return false;
 }
 
-bool AudioFlinger::OffloadThread::waitingAsyncCallback()
+bool OffloadThread::waitingAsyncCallback()
 {
     Mutex::Autolock _l(mLock);
     return waitingAsyncCallback_l();
 }
 
-void AudioFlinger::OffloadThread::flushHw_l()
+void OffloadThread::flushHw_l()
 {
     DirectOutputThread::flushHw_l();
     // Flush anything still waiting in the mixbuffer
@@ -7386,7 +7395,7 @@
     }
 }
 
-void AudioFlinger::OffloadThread::invalidateTracks(audio_stream_type_t streamType)
+void OffloadThread::invalidateTracks(audio_stream_type_t streamType)
 {
     Mutex::Autolock _l(mLock);
     if (PlaybackThread::invalidateTracks_l(streamType)) {
@@ -7394,7 +7403,7 @@
     }
 }
 
-void AudioFlinger::OffloadThread::invalidateTracks(std::set<audio_port_handle_t>& portIds) {
+void OffloadThread::invalidateTracks(std::set<audio_port_handle_t>& portIds) {
     Mutex::Autolock _l(mLock);
     if (PlaybackThread::invalidateTracks_l(portIds)) {
         mFlushPending = true;
@@ -7403,23 +7412,30 @@
 
 // ----------------------------------------------------------------------------
 
-AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
-        AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady)
-    :   MixerThread(audioFlinger, mainThread->getOutput(), id,
+/* static */
+sp<IAfDuplicatingThread> IAfDuplicatingThread::create(
+        const sp<IAfThreadCallback>& afThreadCallback,
+        IAfPlaybackThread* mainThread, audio_io_handle_t id, bool systemReady) {
+    return sp<DuplicatingThread>::make(afThreadCallback, mainThread, id, systemReady);
+}
+
+DuplicatingThread::DuplicatingThread(const sp<IAfThreadCallback>& afThreadCallback,
+       IAfPlaybackThread* mainThread, audio_io_handle_t id, bool systemReady)
+    :   MixerThread(afThreadCallback, mainThread->getOutput(), id,
                     systemReady, DUPLICATING),
         mWaitTimeMs(UINT_MAX)
 {
     addOutputTrack(mainThread);
 }
 
-AudioFlinger::DuplicatingThread::~DuplicatingThread()
+DuplicatingThread::~DuplicatingThread()
 {
     for (size_t i = 0; i < mOutputTracks.size(); i++) {
         mOutputTracks[i]->destroy();
     }
 }
 
-void AudioFlinger::DuplicatingThread::threadLoop_mix()
+void DuplicatingThread::threadLoop_mix()
 {
     // mix buffers...
     if (outputsReady()) {
@@ -7437,7 +7453,7 @@
     mStandbyTimeNs = systemTime() + mStandbyDelayNs;
 }
 
-void AudioFlinger::DuplicatingThread::threadLoop_sleepTime()
+void DuplicatingThread::threadLoop_sleepTime()
 {
     if (mSleepTimeUs == 0) {
         if (mMixerStatus == MIXER_TRACKS_ENABLED) {
@@ -7457,7 +7473,7 @@
     }
 }
 
-ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
+ssize_t DuplicatingThread::threadLoop_write()
 {
     for (size_t i = 0; i < outputTracks.size(); i++) {
         const ssize_t actualWritten = outputTracks[i]->write(mSinkBuffer, writeFrames);
@@ -7485,7 +7501,7 @@
     return (ssize_t)mSinkBufferSize;
 }
 
-void AudioFlinger::DuplicatingThread::threadLoop_standby()
+void DuplicatingThread::threadLoop_standby()
 {
     // DuplicatingThread implements standby by stopping all tracks
     for (size_t i = 0; i < outputTracks.size(); i++) {
@@ -7493,7 +7509,7 @@
     }
 }
 
-void AudioFlinger::DuplicatingThread::dumpInternals_l(int fd, const Vector<String16>& args)
+void DuplicatingThread::dumpInternals_l(int fd, const Vector<String16>& args)
 {
     MixerThread::dumpInternals_l(fd, args);
 
@@ -7503,7 +7519,7 @@
     if (numTracks > 0) {
         ss << ":";
         for (const auto &track : mOutputTracks) {
-            const sp<ThreadBase> thread = track->thread().promote();
+            const auto thread = track->thread().promote();
             ss << " (" << track->id() << " : ";
             if (thread.get() != nullptr) {
                 ss << thread.get() << ", " << thread->id();
@@ -7518,17 +7534,17 @@
     write(fd, result.c_str(), result.size());
 }
 
-void AudioFlinger::DuplicatingThread::saveOutputTracks()
+void DuplicatingThread::saveOutputTracks()
 {
     outputTracks = mOutputTracks;
 }
 
-void AudioFlinger::DuplicatingThread::clearOutputTracks()
+void DuplicatingThread::clearOutputTracks()
 {
     outputTracks.clear();
 }
 
-void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
+void DuplicatingThread::addOutputTrack(IAfPlaybackThread* thread)
 {
     Mutex::Autolock _l(mLock);
     // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass.
@@ -7547,7 +7563,7 @@
     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,
@@ -7565,7 +7581,7 @@
     updateWaitTime_l();
 }
 
-void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
+void DuplicatingThread::removeOutputTrack(IAfPlaybackThread* thread)
 {
     Mutex::Autolock _l(mLock);
     for (size_t i = 0; i < mOutputTracks.size(); i++) {
@@ -7583,11 +7599,11 @@
 }
 
 // caller must hold mLock
-void AudioFlinger::DuplicatingThread::updateWaitTime_l()
+void DuplicatingThread::updateWaitTime_l()
 {
     mWaitTimeMs = UINT_MAX;
     for (size_t i = 0; i < mOutputTracks.size(); i++) {
-        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
+        const auto strong = mOutputTracks[i]->thread().promote();
         if (strong != 0) {
             uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
             if (waitTimeMs < mWaitTimeMs) {
@@ -7597,18 +7613,18 @@
     }
 }
 
-bool AudioFlinger::DuplicatingThread::outputsReady()
+bool DuplicatingThread::outputsReady()
 {
     for (size_t i = 0; i < outputTracks.size(); i++) {
-        sp<ThreadBase> thread = outputTracks[i]->thread().promote();
+        const auto thread = outputTracks[i]->thread().promote();
         if (thread == 0) {
             ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p",
                     outputTracks[i].get());
             return false;
         }
-        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+        IAfPlaybackThread* const playbackThread = thread->asIAfPlaybackThread().get();
         // see note at standby() declaration
-        if (playbackThread->standby() && !playbackThread->isSuspended()) {
+        if (playbackThread->inStandby() && !playbackThread->isSuspended()) {
             ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(),
                     thread.get());
             return false;
@@ -7617,7 +7633,7 @@
     return true;
 }
 
-void AudioFlinger::DuplicatingThread::sendMetadataToBackend_l(
+void DuplicatingThread::sendMetadataToBackend_l(
         const StreamOutHalInterface::SourceMetadata& metadata)
 {
     for (auto& outputTrack : outputTracks) { // not mOutputTracks
@@ -7625,12 +7641,12 @@
     }
 }
 
-uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const
+uint32_t DuplicatingThread::activeSleepTimeUs() const
 {
     return (mWaitTimeMs * 1000) / 2;
 }
 
-void AudioFlinger::DuplicatingThread::cacheParameters_l()
+void DuplicatingThread::cacheParameters_l()
 {
     // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first
     updateWaitTime_l();
@@ -7640,16 +7656,26 @@
 
 // ----------------------------------------------------------------------------
 
-AudioFlinger::SpatializerThread::SpatializerThread(const sp<AudioFlinger>& audioFlinger,
+/* static */
+sp<IAfPlaybackThread> IAfPlaybackThread::createSpatializerThread(
+        const sp<IAfThreadCallback>& afThreadCallback,
+        AudioStreamOut* output,
+        audio_io_handle_t id,
+        bool systemReady,
+        audio_config_base_t* mixerConfig) {
+    return sp<SpatializerThread>::make(afThreadCallback, output, id, systemReady, mixerConfig);
+}
+
+SpatializerThread::SpatializerThread(const sp<IAfThreadCallback>& afThreadCallback,
                                                              AudioStreamOut* output,
                                                              audio_io_handle_t id,
                                                              bool systemReady,
                                                              audio_config_base_t *mixerConfig)
-    : MixerThread(audioFlinger, output, id, systemReady, SPATIALIZER, mixerConfig)
+    : MixerThread(afThreadCallback, output, id, systemReady, SPATIALIZER, mixerConfig)
 {
 }
 
-void AudioFlinger::SpatializerThread::onFirstRef() {
+void SpatializerThread::onFirstRef() {
     MixerThread::onFirstRef();
 
     const pid_t tid = getTid();
@@ -7664,7 +7690,7 @@
     }
 }
 
-void AudioFlinger::SpatializerThread::setHalLatencyMode_l() {
+void SpatializerThread::setHalLatencyMode_l() {
     // if mSupportedLatencyModes is empty, the HAL stream does not support
     // latency mode control and we can exit.
     if (mSupportedLatencyModes.empty()) {
@@ -7702,7 +7728,7 @@
     }
 }
 
-status_t AudioFlinger::SpatializerThread::setRequestedLatencyMode(audio_latency_mode_t mode) {
+status_t SpatializerThread::setRequestedLatencyMode(audio_latency_mode_t mode) {
     if (mode != AUDIO_LATENCY_MODE_LOW && mode != AUDIO_LATENCY_MODE_FREE) {
         return BAD_VALUE;
     }
@@ -7711,7 +7737,7 @@
     return NO_ERROR;
 }
 
-void AudioFlinger::SpatializerThread::checkOutputStageEffects()
+void SpatializerThread::checkOutputStageEffects()
 {
     bool hasVirtualizer = false;
     bool hasDownMixer = false;
@@ -7736,7 +7762,7 @@
         finalDownMixer.clear();
     } else if (!hasDownMixer) {
         std::vector<effect_descriptor_t> descriptors;
-        status_t status = mAudioFlinger->mEffectsFactoryHal->getDescriptors(
+        status_t status = mAfThreadCallback->getEffectsFactoryHal()->getDescriptors(
                                                         EFFECT_UIID_DOWNMIX, &descriptors);
         if (status != NO_ERROR) {
             return;
@@ -7767,12 +7793,19 @@
 //      Record
 // ----------------------------------------------------------------------------
 
-AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
+sp<IAfRecordThread> IAfRecordThread::create(const sp<IAfThreadCallback>& afThreadCallback,
+        AudioStreamIn* input,
+        audio_io_handle_t id,
+        bool systemReady) {
+    return sp<RecordThread>::make(afThreadCallback, input, id, systemReady);
+}
+
+RecordThread::RecordThread(const sp<IAfThreadCallback>& afThreadCallback,
                                          AudioStreamIn *input,
                                          audio_io_handle_t id,
                                          bool systemReady
                                          ) :
-    ThreadBase(audioFlinger, id, RECORD, systemReady, false /* isOut */),
+    ThreadBase(afThreadCallback, id, RECORD, systemReady, false /* isOut */),
     mInput(input),
     mSource(mInput),
     mActiveTracks(&this->mLocalLog),
@@ -7793,7 +7826,7 @@
     , mBtNrecSuspended(false)
 {
     snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id);
-    mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName);
+    mNBLogWriter = afThreadCallback->newWriter_l(kLogSize, mThreadName);
 
     if (mInput->audioHwDev != nullptr) {
         mIsMsdDevice = strcmp(
@@ -7901,7 +7934,8 @@
 #ifdef TEE_SINK
         // FIXME
 #endif
-        mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture");
+        mFastCaptureNBLogWriter =
+                afThreadCallback->newWriter_l(kFastCaptureLogSize, "FastCapture");
         state->mNBLogWriter = mFastCaptureNBLogWriter.get();
         sq->end();
         sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED);
@@ -7926,7 +7960,7 @@
     // FIXME mNormalSource
 }
 
-AudioFlinger::RecordThread::~RecordThread()
+RecordThread::~RecordThread()
 {
     if (mFastCapture != 0) {
         FastCaptureStateQueue *sq = mFastCapture->sq();
@@ -7943,36 +7977,36 @@
         mFastCapture->join();
         mFastCapture.clear();
     }
-    mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter);
-    mAudioFlinger->unregisterWriter(mNBLogWriter);
+    mAfThreadCallback->unregisterWriter(mFastCaptureNBLogWriter);
+    mAfThreadCallback->unregisterWriter(mNBLogWriter);
     free(mRsmpInBuffer);
 }
 
-void AudioFlinger::RecordThread::onFirstRef()
+void RecordThread::onFirstRef()
 {
     run(mThreadName, PRIORITY_URGENT_AUDIO);
 }
 
-void AudioFlinger::RecordThread::preExit()
+void RecordThread::preExit()
 {
     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();
     mStartStopCond.broadcast();
 }
 
-bool AudioFlinger::RecordThread::threadLoop()
+bool RecordThread::threadLoop()
 {
     nsecs_t lastWarning = 0;
 
     inputStandBy();
 
 reacquire_wakelock:
-    sp<RecordTrack> activeTrack;
+    sp<IAfRecordTrack> activeTrack;
     {
         Mutex::Autolock _l(mLock);
         acquireWakeLock_l();
@@ -7988,13 +8022,13 @@
         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;
 
@@ -8047,40 +8081,40 @@
                     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);
@@ -8172,7 +8206,7 @@
                 }
                 state->mCommand = FastCaptureState::READ_WRITE;
 #if 0   // FIXME
-                mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
+                mFastCaptureDumpState.increaseSamplingN(mAfThreadCallback->isLowRamDevice() ?
                         FastThreadDumpState::kSamplingNforLowRamDevice :
                         FastThreadDumpState::kSamplingN);
 #endif
@@ -8327,7 +8361,7 @@
 
         // 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);
@@ -8384,16 +8418,16 @@
             // 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;
                 }
@@ -8407,7 +8441,7 @@
                 // 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
@@ -8415,14 +8449,15 @@
                     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",
@@ -8431,9 +8466,9 @@
                 } 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);
                 }
 
@@ -8443,17 +8478,18 @@
 
                 // 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) {
@@ -8482,7 +8518,7 @@
 
             // update frame information and push timestamp out
             activeTrack->updateTrackFrameInfo(
-                    activeTrack->mServerProxy->framesReleased(),
+                    activeTrack->serverProxy()->framesReleased(),
                     mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER],
                     mSampleRate, mTimestamp);
         }
@@ -8515,7 +8551,7 @@
     {
         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();
@@ -8528,7 +8564,7 @@
     return false;
 }
 
-void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby()
+void RecordThread::standbyIfNotAlreadyInStandby()
 {
     if (!mStandby) {
         inputStandBy();
@@ -8538,7 +8574,7 @@
     }
 }
 
-void AudioFlinger::RecordThread::inputStandBy()
+void RecordThread::inputStandBy()
 {
     // Idle the fast capture if it's currently running
     if (mFastCapture != 0) {
@@ -8579,7 +8615,7 @@
 }
 
 // RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held
-sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l(
+sp<IAfRecordTrack> RecordThread::createRecordTrack_l(
         const sp<Client>& client,
         const audio_attributes_t& attr,
         uint32_t *pSampleRate,
@@ -8598,7 +8634,7 @@
 {
     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;
@@ -8744,10 +8780,10 @@
             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();
@@ -8777,7 +8813,7 @@
     return track;
 }
 
-status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack,
+status_t RecordThread::start(IAfRecordTrack* recordTrack,
                                            AudioSystem::sync_event_t event,
                                            audio_session_t triggerSession)
 {
@@ -8788,8 +8824,8 @@
     if (event == AudioSystem::SYNC_EVENT_NONE) {
         recordTrack->clearSyncStartEvent();
     } else if (event != AudioSystem::SYNC_EVENT_SAME) {
-        recordTrack->mSynchronizedRecordState.startRecording(
-                mAudioFlinger->createSyncEvent(
+        recordTrack->synchronizedRecordState().startRecording(
+                mAfThreadCallback->createSyncEvent(
                         event, triggerSession,
                         recordTrack->sessionId(), syncStartEventCallback, recordTrack));
     }
@@ -8803,13 +8839,13 @@
             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;
         }
@@ -8817,7 +8853,7 @@
         // 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();
@@ -8825,16 +8861,16 @@
             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();
@@ -8861,67 +8897,66 @@
         // 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;
     }
 }
 
-void AudioFlinger::RecordThread::syncStartEventCallback(const wp<audioflinger::SyncEvent>& event)
+void RecordThread::syncStartEventCallback(const wp<SyncEvent>& event)
 {
-    sp<audioflinger::SyncEvent> strongEvent = event.promote();
+    const sp<SyncEvent> strongEvent = event.promote();
 
     if (strongEvent != 0) {
-        sp<RefBase> ptr = std::any_cast<const wp<RefBase>>(strongEvent->cookie()).promote();
-        if (ptr != 0) {
-            RecordTrack *recordTrack = (RecordTrack *)ptr.get();
-            recordTrack->handleSyncStartEvent(strongEvent);
+        sp<IAfTrackBase> ptr =
+                std::any_cast<const wp<IAfTrackBase>>(strongEvent->cookie()).promote();
+        if (ptr != nullptr) {
+            // TODO(b/291317898) handleSyncStartEvent is in IAfTrackBase not IAfRecordTrack.
+            ptr->handleSyncStartEvent(strongEvent);
         }
     }
 }
 
-bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
+bool 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;
 }
 
-bool AudioFlinger::RecordThread::isValidSyncEvent(
-        const sp<audioflinger::SyncEvent>& /* event */) const
+bool RecordThread::isValidSyncEvent(const sp<SyncEvent>& /* event */) const
 {
     return false;
 }
 
-status_t AudioFlinger::RecordThread::setSyncEvent(
-        const sp<audioflinger::SyncEvent>& event __unused)
+status_t RecordThread::setSyncEvent(const sp<SyncEvent>& /* event */)
 {
 #if 0   // This branch is currently dead code, but is preserved in case it will be needed in future
     if (!isValidSyncEvent(event)) {
@@ -8934,7 +8969,7 @@
     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;
@@ -8946,8 +8981,8 @@
 #endif
 }
 
-status_t AudioFlinger::RecordThread::getActiveMicrophones(
-        std::vector<media::MicrophoneInfoFw>* activeMicrophones)
+status_t RecordThread::getActiveMicrophones(
+        std::vector<media::MicrophoneInfoFw>* activeMicrophones) const
 {
     ALOGV("RecordThread::getActiveMicrophones");
     AutoMutex _l(mLock);
@@ -8958,7 +8993,7 @@
     return status;
 }
 
-status_t AudioFlinger::RecordThread::setPreferredMicrophoneDirection(
+status_t RecordThread::setPreferredMicrophoneDirection(
             audio_microphone_direction_t direction)
 {
     ALOGV("setPreferredMicrophoneDirection(%d)", direction);
@@ -8969,7 +9004,7 @@
     return mInput->stream->setPreferredMicrophoneDirection(direction);
 }
 
-status_t AudioFlinger::RecordThread::setPreferredMicrophoneFieldDimension(float zoom)
+status_t RecordThread::setPreferredMicrophoneFieldDimension(float zoom)
 {
     ALOGV("setPreferredMicrophoneFieldDimension(%f)", zoom);
     AutoMutex _l(mLock);
@@ -8979,14 +9014,14 @@
     return mInput->stream->setPreferredMicrophoneFieldDimension(zoom);
 }
 
-status_t AudioFlinger::RecordThread::shareAudioHistory(
+status_t RecordThread::shareAudioHistory(
         const std::string& sharedAudioPackageName, audio_session_t sharedSessionId,
         int64_t sharedAudioStartMs) {
     AutoMutex _l(mLock);
     return shareAudioHistory_l(sharedAudioPackageName, sharedSessionId, sharedAudioStartMs);
 }
 
-status_t AudioFlinger::RecordThread::shareAudioHistory_l(
+status_t RecordThread::shareAudioHistory_l(
         const std::string& sharedAudioPackageName, audio_session_t sharedSessionId,
         int64_t sharedAudioStartMs) {
 
@@ -9026,20 +9061,20 @@
     return NO_ERROR;
 }
 
-void AudioFlinger::RecordThread::resetAudioHistory_l() {
+void RecordThread::resetAudioHistory_l() {
     mSharedAudioSessionId = AUDIO_SESSION_NONE;
     mSharedAudioStartFrames = -1;
     mSharedAudioPackageName = "";
 }
 
-AudioFlinger::ThreadBase::MetadataUpdate AudioFlinger::RecordThread::updateMetadata_l()
+ThreadBase::MetadataUpdate RecordThread::updateMetadata_l()
 {
     if (!isStreamInitialized() || !mActiveTracks.readAndClearHasChanged()) {
         return {}; // nothing to do
     }
     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);
@@ -9049,10 +9084,10 @@
 }
 
 // destroyTrack_l() must be called with ThreadBase::mLock held
-void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track)
+void 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) {
@@ -9060,7 +9095,7 @@
     }
 }
 
-void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track)
+void RecordThread::removeTrack_l(const sp<IAfRecordTrack>& track)
 {
     String8 result;
     track->appendDump(result, false /* active */);
@@ -9074,7 +9109,7 @@
     }
 }
 
-void AudioFlinger::RecordThread::dumpInternals_l(int fd, const Vector<String16>& args __unused)
+void RecordThread::dumpInternals_l(int fd, const Vector<String16>& /* args */)
 {
     AudioStreamIn *input = mInput;
     audio_input_flags_t flags = input != NULL ? input->flags : AUDIO_INPUT_FLAG_NONE;
@@ -9102,7 +9137,7 @@
     copy->dump(fd);
 }
 
-void AudioFlinger::RecordThread::dumpTracks_l(int fd, const Vector<String16>& args __unused)
+void RecordThread::dumpTracks_l(int fd, const Vector<String16>& /* args */)
 {
     String8 result;
     size_t numtracks = mTracks.size();
@@ -9115,7 +9150,7 @@
         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) {
@@ -9135,7 +9170,7 @@
         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 */);
@@ -9146,21 +9181,21 @@
     write(fd, result.string(), result.size());
 }
 
-void AudioFlinger::RecordThread::setRecordSilenced(audio_port_handle_t portId, bool silenced)
+void RecordThread::setRecordSilenced(audio_port_handle_t portId, bool silenced)
 {
     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 = mRecordTrack->thread().promote();
+    auto* const recordThread = static_cast<RecordThread *>(threadBase->asIAfRecordThread().get());
     mRsmpInUnrel = 0;
     const int32_t rear = recordThread->mRsmpInRear;
     ssize_t deltaFrames = 0;
@@ -9180,11 +9215,11 @@
     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 = mRecordTrack->thread().promote();
+    auto* const recordThread = static_cast<RecordThread *>(threadBase->asIAfRecordThread().get());
     const int32_t rear = recordThread->mRsmpInRear;
     const int32_t front = mRsmpInFront;
     const ssize_t filled = audio_utils::safe_sub_overflow(rear, front);
@@ -9214,16 +9249,16 @@
 }
 
 // AudioBufferProvider interface
-status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
+status_t ResamplerBufferProvider::getNextBuffer(
         AudioBufferProvider::Buffer* buffer)
 {
-    sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
+    const auto threadBase = 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<RecordThread *>(threadBase->asIAfRecordThread().get());
     int32_t rear = recordThread->mRsmpInRear;
     int32_t front = mRsmpInFront;
     ssize_t filled = audio_utils::safe_sub_overflow(rear, front);
@@ -9257,7 +9292,7 @@
 }
 
 // AudioBufferProvider interface
-void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer(
+void ResamplerBufferProvider::releaseBuffer(
         AudioBufferProvider::Buffer* buffer)
 {
     int32_t stepCount = static_cast<int32_t>(buffer->frameCount);
@@ -9271,18 +9306,18 @@
     buffer->frameCount = 0;
 }
 
-void AudioFlinger::RecordThread::checkBtNrec()
+void RecordThread::checkBtNrec()
 {
     Mutex::Autolock _l(mLock);
     checkBtNrec_l();
 }
 
-void AudioFlinger::RecordThread::checkBtNrec_l()
+void RecordThread::checkBtNrec_l()
 {
     // disable AEC and NS if the device is a BT SCO headset supporting those
     // pre processings
     bool suspend = audio_is_bluetooth_sco_device(inDeviceType()) &&
-                        mAudioFlinger->btNrecIsOff();
+                        mAfThreadCallback->btNrecIsOff();
     if (mBtNrecSuspended.exchange(suspend) != suspend) {
         for (size_t i = 0; i < mEffectChains.size(); i++) {
             setEffectSuspended_l(FX_IID_AEC, suspend, mEffectChains[i]->sessionId());
@@ -9292,7 +9327,7 @@
 }
 
 
-bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair,
+bool RecordThread::checkForNewParameter_l(const String8& keyValuePair,
                                                         status_t& status)
 {
     bool reconfig = false;
@@ -9380,7 +9415,7 @@
     return reconfig;
 }
 
-String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
+String8 RecordThread::getParameters(const String8& keys)
 {
     Mutex::Autolock _l(mLock);
     if (initCheck() == NO_ERROR) {
@@ -9392,7 +9427,7 @@
     return {};
 }
 
-void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
+void RecordThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
                                                  audio_port_handle_t portId) {
     sp<AudioIoDescriptor> desc;
     switch (event) {
@@ -9410,10 +9445,10 @@
         desc = sp<AudioIoDescriptor>::make(mId);
         break;
     }
-    mAudioFlinger->ioConfigChanged(event, desc, pid);
+    mAfThreadCallback->ioConfigChanged(event, desc, pid);
 }
 
-void AudioFlinger::RecordThread::readInputParameters_l()
+void RecordThread::readInputParameters_l()
 {
     status_t result = mInput->stream->getAudioProperties(&mSampleRate, &mChannelMask, &mHALFormat);
     LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving audio properties from HAL: %d", result);
@@ -9456,7 +9491,7 @@
         .record();
 }
 
-uint32_t AudioFlinger::RecordThread::getInputFramesLost()
+uint32_t RecordThread::getInputFramesLost() const
 {
     Mutex::Autolock _l(mLock);
     uint32_t result;
@@ -9466,12 +9501,12 @@
     return 0;
 }
 
-KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const
+KeyedVector<audio_session_t, bool> RecordThread::sessionIds() const
 {
     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);
@@ -9480,7 +9515,7 @@
     return ids;
 }
 
-AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
+AudioStreamIn* RecordThread::clearInput()
 {
     Mutex::Autolock _l(mLock);
     AudioStreamIn *input = mInput;
@@ -9489,7 +9524,7 @@
 }
 
 // this method must always be called either with ThreadBase mLock held or inside the thread loop
-sp<StreamHalInterface> AudioFlinger::RecordThread::stream() const
+sp<StreamHalInterface> RecordThread::stream() const
 {
     if (mInput == NULL) {
         return NULL;
@@ -9497,7 +9532,7 @@
     return mInput->stream;
 }
 
-status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<IAfEffectChain>& chain)
+status_t RecordThread::addEffectChain_l(const sp<IAfEffectChain>& chain)
 {
     ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
     chain->setThread(this);
@@ -9515,7 +9550,7 @@
     return NO_ERROR;
 }
 
-size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<IAfEffectChain>& chain)
+size_t RecordThread::removeEffectChain_l(const sp<IAfEffectChain>& chain)
 {
     ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
 
@@ -9528,7 +9563,7 @@
     return mEffectChains.size();
 }
 
-status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch,
+status_t RecordThread::createAudioPatch_l(const struct audio_patch* patch,
                                                           audio_patch_handle_t *handle)
 {
     status_t status = NO_ERROR;
@@ -9585,7 +9620,7 @@
     return status;
 }
 
-status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
+status_t RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
 {
     status_t status = NO_ERROR;
 
@@ -9604,7 +9639,7 @@
     return status;
 }
 
-void AudioFlinger::RecordThread::updateOutDevices(const DeviceDescriptorBaseVector& outDevices)
+void RecordThread::updateOutDevices(const DeviceDescriptorBaseVector& outDevices)
 {
     Mutex::Autolock _l(mLock);
     mOutDevices = outDevices;
@@ -9614,7 +9649,7 @@
     }
 }
 
-int32_t AudioFlinger::RecordThread::getOldestFront_l()
+int32_t RecordThread::getOldestFront_l()
 {
     if (mTracks.size() == 0) {
         return mRsmpInRear;
@@ -9622,7 +9657,7 @@
     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) {
@@ -9636,19 +9671,19 @@
     return oldestFront;
 }
 
-void AudioFlinger::RecordThread::updateFronts_l(int32_t offset)
+void RecordThread::updateFronts_l(int32_t offset)
 {
     if (offset == 0) {
         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);
     }
 }
 
-void AudioFlinger::RecordThread::resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs)
+void RecordThread::resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs)
 {
     // This is the formula for calculating the temporary buffer size.
     // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to
@@ -9672,7 +9707,7 @@
     mRsmpInRear = 0;
 
     ALOG_ASSERT(maxSharedAudioHistoryMs >= 0
-            && maxSharedAudioHistoryMs <= AudioFlinger::kMaxSharedAudioHistoryMs,
+            && maxSharedAudioHistoryMs <= kMaxSharedAudioHistoryMs,
             "resizeInputBuffer_l() called with invalid max shared history %d",
             maxSharedAudioHistoryMs);
     if (maxSharedAudioHistoryMs != 0) {
@@ -9741,7 +9776,7 @@
     mRsmpInBuffer = rsmpInBuffer;
 }
 
-void AudioFlinger::RecordThread::addPatchTrack(const sp<PatchRecord>& record)
+void RecordThread::addPatchTrack(const sp<IAfPatchRecord>& record)
 {
     Mutex::Autolock _l(mLock);
     mTracks.add(record);
@@ -9750,7 +9785,7 @@
     }
 }
 
-void AudioFlinger::RecordThread::deletePatchTrack(const sp<PatchRecord>& record)
+void RecordThread::deletePatchTrack(const sp<IAfPatchRecord>& record)
 {
     Mutex::Autolock _l(mLock);
     if (mSource == record->getSource()) {
@@ -9759,7 +9794,7 @@
     destroyTrack_l(record);
 }
 
-void AudioFlinger::RecordThread::toAudioPortConfig(struct audio_port_config *config)
+void RecordThread::toAudioPortConfig(struct audio_port_config* config)
 {
     ThreadBase::toAudioPortConfig(config);
     config->role = AUDIO_PORT_ROLE_SINK;
@@ -9775,59 +9810,88 @@
 //      Mmap
 // ----------------------------------------------------------------------------
 
-AudioFlinger::MmapThreadHandle::MmapThreadHandle(const sp<MmapThread>& thread)
+// Mmap stream control interface implementation. Each MmapThreadHandle controls one
+// MmapPlaybackThread or MmapCaptureThread instance.
+class MmapThreadHandle : public MmapStreamInterface {
+public:
+    explicit MmapThreadHandle(const sp<IAfMmapThread>& thread);
+    ~MmapThreadHandle() override;
+
+    // MmapStreamInterface virtuals
+    status_t createMmapBuffer(int32_t minSizeFrames,
+        struct audio_mmap_buffer_info* info) final;
+    status_t getMmapPosition(struct audio_mmap_position* position) final;
+    status_t getExternalPosition(uint64_t* position, int64_t* timeNanos) final;
+    status_t start(const AudioClient& client,
+           const audio_attributes_t* attr, audio_port_handle_t* handle) final;
+    status_t stop(audio_port_handle_t handle) final;
+    status_t standby() final;
+    status_t reportData(const void* buffer, size_t frameCount) final;
+private:
+    const sp<IAfMmapThread> mThread;
+};
+
+/* static */
+sp<MmapStreamInterface> IAfMmapThread::createMmapStreamInterfaceAdapter(
+        const sp<IAfMmapThread>& mmapThread) {
+    return sp<MmapThreadHandle>::make(mmapThread);
+}
+
+MmapThreadHandle::MmapThreadHandle(const sp<IAfMmapThread>& thread)
     : mThread(thread)
 {
     assert(thread != 0); // thread must start non-null and stay non-null
 }
 
-AudioFlinger::MmapThreadHandle::~MmapThreadHandle()
+// MmapStreamInterface could be directly implemented by MmapThread excepting this
+// special handling on adapter dtor.
+MmapThreadHandle::~MmapThreadHandle()
 {
     mThread->disconnect();
 }
 
-status_t AudioFlinger::MmapThreadHandle::createMmapBuffer(int32_t minSizeFrames,
+status_t MmapThreadHandle::createMmapBuffer(int32_t minSizeFrames,
                                   struct audio_mmap_buffer_info *info)
 {
     return mThread->createMmapBuffer(minSizeFrames, info);
 }
 
-status_t AudioFlinger::MmapThreadHandle::getMmapPosition(struct audio_mmap_position *position)
+status_t MmapThreadHandle::getMmapPosition(struct audio_mmap_position* position)
 {
     return mThread->getMmapPosition(position);
 }
 
-status_t AudioFlinger::MmapThreadHandle::getExternalPosition(uint64_t *position,
+status_t MmapThreadHandle::getExternalPosition(uint64_t* position,
                                                              int64_t *timeNanos) {
     return mThread->getExternalPosition(position, timeNanos);
 }
 
-status_t AudioFlinger::MmapThreadHandle::start(const AudioClient& client,
+status_t MmapThreadHandle::start(const AudioClient& client,
         const audio_attributes_t *attr, audio_port_handle_t *handle)
-
 {
     return mThread->start(client, attr, handle);
 }
 
-status_t AudioFlinger::MmapThreadHandle::stop(audio_port_handle_t handle)
+status_t MmapThreadHandle::stop(audio_port_handle_t handle)
 {
     return mThread->stop(handle);
 }
 
-status_t AudioFlinger::MmapThreadHandle::standby()
+status_t MmapThreadHandle::standby()
 {
     return mThread->standby();
 }
 
-status_t AudioFlinger::MmapThreadHandle::reportData(const void* buffer, size_t frameCount) {
+status_t MmapThreadHandle::reportData(const void* buffer, size_t frameCount)
+{
     return mThread->reportData(buffer, frameCount);
 }
 
 
-AudioFlinger::MmapThread::MmapThread(
-        const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+MmapThread::MmapThread(
+        const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
         AudioHwDevice *hwDev, const sp<StreamHalInterface>& stream, bool systemReady, bool isOut)
-    : ThreadBase(audioFlinger, id, (isOut ? MMAP_PLAYBACK : MMAP_CAPTURE), systemReady, isOut),
+    : ThreadBase(afThreadCallback, id, (isOut ? MMAP_PLAYBACK : MMAP_CAPTURE), systemReady, isOut),
       mSessionId(AUDIO_SESSION_NONE),
       mPortId(AUDIO_PORT_HANDLE_NONE),
       mHalStream(stream), mHalDevice(hwDev->hwDevice()), mAudioHwDev(hwDev),
@@ -9839,25 +9903,21 @@
     readHalParameters_l();
 }
 
-AudioFlinger::MmapThread::~MmapThread()
-{
-}
-
-void AudioFlinger::MmapThread::onFirstRef()
+void MmapThread::onFirstRef()
 {
     run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO);
 }
 
-void AudioFlinger::MmapThread::disconnect()
+void 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.
@@ -9869,7 +9929,7 @@
 }
 
 
-void AudioFlinger::MmapThread::configure(const audio_attributes_t *attr,
+void MmapThread::configure(const audio_attributes_t* attr,
                                                 audio_stream_type_t streamType __unused,
                                                 audio_session_t sessionId,
                                                 const sp<MmapStreamCallback>& callback,
@@ -9883,7 +9943,7 @@
     mPortId = portId;
 }
 
-status_t AudioFlinger::MmapThread::createMmapBuffer(int32_t minSizeFrames,
+status_t MmapThread::createMmapBuffer(int32_t minSizeFrames,
                                   struct audio_mmap_buffer_info *info)
 {
     if (mHalStream == 0) {
@@ -9893,7 +9953,7 @@
     return mHalStream->createMmapBuffer(minSizeFrames, info);
 }
 
-status_t AudioFlinger::MmapThread::getMmapPosition(struct audio_mmap_position *position)
+status_t MmapThread::getMmapPosition(struct audio_mmap_position* position) const
 {
     if (mHalStream == 0) {
         return NO_INIT;
@@ -9901,7 +9961,7 @@
     return mHalStream->getMmapPosition(position);
 }
 
-status_t AudioFlinger::MmapThread::exitStandby_l()
+status_t MmapThread::exitStandby_l()
 {
     // The HAL must receive track metadata before starting the stream
     updateMetadata_l();
@@ -9918,7 +9978,7 @@
     return NO_ERROR;
 }
 
-status_t AudioFlinger::MmapThread::start(const AudioClient& client,
+status_t MmapThread::start(const AudioClient& client,
                                          const audio_attributes_t *attr,
                                          audio_port_handle_t *handle)
 {
@@ -9939,7 +9999,7 @@
     audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
 
     audio_io_handle_t io = mId;
-    AttributionSourceState adjAttributionSource = AudioFlinger::checkAttributionSourcePackage(
+    const AttributionSourceState adjAttributionSource = afutils::checkAttributionSourcePackage(
             client.attributionSource);
 
     if (isOutput()) {
@@ -10022,7 +10082,8 @@
     }
 
     // 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);
@@ -10034,7 +10095,7 @@
         // 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();
@@ -10064,7 +10125,7 @@
     return ret;
 }
 
-status_t AudioFlinger::MmapThread::stop(audio_port_handle_t handle)
+status_t MmapThread::stop(audio_port_handle_t handle)
 {
     ALOGV("%s handle %d", __FUNCTION__, handle);
 
@@ -10079,8 +10140,8 @@
 
     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;
@@ -10118,7 +10179,7 @@
     return NO_ERROR;
 }
 
-status_t AudioFlinger::MmapThread::standby()
+status_t MmapThread::standby()
 {
     ALOGV("%s", __FUNCTION__);
 
@@ -10138,12 +10199,12 @@
     return NO_ERROR;
 }
 
-status_t AudioFlinger::MmapThread::reportData(const void* /*buffer*/, size_t /*frameCount*/) {
+status_t MmapThread::reportData(const void* /*buffer*/, size_t /*frameCount*/) {
     // This is a stub implementation. The MmapPlaybackThread overrides this function.
     return INVALID_OPERATION;
 }
 
-void AudioFlinger::MmapThread::readHalParameters_l()
+void MmapThread::readHalParameters_l()
 {
     status_t result = mHalStream->getAudioProperties(&mSampleRate, &mChannelMask, &mHALFormat);
     LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving audio properties from HAL: %d", result);
@@ -10179,7 +10240,7 @@
         .record();
 }
 
-bool AudioFlinger::MmapThread::threadLoop()
+bool MmapThread::threadLoop()
 {
     checkSilentMode_l();
 
@@ -10250,7 +10311,7 @@
 }
 
 // checkForNewParameter_l() must be called with ThreadBase::mLock held
-bool AudioFlinger::MmapThread::checkForNewParameter_l(const String8& keyValuePair,
+bool MmapThread::checkForNewParameter_l(const String8& keyValuePair,
                                                               status_t& status)
 {
     AudioParameter param = AudioParameter(keyValuePair);
@@ -10268,7 +10329,7 @@
     return false;
 }
 
-String8 AudioFlinger::MmapThread::getParameters(const String8& keys)
+String8 MmapThread::getParameters(const String8& keys)
 {
     Mutex::Autolock _l(mLock);
     String8 out_s8;
@@ -10278,7 +10339,7 @@
     return {};
 }
 
-void AudioFlinger::MmapThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
+void MmapThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
                                                audio_port_handle_t portId __unused) {
     sp<AudioIoDescriptor> desc;
     bool isInput = false;
@@ -10300,10 +10361,10 @@
         desc = sp<AudioIoDescriptor>::make(mId);
         break;
     }
-    mAudioFlinger->ioConfigChanged(event, desc, pid);
+    mAfThreadCallback->ioConfigChanged(event, desc, pid);
 }
 
-status_t AudioFlinger::MmapThread::createAudioPatch_l(const struct audio_patch *patch,
+status_t MmapThread::createAudioPatch_l(const struct audio_patch* patch,
                                                           audio_patch_handle_t *handle)
 NO_THREAD_SAFETY_ANALYSIS  // elease and re-acquire mLock
 {
@@ -10394,7 +10455,7 @@
     return status;
 }
 
-status_t AudioFlinger::MmapThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
+status_t MmapThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
 {
     status_t status = NO_ERROR;
 
@@ -10416,7 +10477,7 @@
     return status;
 }
 
-void AudioFlinger::MmapThread::toAudioPortConfig(struct audio_port_config *config)
+void MmapThread::toAudioPortConfig(struct audio_port_config* config)
 {
     ThreadBase::toAudioPortConfig(config);
     if (isOutput()) {
@@ -10430,14 +10491,14 @@
     }
 }
 
-status_t AudioFlinger::MmapThread::addEffectChain_l(const sp<IAfEffectChain>& chain)
+status_t MmapThread::addEffectChain_l(const sp<IAfEffectChain>& chain)
 {
     audio_session_t session = chain->sessionId();
 
     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();
@@ -10454,7 +10515,7 @@
     return NO_ERROR;
 }
 
-size_t AudioFlinger::MmapThread::removeEffectChain_l(const sp<IAfEffectChain>& chain)
+size_t MmapThread::removeEffectChain_l(const sp<IAfEffectChain>& chain)
 {
     audio_session_t session = chain->sessionId();
 
@@ -10465,7 +10526,7 @@
             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();
@@ -10477,29 +10538,29 @@
     return mEffectChains.size();
 }
 
-void AudioFlinger::MmapThread::threadLoop_standby()
+void MmapThread::threadLoop_standby()
 {
     mHalStream->standby();
 }
 
-void AudioFlinger::MmapThread::threadLoop_exit()
+void MmapThread::threadLoop_exit()
 {
     // Do not call callback->onTearDown() because it is redundant for thread exit
     // and because it can cause a recursive mutex lock on stop().
 }
 
-status_t AudioFlinger::MmapThread::setSyncEvent(const sp<audioflinger::SyncEvent>& /* event */)
+status_t MmapThread::setSyncEvent(const sp<SyncEvent>& /* event */)
 {
     return BAD_VALUE;
 }
 
-bool AudioFlinger::MmapThread::isValidSyncEvent(
-        const sp<audioflinger::SyncEvent>& /* event */) const
+bool MmapThread::isValidSyncEvent(
+        const sp<SyncEvent>& /* event */) const
 {
     return false;
 }
 
-status_t AudioFlinger::MmapThread::checkEffectCompatibility_l(
+status_t MmapThread::checkEffectCompatibility_l(
         const effect_descriptor_t *desc, audio_session_t sessionId)
 {
     // No global effect sessions on mmap threads
@@ -10533,11 +10594,11 @@
     return NO_ERROR;
 }
 
-void AudioFlinger::MmapThread::checkInvalidTracks_l()
+void MmapThread::checkInvalidTracks_l()
 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) {
@@ -10554,7 +10615,7 @@
     }
 }
 
-void AudioFlinger::MmapThread::dumpInternals_l(int fd, const Vector<String16>& args __unused)
+void MmapThread::dumpInternals_l(int fd, const Vector<String16>& /* args */)
 {
     dprintf(fd, "  Attributes: content type %d usage %d source %d\n",
             mAttr.content_type, mAttr.usage, mAttr.source);
@@ -10564,7 +10625,7 @@
     }
 }
 
-void AudioFlinger::MmapThread::dumpTracks_l(int fd, const Vector<String16>& args __unused)
+void MmapThread::dumpTracks_l(int fd, const Vector<String16>& /* args */)
 {
     String8 result;
     size_t numtracks = mActiveTracks.size();
@@ -10574,7 +10635,7 @@
         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 */);
         }
@@ -10584,10 +10645,17 @@
     write(fd, result.string(), result.size());
 }
 
-AudioFlinger::MmapPlaybackThread::MmapPlaybackThread(
-        const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+/* static */
+sp<IAfMmapPlaybackThread> IAfMmapPlaybackThread::create(
+        const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
+        AudioHwDevice* hwDev,  AudioStreamOut* output, bool systemReady) {
+    return sp<MmapPlaybackThread>::make(afThreadCallback, id, hwDev, output, systemReady);
+}
+
+MmapPlaybackThread::MmapPlaybackThread(
+        const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
         AudioHwDevice *hwDev,  AudioStreamOut *output, bool systemReady)
-    : MmapThread(audioFlinger, id, hwDev, output->stream, systemReady, true /* isOut */),
+    : MmapThread(afThreadCallback, id, hwDev, output->stream, systemReady, true /* isOut */),
       mStreamType(AUDIO_STREAM_MUSIC),
       mStreamVolume(1.0),
       mStreamMute(false),
@@ -10595,8 +10663,8 @@
 {
     snprintf(mThreadName, kThreadNameLength, "AudioMmapOut_%X", id);
     mChannelCount = audio_channel_count_from_out_mask(mChannelMask);
-    mMasterVolume = audioFlinger->masterVolume_l();
-    mMasterMute = audioFlinger->masterMute_l();
+    mMasterVolume = afThreadCallback->masterVolume_l();
+    mMasterMute = afThreadCallback->masterMute_l();
     if (mAudioHwDev) {
         if (mAudioHwDev->canSetMasterVolume()) {
             mMasterVolume = 1.0;
@@ -10608,7 +10676,7 @@
     }
 }
 
-void AudioFlinger::MmapPlaybackThread::configure(const audio_attributes_t *attr,
+void MmapPlaybackThread::configure(const audio_attributes_t* attr,
                                                 audio_stream_type_t streamType,
                                                 audio_session_t sessionId,
                                                 const sp<MmapStreamCallback>& callback,
@@ -10619,7 +10687,7 @@
     mStreamType = streamType;
 }
 
-AudioStreamOut* AudioFlinger::MmapPlaybackThread::clearOutput()
+AudioStreamOut* MmapPlaybackThread::clearOutput()
 {
     Mutex::Autolock _l(mLock);
     AudioStreamOut *output = mOutput;
@@ -10627,7 +10695,7 @@
     return output;
 }
 
-void AudioFlinger::MmapPlaybackThread::setMasterVolume(float value)
+void MmapPlaybackThread::setMasterVolume(float value)
 {
     Mutex::Autolock _l(mLock);
     // Don't apply master volume in SW if our HAL can do it for us.
@@ -10639,7 +10707,7 @@
     }
 }
 
-void AudioFlinger::MmapPlaybackThread::setMasterMute(bool muted)
+void MmapPlaybackThread::setMasterMute(bool muted)
 {
     Mutex::Autolock _l(mLock);
     // Don't apply master mute in SW if our HAL can do it for us.
@@ -10650,7 +10718,7 @@
     }
 }
 
-void AudioFlinger::MmapPlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
+void MmapPlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
 {
     Mutex::Autolock _l(mLock);
     if (stream == mStreamType) {
@@ -10659,7 +10727,7 @@
     }
 }
 
-float AudioFlinger::MmapPlaybackThread::streamVolume(audio_stream_type_t stream) const
+float MmapPlaybackThread::streamVolume(audio_stream_type_t stream) const
 {
     Mutex::Autolock _l(mLock);
     if (stream == mStreamType) {
@@ -10668,7 +10736,7 @@
     return 0.0f;
 }
 
-void AudioFlinger::MmapPlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
+void MmapPlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
 {
     Mutex::Autolock _l(mLock);
     if (stream == mStreamType) {
@@ -10677,22 +10745,22 @@
     }
 }
 
-void AudioFlinger::MmapPlaybackThread::invalidateTracks(audio_stream_type_t streamType)
+void MmapPlaybackThread::invalidateTracks(audio_stream_type_t streamType)
 {
     Mutex::Autolock _l(mLock);
     if (streamType == mStreamType) {
-        for (const sp<MmapTrack> &track : mActiveTracks) {
+        for (const sp<IAfMmapTrack>& track : mActiveTracks) {
             track->invalidate();
         }
         broadcast_l();
     }
 }
 
-void AudioFlinger::MmapPlaybackThread::invalidateTracks(std::set<audio_port_handle_t>& portIds)
+void MmapPlaybackThread::invalidateTracks(std::set<audio_port_handle_t>& portIds)
 {
     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;
@@ -10707,7 +10775,7 @@
     }
 }
 
-void AudioFlinger::MmapPlaybackThread::processVolume_l()
+void MmapPlaybackThread::processVolume_l()
 NO_THREAD_SAFETY_ANALYSIS // access of track->processMuteEvent_l
 {
     float volume;
@@ -10749,9 +10817,9 @@
                 }
             }
         }
-        for (const sp<MmapTrack> &track : mActiveTracks) {
+        for (const sp<IAfMmapTrack>& track : mActiveTracks) {
             track->setMetadataHasChanged();
-            track->processMuteEvent_l(mAudioFlinger->getOrCreateAudioManager(),
+            track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
                 /*muteState=*/{mMasterMute,
                                mStreamVolume == 0.f,
                                mStreamMute,
@@ -10763,13 +10831,13 @@
     }
 }
 
-AudioFlinger::ThreadBase::MetadataUpdate AudioFlinger::MmapPlaybackThread::updateMetadata_l()
+ThreadBase::MetadataUpdate MmapPlaybackThread::updateMetadata_l()
 {
     if (!isStreamInitialized() || !mActiveTracks.readAndClearHasChanged()) {
         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 = {
@@ -10788,7 +10856,7 @@
     return change;
 };
 
-void AudioFlinger::MmapPlaybackThread::checkSilentMode_l()
+void MmapPlaybackThread::checkSilentMode_l()
 {
     if (!mMasterMute) {
         char value[PROPERTY_VALUE_MAX];
@@ -10805,7 +10873,7 @@
     }
 }
 
-void AudioFlinger::MmapPlaybackThread::toAudioPortConfig(struct audio_port_config *config)
+void MmapPlaybackThread::toAudioPortConfig(struct audio_port_config* config)
 {
     MmapThread::toAudioPortConfig(config);
     if (mOutput && mOutput->flags != AUDIO_OUTPUT_FLAG_NONE) {
@@ -10814,8 +10882,8 @@
     }
 }
 
-status_t AudioFlinger::MmapPlaybackThread::getExternalPosition(uint64_t *position,
-                                                               int64_t *timeNanos)
+status_t MmapPlaybackThread::getExternalPosition(uint64_t* position,
+        int64_t* timeNanos) const
 {
     if (mOutput == nullptr) {
         return NO_INIT;
@@ -10828,7 +10896,7 @@
     return status;
 }
 
-status_t AudioFlinger::MmapPlaybackThread::reportData(const void* buffer, size_t frameCount) {
+status_t MmapPlaybackThread::reportData(const void* buffer, size_t frameCount) {
     // Send to MelProcessor for sound dose measurement.
     auto processor = mMelProcessor.load();
     if (processor) {
@@ -10839,7 +10907,7 @@
 }
 
 // startMelComputation_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MmapPlaybackThread::startMelComputation_l(
+void MmapPlaybackThread::startMelComputation_l(
         const sp<audio_utils::MelProcessor>& processor)
 {
     ALOGV("%s: starting mel processor for thread %d", __func__, id());
@@ -10853,7 +10921,7 @@
 }
 
 // stopMelComputation_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MmapPlaybackThread::stopMelComputation_l()
+void MmapPlaybackThread::stopMelComputation_l()
 {
     ALOGV("%s: pausing mel processor for thread %d", __func__, id());
     auto melProcessor = mMelProcessor.load();
@@ -10862,7 +10930,7 @@
     }
 }
 
-void AudioFlinger::MmapPlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args)
+void MmapPlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args)
 {
     MmapThread::dumpInternals_l(fd, args);
 
@@ -10871,17 +10939,24 @@
     dprintf(fd, "  Master volume: %f Master mute %d\n", mMasterVolume, mMasterMute);
 }
 
-AudioFlinger::MmapCaptureThread::MmapCaptureThread(
-        const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+/* static */
+sp<IAfMmapCaptureThread> IAfMmapCaptureThread::create(
+        const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
+        AudioHwDevice* hwDev,  AudioStreamIn* input, bool systemReady) {
+    return sp<MmapCaptureThread>::make(afThreadCallback, id, hwDev, input, systemReady);
+}
+
+MmapCaptureThread::MmapCaptureThread(
+        const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
         AudioHwDevice *hwDev,  AudioStreamIn *input, bool systemReady)
-    : MmapThread(audioFlinger, id, hwDev, input->stream, systemReady, false /* isOut */),
+    : MmapThread(afThreadCallback, id, hwDev, input->stream, systemReady, false /* isOut */),
       mInput(input)
 {
     snprintf(mThreadName, kThreadNameLength, "AudioMmapIn_%X", id);
     mChannelCount = audio_channel_count_from_in_mask(mChannelMask);
 }
 
-status_t AudioFlinger::MmapCaptureThread::exitStandby_l()
+status_t MmapCaptureThread::exitStandby_l()
 {
     {
         // mInput might have been cleared by clearInput()
@@ -10892,7 +10967,7 @@
     return MmapThread::exitStandby_l();
 }
 
-AudioFlinger::AudioStreamIn* AudioFlinger::MmapCaptureThread::clearInput()
+AudioStreamIn* MmapCaptureThread::clearInput()
 {
     Mutex::Autolock _l(mLock);
     AudioStreamIn *input = mInput;
@@ -10900,8 +10975,7 @@
     return input;
 }
 
-
-void AudioFlinger::MmapCaptureThread::processVolume_l()
+void MmapCaptureThread::processVolume_l()
 {
     bool changed = false;
     bool silenced = false;
@@ -10928,13 +11002,13 @@
     }
 }
 
-AudioFlinger::ThreadBase::MetadataUpdate AudioFlinger::MmapCaptureThread::updateMetadata_l()
+ThreadBase::MetadataUpdate MmapCaptureThread::updateMetadata_l()
 {
     if (!isStreamInitialized() || !mActiveTracks.readAndClearHasChanged()) {
         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 = {
@@ -10951,7 +11025,7 @@
     return change;
 }
 
-void AudioFlinger::MmapCaptureThread::setRecordSilenced(audio_port_handle_t portId, bool silenced)
+void MmapCaptureThread::setRecordSilenced(audio_port_handle_t portId, bool silenced)
 {
     Mutex::Autolock _l(mLock);
     for (size_t i = 0; i < mActiveTracks.size() ; i++) {
@@ -10963,7 +11037,7 @@
     setClientSilencedIfExists_l(portId, silenced);
 }
 
-void AudioFlinger::MmapCaptureThread::toAudioPortConfig(struct audio_port_config *config)
+void MmapCaptureThread::toAudioPortConfig(struct audio_port_config* config)
 {
     MmapThread::toAudioPortConfig(config);
     if (mInput && mInput->flags != AUDIO_INPUT_FLAG_NONE) {
@@ -10972,8 +11046,8 @@
     }
 }
 
-status_t AudioFlinger::MmapCaptureThread::getExternalPosition(
-        uint64_t *position, int64_t *timeNanos)
+status_t MmapCaptureThread::getExternalPosition(
+        uint64_t* position, int64_t* timeNanos) const
 {
     if (mInput == nullptr) {
         return NO_INIT;
@@ -10983,12 +11057,19 @@
 
 // ----------------------------------------------------------------------------
 
-AudioFlinger::BitPerfectThread::BitPerfectThread(const sp<AudioFlinger> &audioflinger,
-        AudioStreamOut *output, audio_io_handle_t id, bool systemReady)
-        : MixerThread(audioflinger, output, id, systemReady, BIT_PERFECT) {}
+/* static */
+sp<IAfPlaybackThread> IAfPlaybackThread::createBitPerfectThread(
+        const sp<IAfThreadCallback>& afThreadCallback,
+        AudioStreamOut* output, audio_io_handle_t id, bool systemReady) {
+    return sp<BitPerfectThread>::make(afThreadCallback, output, id, systemReady);
+}
 
-AudioFlinger::PlaybackThread::mixer_state AudioFlinger::BitPerfectThread::prepareTracks_l(
-        Vector<sp<Track>> *tracksToRemove) {
+BitPerfectThread::BitPerfectThread(const sp<IAfThreadCallback> &afThreadCallback,
+        AudioStreamOut *output, audio_io_handle_t id, bool systemReady)
+        : MixerThread(afThreadCallback, output, id, systemReady, BIT_PERFECT) {}
+
+PlaybackThread::mixer_state BitPerfectThread::prepareTracks_l(
+        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;
@@ -11020,7 +11101,7 @@
     return result;
 }
 
-void AudioFlinger::BitPerfectThread::threadLoop_mix() {
+void BitPerfectThread::threadLoop_mix() {
     MixerThread::threadLoop_mix();
     mHasDataCopiedToSinkBuffer = mIsBitPerfect;
 }
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index a5c1048..902806d 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -15,38 +15,24 @@
 ** limitations under the License.
 */
 
-#ifndef INCLUDING_FROM_AUDIOFLINGER_H
-    #error This header file should only be included from AudioFlinger.h
-#endif
+#pragma once
 
-public: // TODO(b/288339104) extract out of AudioFlinger class
-class ThreadBase : public Thread {
+namespace android {
+
+class AsyncCallbackThread;
+
+class ThreadBase : public virtual IAfThreadBase, public Thread {
 public:
-
-#include "TrackBase.h"
-
-    enum type_t {
-        MIXER,              // Thread class is MixerThread
-        DIRECT,             // Thread class is DirectOutputThread
-        DUPLICATING,        // Thread class is DuplicatingThread
-        RECORD,             // Thread class is RecordThread
-        OFFLOAD,            // Thread class is OffloadThread
-        MMAP_PLAYBACK,      // Thread class for MMAP playback stream
-        MMAP_CAPTURE,       // Thread class for MMAP capture stream
-        SPATIALIZER,  //
-        BIT_PERFECT,        // Thread class for BitPerfectThread
-        // If you add any values here, also update ThreadBase::threadTypeToString()
-    };
-
     static const char *threadTypeToString(type_t type);
 
-    ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+    IAfThreadCallback* afThreadCallback() const final { return mAfThreadCallback.get(); }
+
+    ThreadBase(const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
                type_t type, bool systemReady, bool isOut);
-    virtual             ~ThreadBase();
+    ~ThreadBase() override;
 
-    virtual status_t    readyToRun();
-
-    void clearPowerManager();
+    status_t readyToRun() final;
+    void clearPowerManager() final;
 
     // base for record and playback
     enum {
@@ -90,8 +76,6 @@
 
     class ConfigEvent: public RefBase {
     public:
-        virtual ~ConfigEvent() {}
-
         void dump(char *buffer, size_t size) {
             snprintf(buffer, size, "Event type: %d\n", mType);
             if (mData != nullptr) {
@@ -135,7 +119,6 @@
             ConfigEvent(CFG_EVENT_IO) {
             mData = new IoConfigEventData(event, pid, portId);
         }
-        virtual ~IoConfigEvent() {}
     };
 
     class PrioConfigEventData : public ConfigEventData {
@@ -160,7 +143,6 @@
             ConfigEvent(CFG_EVENT_PRIO, true) {
             mData = new PrioConfigEventData(pid, tid, prio, forApp);
         }
-        virtual ~PrioConfigEvent() {}
     };
 
     class SetParameterConfigEventData : public ConfigEventData {
@@ -182,7 +164,6 @@
             mData = new SetParameterConfigEventData(keyValuePairs);
             mWaitStatus = true;
         }
-        virtual ~SetParameterConfigEvent() {}
     };
 
     class CreateAudioPatchConfigEventData : public ConfigEventData {
@@ -207,7 +188,6 @@
             mData = new CreateAudioPatchConfigEventData(patch, handle);
             mWaitStatus = true;
         }
-        virtual ~CreateAudioPatchConfigEvent() {}
     };
 
     class ReleaseAudioPatchConfigEventData : public ConfigEventData {
@@ -229,7 +209,6 @@
             mData = new ReleaseAudioPatchConfigEventData(handle);
             mWaitStatus = true;
         }
-        virtual ~ReleaseAudioPatchConfigEvent() {}
     };
 
     class UpdateOutDevicesConfigEventData : public ConfigEventData {
@@ -250,8 +229,6 @@
             ConfigEvent(CFG_EVENT_UPDATE_OUT_DEVICE) {
             mData = new UpdateOutDevicesConfigEventData(outDevices);
         }
-
-        virtual ~UpdateOutDevicesConfigEvent();
     };
 
     class ResizeBufferConfigEventData : public ConfigEventData {
@@ -272,8 +249,6 @@
             ConfigEvent(CFG_EVENT_RESIZE_BUFFER) {
             mData = new ResizeBufferConfigEventData(maxSharedAudioHistoryMs);
         }
-
-        virtual ~ResizeBufferConfigEvent() {}
     };
 
     class CheckOutputStageEffectsEvent : public ConfigEvent {
@@ -281,8 +256,6 @@
         CheckOutputStageEffectsEvent() :
             ConfigEvent(CFG_EVENT_CHECK_OUTPUT_STAGE_EFFECTS) {
         }
-
-        virtual ~CheckOutputStageEffectsEvent() {}
     };
 
     class HalLatencyModesChangedEvent : public ConfigEvent {
@@ -290,8 +263,6 @@
         HalLatencyModesChangedEvent() :
             ConfigEvent(CFG_EVENT_HAL_LATENCY_MODES_CHANGED) {
         }
-
-        virtual ~HalLatencyModesChangedEvent() {}
     };
 
 
@@ -309,108 +280,87 @@
         wp<ThreadBase> mThread;
     };
 
-    virtual     status_t    initCheck() const = 0;
+    type_t type() const final { return mType; }
+    bool isDuplicating() const final { return (mType == DUPLICATING); }
+    audio_io_handle_t id() const final { return mId;}
 
-                // static externally-visible
-                type_t      type() const { return mType; }
-                bool isDuplicating() const { return (mType == DUPLICATING); }
-
-                audio_io_handle_t id() const { return mId;}
-
-                // dynamic externally-visible
-                uint32_t    sampleRate() const { return mSampleRate; }
-                audio_channel_mask_t channelMask() const { return mChannelMask; }
-    virtual     audio_channel_mask_t mixerChannelMask() const { return mChannelMask; }
-
-                audio_format_t format() const { return mHALFormat; }
-                uint32_t channelCount() const { return mChannelCount; }
-
-                // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects,
-                // and returns the [normal mix] buffer's frame count.
-    virtual     size_t      frameCount() const = 0;
-    virtual     audio_channel_mask_t hapticChannelMask() const { return AUDIO_CHANNEL_NONE; }
-    virtual     uint32_t    latency_l() const { return 0; }
-    virtual     void        setVolumeForOutput_l(float left __unused, float right __unused) const {}
+    uint32_t sampleRate() const final { return mSampleRate; }
+    audio_channel_mask_t channelMask() const final { return mChannelMask; }
+    audio_channel_mask_t mixerChannelMask() const override { return mChannelMask; }
+    audio_format_t format() const final { return mHALFormat; }
+    uint32_t channelCount() const final { return mChannelCount; }
+    audio_channel_mask_t hapticChannelMask() const override { return AUDIO_CHANNEL_NONE; }
+    uint32_t hapticChannelCount() const override { return 0; }
+    uint32_t latency_l() const override { return 0; }
+    void setVolumeForOutput_l(float /* left */, float /* right */) const override {}
 
                 // Return's the HAL's frame count i.e. fast mixer buffer size.
-                size_t      frameCountHAL() const { return mFrameCount; }
-
-                size_t      frameSize() const { return mFrameSize; }
+    size_t frameCountHAL() const final { return mFrameCount; }
+    size_t frameSize() const final { return mFrameSize; }
 
     // Should be "virtual status_t requestExitAndWait()" and override same
     // method in Thread, but Thread::requestExitAndWait() is not yet virtual.
-                void        exit();
-    virtual     bool        checkForNewParameter_l(const String8& keyValuePair,
-                                                    status_t& status) = 0;
-    virtual     status_t    setParameters(const String8& keyValuePairs);
-    virtual     String8     getParameters(const String8& keys) = 0;
-    virtual     void        ioConfigChanged(audio_io_config_event_t event, pid_t pid = 0,
-                                        audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) = 0;
+    void exit() final;
+    status_t setParameters(const String8& keyValuePairs) final;
+
                 // sendConfigEvent_l() must be called with ThreadBase::mLock held
                 // Can temporarily release the lock if waiting for a reply from
                 // processConfigEvents_l().
-                status_t    sendConfigEvent_l(sp<ConfigEvent>& event);
-                void        sendIoConfigEvent(audio_io_config_event_t event, pid_t pid = 0,
-                                              audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
-                void        sendIoConfigEvent_l(audio_io_config_event_t event, pid_t pid = 0,
-                                            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
-                void        sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp);
-                void        sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio, bool forApp);
-                status_t    sendSetParameterConfigEvent_l(const String8& keyValuePair);
-                status_t    sendCreateAudioPatchConfigEvent(const struct audio_patch *patch,
-                                                            audio_patch_handle_t *handle);
-                status_t    sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle);
-                status_t    sendUpdateOutDeviceConfigEvent(
-                                    const DeviceDescriptorBaseVector& outDevices);
-                void        sendResizeBufferConfigEvent_l(int32_t maxSharedAudioHistoryMs);
-                void        sendCheckOutputStageEffectsEvent();
-                void        sendCheckOutputStageEffectsEvent_l();
-                void        sendHalLatencyModesChangedEvent_l();
+    status_t sendConfigEvent_l(sp<ConfigEvent>& event);
+    void sendIoConfigEvent(audio_io_config_event_t event, pid_t pid = 0,
+            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final;
+    void sendIoConfigEvent_l(audio_io_config_event_t event, pid_t pid = 0,
+            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final;
+    void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp) final;
+    void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio, bool forApp) final;
+    status_t sendSetParameterConfigEvent_l(const String8& keyValuePair) final;
+    status_t sendCreateAudioPatchConfigEvent(const struct audio_patch* patch,
+            audio_patch_handle_t* handle) final;
+    status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle) final;
+    status_t sendUpdateOutDeviceConfigEvent(
+            const DeviceDescriptorBaseVector& outDevices) final;
+    void sendResizeBufferConfigEvent_l(int32_t maxSharedAudioHistoryMs) final;
+    void sendCheckOutputStageEffectsEvent() final;
+    void sendCheckOutputStageEffectsEvent_l() final;
+    void sendHalLatencyModesChangedEvent_l() final;
 
-                void        processConfigEvents_l();
-    virtual     void        setCheckOutputStageEffects() {}
-    virtual     void        cacheParameters_l() = 0;
-    virtual     status_t    createAudioPatch_l(const struct audio_patch *patch,
-                                               audio_patch_handle_t *handle) = 0;
-    virtual     status_t    releaseAudioPatch_l(const audio_patch_handle_t handle) = 0;
-    virtual     void        updateOutDevices(const DeviceDescriptorBaseVector& outDevices);
-    virtual     void        toAudioPortConfig(struct audio_port_config *config) = 0;
+    void processConfigEvents_l() final;
+    void setCheckOutputStageEffects() override {}
+    void updateOutDevices(const DeviceDescriptorBaseVector& outDevices) override;
+    void toAudioPortConfig(struct audio_port_config* config) override;
+    void resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs) override;
 
-    virtual     void        resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs);
+    // see note at declaration of mStandby, mOutDevice and mInDevice
+    bool inStandby() const override { return mStandby; }
+    const DeviceTypeSet outDeviceTypes() const final {
+        return getAudioDeviceTypes(mOutDeviceTypeAddrs);
+    }
+    audio_devices_t inDeviceType() const final { return mInDeviceTypeAddr.mType; }
+    DeviceTypeSet getDeviceTypes() const final {
+        return isOutput() ? outDeviceTypes() : DeviceTypeSet({inDeviceType()});
+    }
 
-                // see note at declaration of mStandby, mOutDevice and mInDevice
-                bool        standby() const { return mStandby; }
-                const DeviceTypeSet outDeviceTypes() const {
-                    return getAudioDeviceTypes(mOutDeviceTypeAddrs);
-                }
-                audio_devices_t inDeviceType() const { return mInDeviceTypeAddr.mType; }
-                DeviceTypeSet getDeviceTypes() const {
-                    return isOutput() ? outDeviceTypes() : DeviceTypeSet({inDeviceType()});
-                }
+    const AudioDeviceTypeAddrVector& outDeviceTypeAddrs() const final {
+        return mOutDeviceTypeAddrs;
+    }
+    const AudioDeviceTypeAddr& inDeviceTypeAddr() const final {
+        return mInDeviceTypeAddr;
+    }
 
-                const AudioDeviceTypeAddrVector& outDeviceTypeAddrs() const {
-                    return mOutDeviceTypeAddrs;
-                }
-                const AudioDeviceTypeAddr& inDeviceTypeAddr() const {
-                    return mInDeviceTypeAddr;
-                }
+    bool isOutput() const final { return mIsOut; }
 
-                bool        isOutput() const { return mIsOut; }
+    bool isOffloadOrMmap() const final {
+        switch (mType) {
+        case OFFLOAD:
+        case MMAP_PLAYBACK:
+        case MMAP_CAPTURE:
+            return true;
+        default:
+            return false;
+        }
+    }
 
-                bool        isOffloadOrMmap() const {
-                    switch (mType) {
-                    case OFFLOAD:
-                    case MMAP_PLAYBACK:
-                    case MMAP_CAPTURE:
-                        return true;
-                    default:
-                        return false;
-                    }
-                }
-
-    virtual     sp<StreamHalInterface> stream() const = 0;
-
-                sp<IAfEffectHandle> createEffect_l(
+    sp<IAfEffectHandle> createEffect_l(
                                     const sp<Client>& client,
                                     const sp<media::IEffectClient>& effectClient,
                                     int32_t priority,
@@ -420,7 +370,7 @@
                                     status_t *status /*non-NULL*/,
                                     bool pinned,
                                     bool probe,
-                                    bool notifyFramesProcessed);
+                                    bool notifyFramesProcessed) final;
 
                 // return values for hasAudioSession (bit field)
                 enum effect_state {
@@ -436,47 +386,40 @@
                                                // bit-perfect track
                 };
 
-                // get effect chain corresponding to session Id.
-                sp<IAfEffectChain> getEffectChain(audio_session_t sessionId);
-                // same as getEffectChain() but must be called with ThreadBase mutex locked
-                sp<IAfEffectChain> getEffectChain_l(audio_session_t sessionId) const;
-                std::vector<int> getEffectIds_l(audio_session_t sessionId);
-                // add an effect chain to the chain list (mEffectChains)
-    virtual     status_t addEffectChain_l(const sp<IAfEffectChain>& chain) = 0;
-                // remove an effect chain from the chain list (mEffectChains)
-    virtual     size_t removeEffectChain_l(const sp<IAfEffectChain>& chain) = 0;
+    // get effect chain corresponding to session Id.
+    sp<IAfEffectChain> getEffectChain(audio_session_t sessionId) const final;
+    // same as getEffectChain() but must be called with ThreadBase mutex locked
+    sp<IAfEffectChain> getEffectChain_l(audio_session_t sessionId) const final;
+    std::vector<int> getEffectIds_l(audio_session_t sessionId) const final;
+
                 // lock all effect chains Mutexes. Must be called before releasing the
                 // ThreadBase mutex before processing the mixer and effects. This guarantees the
                 // integrity of the chains during the process.
                 // Also sets the parameter 'effectChains' to current value of mEffectChains.
-                void lockEffectChains_l(Vector<sp<IAfEffectChain>>& effectChains);
+    void lockEffectChains_l(Vector<sp<IAfEffectChain>>& effectChains) final;
                 // unlock effect chains after process
-                void unlockEffectChains(const Vector<sp<IAfEffectChain>>& effectChains);
+    void unlockEffectChains(const Vector<sp<IAfEffectChain>>& effectChains) final;
                 // get a copy of mEffectChains vector
-                Vector<sp<IAfEffectChain>> getEffectChains_l() const { return mEffectChains; };
+    Vector<sp<IAfEffectChain>> getEffectChains_l() const final { return mEffectChains; };
                 // set audio mode to all effect chains
-                void setMode(audio_mode_t mode);
+    void setMode(audio_mode_t mode) final;
                 // get effect module with corresponding ID on specified audio session
-                sp<IAfEffectModule> getEffect(audio_session_t sessionId, int effectId);
-                sp<IAfEffectModule> getEffect_l(audio_session_t sessionId, int effectId);
+    sp<IAfEffectModule> getEffect(audio_session_t sessionId, int effectId) const final;
+    sp<IAfEffectModule> getEffect_l(audio_session_t sessionId, int effectId) const final;
                 // add and effect module. Also creates the effect chain is none exists for
                 // the effects audio session. Only called in a context of moving an effect
                 // from one thread to another
-                status_t addEffect_l(const sp<IAfEffectModule>& effect);
+    status_t addEffect_l(const sp<IAfEffectModule>& effect) final;
                 // remove and effect module. Also removes the effect chain is this was the last
                 // effect
-                void removeEffect_l(const sp<IAfEffectModule>& effect, bool release = false);
+    void removeEffect_l(const sp<IAfEffectModule>& effect, bool release = false) final;
                 // disconnect an effect handle from module and destroy module if last handle
-                void disconnectEffectHandle(IAfEffectHandle *handle, bool unpinIfLast);
+    void disconnectEffectHandle(IAfEffectHandle* handle, bool unpinIfLast) final;
                 // detach all tracks connected to an auxiliary effect
-    virtual     void detachAuxEffect_l(int effectId __unused) {}
-                // returns a combination of:
-                // - EFFECT_SESSION if effects on this audio session exist in one chain
-                // - TRACK_SESSION if tracks on this audio session exist
-                // - FAST_SESSION if fast tracks on this audio session exist
-                // - SPATIALIZED_SESSION if spatialized tracks on this audio session exist
-    virtual     uint32_t hasAudioSession_l(audio_session_t sessionId) const = 0;
-                uint32_t hasAudioSession(audio_session_t sessionId) const {
+    void detachAuxEffect_l(int /* effectId */) override {}
+    // TODO(b/291317898) - remove hasAudioSession_l below.
+    uint32_t hasAudioSession_l(audio_session_t sessionId) const override = 0;
+    uint32_t hasAudioSession(audio_session_t sessionId) const final {
                     Mutex::Autolock _l(mLock);
                     return hasAudioSession_l(sessionId);
                 }
@@ -488,7 +431,7 @@
                         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()) {
@@ -510,19 +453,17 @@
 
                 // the value returned by default implementation is not important as the
                 // strategy is only meaningful for PlaybackThread which implements this method
-                virtual product_strategy_t getStrategyForSession_l(
-                        audio_session_t sessionId __unused) {
+    product_strategy_t getStrategyForSession_l(
+            audio_session_t /* sessionId */) const override {
                     return static_cast<product_strategy_t>(0);
                 }
 
                 // check if some effects must be suspended/restored when an effect is enabled
                 // or disabled
-                void checkSuspendOnEffectEnabled(bool enabled,
+    void checkSuspendOnEffectEnabled(bool enabled,
                                                  audio_session_t sessionId,
-                                                 bool threadLocked);
+                                                 bool threadLocked) final;
 
-                virtual status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) = 0;
-                virtual bool isValidSyncEvent(const sp<audioflinger::SyncEvent>& event) const = 0;
 
                 // Return a reference to a per-thread heap which can be used to allocate IMemory
                 // objects that will be read-only to client processes, read/write to mediaserver,
@@ -530,36 +471,35 @@
                 // The heap is per-thread rather than common across all threads, because
                 // clients can't be trusted not to modify the offset of the IMemory they receive.
                 // If a thread does not have such a heap, this method returns 0.
-                virtual sp<MemoryDealer>    readOnlyHeap() const { return 0; }
+    sp<MemoryDealer> readOnlyHeap() const override { return nullptr; }
 
-                virtual sp<IMemory> pipeMemory() const { return 0; }
+    sp<IMemory> pipeMemory() const override { return nullptr; }
 
-                        void systemReady();
+    void systemReady() final;
 
-                // checkEffectCompatibility_l() must be called with ThreadBase::mLock held
-                virtual status_t    checkEffectCompatibility_l(const effect_descriptor_t *desc,
-                                                               audio_session_t sessionId) = 0;
+    void broadcast_l() final;
 
-                        void        broadcast_l();
+    bool isTimestampCorrectionEnabled() const override { return false; }
 
-                virtual bool        isTimestampCorrectionEnabled() const { return false; }
+    bool isMsdDevice() const final { return mIsMsdDevice; }
 
-                bool                isMsdDevice() const { return mIsMsdDevice; }
-
-                void                dump(int fd, const Vector<String16>& args);
+    void dump(int fd, const Vector<String16>& args) override;
 
                 // deliver stats to mediametrics.
-                void                sendStatistics(bool force);
+    void sendStatistics(bool force) final;
 
+    Mutex& mutex() const final {
+        return mLock;
+    }
     mutable     Mutex                   mLock;
 
-                void onEffectEnable(const sp<IAfEffectModule>& effect);
-                void onEffectDisable();
+    void onEffectEnable(const sp<IAfEffectModule>& effect) final;
+    void onEffectDisable() final;
 
                 // invalidateTracksForAudioSession_l must be called with holding mLock.
-    virtual     void invalidateTracksForAudioSession_l(audio_session_t sessionId __unused) const { }
+    void invalidateTracksForAudioSession_l(audio_session_t /* sessionId */) const override {}
                 // Invalidate all the tracks with the given audio session.
-                void invalidateTracksForAudioSession(audio_session_t sessionId) const {
+    void invalidateTracksForAudioSession(audio_session_t sessionId) const final {
                     Mutex::Autolock _l(mLock);
                     invalidateTracksForAudioSession_l(sessionId);
                 }
@@ -568,17 +508,15 @@
                 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();
                         }
                     }
                 }
 
-    virtual     bool isStreamInitialized() = 0;
-
-    virtual     void startMelComputation_l(const sp<audio_utils::MelProcessor>& processor);
-    virtual     void stopMelComputation_l();
+    void startMelComputation_l(const sp<audio_utils::MelProcessor>& processor) override;
+    void stopMelComputation_l() override;
 
 protected:
 
@@ -602,7 +540,7 @@
                 // occurs when all suspend requests are cancelled.
                 void setEffectSuspended_l(const effect_uuid_t *type,
                                           bool suspend,
-                                          audio_session_t sessionId);
+                                          audio_session_t sessionId) final;
                 // updated mSuspendedSessions when an effect is suspended or restored
                 void        updateSuspendedSessions_l(const effect_uuid_t *type,
                                                       bool suspend,
@@ -629,7 +567,7 @@
                                 return INVALID_OPERATION;
                             }
 public:
-// TODO(b/288339104) organize with publics
+// TODO(b/291317898) organize with publics
                 product_strategy_t getStrategyForStream(audio_stream_type_t stream) const;
 protected:
 
@@ -639,15 +577,12 @@
                             { }
     virtual     void        dumpTracks_l(int fd __unused, const Vector<String16>& args __unused) { }
 
-
-    friend class AudioFlinger;      // for mEffectChains and mAudioManager
-
                 const type_t            mType;
 
                 // Used by parameters, config events, addTrack_l, exit
                 Condition               mWaitWorkCV;
 
-                const sp<AudioFlinger>  mAudioFlinger;
+                const sp<IAfThreadCallback>  mAfThreadCallback;
                 ThreadMetrics           mThreadMetrics;
                 const bool              mIsOut;
 
@@ -784,7 +719,7 @@
                     bool            isEmpty() const {
                         return mActiveTracks.isEmpty();
                     }
-                    ssize_t         indexOf(const sp<T>& item) {
+                    ssize_t indexOf(const sp<T>& item) const {
                         return mActiveTracks.indexOf(item);
                     }
                     sp<T>           operator[](size_t index) const {
@@ -829,8 +764,6 @@
                         return wakeLockUids; // moved by underlying SharedBuffer
                     }
 
-                    std::map<uid_t, std::pair<ssize_t /* previous */, ssize_t /* current */>>
-                                        mBatteryCounter;
                     SortedVector<sp<T>> mActiveTracks;
                     int                 mActiveTracksGeneration;
                     int                 mLastActiveTracksGeneration;
@@ -847,35 +780,14 @@
                 void dumpEffectChains_l(int fd, const Vector<String16>& args);
 };
 
-class VolumeInterface {
- public:
-
-    virtual ~VolumeInterface() {}
-
-    virtual void        setMasterVolume(float value) = 0;
-    virtual void        setMasterMute(bool muted) = 0;
-    virtual void        setStreamVolume(audio_stream_type_t stream, float value) = 0;
-    virtual void        setStreamMute(audio_stream_type_t stream, bool muted) = 0;
-    virtual float       streamVolume(audio_stream_type_t stream) const = 0;
-
-};
-
 // --- PlaybackThread ---
-class PlaybackThread : public ThreadBase, public StreamOutHalInterfaceCallback,
-                       public VolumeInterface, public StreamOutHalInterfaceEventCallback {
+class PlaybackThread : public ThreadBase, public virtual IAfPlaybackThread,
+                       public StreamOutHalInterfaceCallback,
+                       public virtual VolumeInterface, public StreamOutHalInterfaceEventCallback {
 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
-        MIXER_TRACKS_READY,      // at least one active track, and at least one track has data
-        MIXER_DRAIN_TRACK,      // drain currently playing track
-        MIXER_DRAIN_ALL,        // fully drain the hardware
-        // standby mode does not have an enum value
-        // suspend by audio policy manager is orthogonal to mixer state
-    };
+    sp<IAfPlaybackThread> asIAfPlaybackThread() final {
+        return sp<IAfPlaybackThread>::fromExisting(this);
+    }
 
     // retry count before removing active track in case of underrun on offloaded thread:
     // we need to make sure that AudioTrack client has enough time to send large buffers
@@ -883,7 +795,6 @@
     // handled for offloaded tracks
     static const int8_t kMaxTrackRetriesOffload = 20;
     static const int8_t kMaxTrackStartupRetriesOffload = 100;
-    static const int8_t kMaxTrackStopRetriesOffload = 2;
     static constexpr uint32_t kMaxTracksPerUid = 40;
     static constexpr size_t kMaxTracks = 256;
 
@@ -893,19 +804,23 @@
     // for initial conditions or large delays.
     static const nsecs_t kMaxNextBufferDelayNs = 100000000;
 
-    PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+    PlaybackThread(const sp<IAfThreadCallback>& afThreadCallback, AudioStreamOut* output,
                    audio_io_handle_t id, type_t type, bool systemReady,
                    audio_config_base_t *mixerConfig = nullptr);
-    virtual             ~PlaybackThread();
+    ~PlaybackThread() override;
 
     // Thread virtuals
-    virtual     bool        threadLoop();
+    bool threadLoop() final;
 
     // RefBase
-    virtual     void        onFirstRef();
+    void onFirstRef() override;
 
-    virtual     status_t    checkEffectCompatibility_l(const effect_descriptor_t *desc,
-                                                       audio_session_t sessionId);
+    status_t checkEffectCompatibility_l(
+            const effect_descriptor_t* desc, audio_session_t sessionId) final;
+
+    void addOutputTrack_l(const sp<IAfTrack>& track) final {
+        mTracks.add(track);
+    }
 
 protected:
     // Code snippets that were lifted up out of threadLoop()
@@ -915,14 +830,14 @@
     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
@@ -930,18 +845,21 @@
     virtual     void        onDrainReady();
     virtual     void        onError();
 
+public: // AsyncCallbackThread
                 void        resetWriteBlocked(uint32_t sequence);
                 void        resetDraining(uint32_t sequence);
+protected:
 
     virtual     bool        waitingAsyncCallback();
     virtual     bool        waitingAsyncCallback_l();
     virtual     bool        shouldStandby_l();
     virtual     void        onAddNewTrack_l();
+public:  // AsyncCallbackThread
                 void        onAsyncError(); // error reported by AsyncCallbackThread
-
+protected:
     // StreamHalInterfaceCodecFormatCallback implementation
                 void        onCodecFormatChanged(
-                                const std::basic_string<uint8_t>& metadataBs) override;
+            const std::basic_string<uint8_t>& metadataBs) final;
 
     // ThreadBase virtuals
     virtual     void        preExit();
@@ -956,29 +874,28 @@
     virtual     void        setHalLatencyMode_l() {}
 
 
-                void        dumpInternals_l(int fd, const Vector<String16>& args) override;
-                void        dumpTracks_l(int fd, const Vector<String16>& args) override;
+    void dumpInternals_l(int fd, const Vector<String16>& args) override;
+    void dumpTracks_l(int fd, const Vector<String16>& args) final;
 
 public:
 
-    virtual     status_t    initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; }
+    status_t initCheck() const final { return mOutput == nullptr ? NO_INIT : NO_ERROR; }
 
                 // return estimated latency in milliseconds, as reported by HAL
-                uint32_t    latency() const;
+    uint32_t latency() const final;
                 // same, but lock must already be held
-                uint32_t    latency_l() const override;
+    uint32_t latency_l() const final;
 
                 // VolumeInterface
-    virtual     void        setMasterVolume(float value);
-    virtual     void        setMasterBalance(float balance);
-    virtual     void        setMasterMute(bool muted);
-    virtual     void        setStreamVolume(audio_stream_type_t stream, float value);
-    virtual     void        setStreamMute(audio_stream_type_t stream, bool muted);
-    virtual     float       streamVolume(audio_stream_type_t stream) const;
+    void setMasterVolume(float value) final;
+    void setMasterBalance(float balance) override;
+    void setMasterMute(bool muted) final;
+    void setStreamVolume(audio_stream_type_t stream, float value) final;
+    void setStreamMute(audio_stream_type_t stream, bool muted) final;
+    float streamVolume(audio_stream_type_t stream) const final;
+    void setVolumeForOutput_l(float left, float right) const final;
 
-                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,
@@ -999,15 +916,20 @@
                                 audio_port_handle_t portId,
                                 const sp<media::IAudioTrackCallback>& callback,
                                 bool isSpatialized,
-                                bool isBitPerfect);
+                                bool isBitPerfect) final;
 
-                AudioStreamOut* getOutput() const;
-                AudioStreamOut* clearOutput();
-                virtual sp<StreamHalInterface> stream() const;
+    bool isTrackActive(const sp<IAfTrack>& track) const final {
+        return mActiveTracks.indexOf(track) >= 0;
+    }
+
+    AudioStreamOut* getOutput_l() const final { return mOutput; }
+    AudioStreamOut* getOutput() const final;
+    AudioStreamOut* clearOutput() final;
+    sp<StreamHalInterface> stream() const final;
 
                 // a very large number of suspend() will eventually wraparound, but unlikely
-                void        suspend() { (void) android_atomic_inc(&mSuspended); }
-                void        restore()
+    void suspend() final { (void) android_atomic_inc(&mSuspended); }
+    void restore() final
                                 {
                                     // if restore() is done without suspend(), get back into
                                     // range so that the next suspend() will operate correctly
@@ -1015,123 +937,127 @@
                                         android_atomic_release_store(0, &mSuspended);
                                     }
                                 }
-                bool        isSuspended() const
+    bool isSuspended() const final
                                 { return android_atomic_acquire_load(&mSuspended) > 0; }
 
-    virtual     String8     getParameters(const String8& keys);
-    virtual     void        ioConfigChanged(audio_io_config_event_t event, pid_t pid = 0,
-                                            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
-                status_t    getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
+    String8 getParameters(const String8& keys);
+    void ioConfigChanged(audio_io_config_event_t event, pid_t pid = 0,
+            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final;
+    status_t getRenderPosition(uint32_t* halFrames, uint32_t* dspFrames) const final;
                 // Consider also removing and passing an explicit mMainBuffer initialization
-                // parameter to AF::PlaybackThread::Track::Track().
-                float *sinkBuffer() const {
+                // parameter to AF::IAfTrack::Track().
+    float* sinkBuffer() const final {
                     return reinterpret_cast<float *>(mSinkBuffer); };
 
-    virtual     void detachAuxEffect_l(int effectId);
-                status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track>& track,
-                        int EffectId);
-                status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track>& track,
-                        int EffectId);
+    void detachAuxEffect_l(int effectId) final;
 
-                virtual status_t addEffectChain_l(const sp<IAfEffectChain>& chain);
-                virtual size_t removeEffectChain_l(const sp<IAfEffectChain>& chain);
-                        uint32_t hasAudioSession_l(audio_session_t sessionId) const override {
+    status_t attachAuxEffect(const sp<IAfTrack>& track, int EffectId) final;
+    status_t attachAuxEffect_l(const sp<IAfTrack>& track, int EffectId) final;
+
+    status_t addEffectChain_l(const sp<IAfEffectChain>& chain) final;
+    size_t removeEffectChain_l(const sp<IAfEffectChain>& chain) final;
+    uint32_t hasAudioSession_l(audio_session_t sessionId) const final {
                             return ThreadBase::hasAudioSession_l(sessionId, mTracks);
                         }
-                virtual product_strategy_t getStrategyForSession_l(audio_session_t sessionId);
+    product_strategy_t getStrategyForSession_l(audio_session_t sessionId) const final;
 
 
-                status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) override;
-                bool     isValidSyncEvent(const sp<audioflinger::SyncEvent>& event) const override;
+    status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) final;
+    bool isValidSyncEvent(const sp<audioflinger::SyncEvent>& event) const final;
 
                 // called with AudioFlinger lock held
-                        bool     invalidateTracks_l(audio_stream_type_t streamType);
-                        bool     invalidateTracks_l(std::set<audio_port_handle_t>& portIds);
-                virtual void     invalidateTracks(audio_stream_type_t streamType);
+    bool invalidateTracks_l(audio_stream_type_t streamType) final;
+    bool invalidateTracks_l(std::set<audio_port_handle_t>& portIds) final;
+    void invalidateTracks(audio_stream_type_t streamType) override;
                 // Invalidate tracks by a set of port ids. The port id will be removed from
                 // the given set if the corresponding track is found and invalidated.
-                virtual void     invalidateTracks(std::set<audio_port_handle_t>& portIds);
+    void invalidateTracks(std::set<audio_port_handle_t>& portIds) override;
 
-    virtual     size_t      frameCount() const { return mNormalFrameCount; }
+    size_t frameCount() const final { return mNormalFrameCount; }
 
-                audio_channel_mask_t mixerChannelMask() const override {
+    audio_channel_mask_t mixerChannelMask() const final {
                     return mMixerChannelMask;
                 }
 
-                status_t    getTimestamp_l(AudioTimestamp& timestamp);
+    status_t getTimestamp_l(AudioTimestamp& timestamp) final;
 
-                void        addPatchTrack(const sp<PatchTrack>& track);
-                void        deletePatchTrack(const sp<PatchTrack>& track);
+    void addPatchTrack(const sp<IAfPatchTrack>& track) final;
+    void deletePatchTrack(const sp<IAfPatchTrack>& track) final;
 
-    virtual     void        toAudioPortConfig(struct audio_port_config *config);
+    void toAudioPortConfig(struct audio_port_config* config) final;
 
                 // Return the asynchronous signal wait time.
-    virtual     int64_t     computeWaitTimeNs_l() const { return INT64_MAX; }
+    int64_t computeWaitTimeNs_l() const override { return INT64_MAX; }
                 // returns true if the track is allowed to be added to the thread.
-    virtual     bool        isTrackAllowed_l(
+    bool isTrackAllowed_l(
                                     audio_channel_mask_t channelMask __unused,
                                     audio_format_t format __unused,
                                     audio_session_t sessionId __unused,
-                                    uid_t uid) const {
+                                    uid_t uid) const override {
                                 return trackCountForUid_l(uid) < PlaybackThread::kMaxTracksPerUid
                                        && mTracks.size() < PlaybackThread::kMaxTracks;
                             }
 
-                bool        isTimestampCorrectionEnabled() const override {
+    bool isTimestampCorrectionEnabled() const final {
                                 return audio_is_output_devices(mTimestampCorrectedDevice)
                                         && outDeviceTypes().count(mTimestampCorrectedDevice) != 0;
                             }
 
-    virtual     bool        isStreamInitialized() {
+    bool isStreamInitialized() const final {
                                 return !(mOutput == nullptr || mOutput->stream == nullptr);
                             }
 
-                audio_channel_mask_t hapticChannelMask() const override {
+    audio_channel_mask_t hapticChannelMask() const final {
                                          return mHapticChannelMask;
                                      }
-                bool supportsHapticPlayback() const {
+
+    uint32_t hapticChannelCount() const final {
+        return mHapticChannelCount;
+    }
+
+    bool supportsHapticPlayback() const final {
                     return (mHapticChannelMask & AUDIO_CHANNEL_HAPTIC_ALL) != AUDIO_CHANNEL_NONE;
                 }
 
-                void setDownStreamPatch(const struct audio_patch *patch) {
+    void setDownStreamPatch(const struct audio_patch* patch) final {
                     Mutex::Autolock _l(mLock);
                     mDownStreamPatch = *patch;
                 }
 
-                PlaybackThread::Track* getTrackById_l(audio_port_handle_t trackId);
+    IAfTrack* getTrackById_l(audio_port_handle_t trackId) final;
 
-                bool hasMixer() const {
+    bool hasMixer() const final {
                     return mType == MIXER || mType == DUPLICATING || mType == SPATIALIZER;
                 }
 
-    virtual     status_t setRequestedLatencyMode(
-            audio_latency_mode_t mode __unused) { return INVALID_OPERATION; }
+    status_t setRequestedLatencyMode(
+            audio_latency_mode_t /* mode */) override { return INVALID_OPERATION; }
 
-    virtual     status_t getSupportedLatencyModes(
-                        std::vector<audio_latency_mode_t>* modes __unused) {
+    status_t getSupportedLatencyModes(
+            std::vector<audio_latency_mode_t>* /* modes */) override {
                     return INVALID_OPERATION;
                 }
 
-    virtual     status_t setBluetoothVariableLatencyEnabled(bool enabled __unused) {
+    status_t setBluetoothVariableLatencyEnabled(bool /* enabled */) override{
                     return INVALID_OPERATION;
                 }
 
-                void startMelComputation_l(const sp<audio_utils::MelProcessor>& processor) override;
-                void stopMelComputation_l() override;
+    void startMelComputation_l(const sp<audio_utils::MelProcessor>& processor) override;
+    void stopMelComputation_l() override;
 
-                void setStandby() {
+    void setStandby() final {
                     Mutex::Autolock _l(mLock);
                     setStandby_l();
                 }
 
-                void setStandby_l() {
+    void setStandby_l() final {
                     mStandby = true;
                     mHalStarted = false;
                     mKernelPositionOnStandby =
                         mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
                 }
 
-                bool waitForHalStart() {
+    bool waitForHalStart() final {
                     Mutex::Autolock _l(mLock);
                     static const nsecs_t kWaitHalTimeoutNs = seconds(2);
                     nsecs_t endWaitTimetNs = systemTime() + kWaitHalTimeoutNs;
@@ -1246,7 +1172,6 @@
 
     audio_channel_mask_t            mMixerChannelMask = AUDIO_CHANNEL_NONE;
 
-private:
     // mMasterMute is in both PlaybackThread and in AudioFlinger.  When a
     // PlaybackThread needs to find out if master-muted, it checks it's local
     // copy rather than the one in AudioFlinger.  This optimization saves a lock.
@@ -1260,8 +1185,7 @@
                             : mTimestampVerifier.DISCONTINUITY_MODE_CONTINUOUS;
                 }
 
-protected:
-    ActiveTracks<Track>     mActiveTracks;
+    ActiveTracks<IAfTrack> mActiveTracks;
 
     // Time to sleep between cycles when:
     virtual uint32_t        activeSleepTimeUs() const;      // mixer state MIXER_TRACKS_ENABLED
@@ -1271,7 +1195,7 @@
     // No sleep in standby mode; waits on a condition
 
     // Code snippets that are temporarily lifted up out of threadLoop() until the merge
-                void        checkSilentMode_l();
+    virtual void checkSilentMode_l() final;  // consider unification with MMapThread
 
     // Non-trivial for DUPLICATING only
     virtual     void        saveOutputTracks() { }
@@ -1289,26 +1213,23 @@
                                    audio_patch_handle_t *handle);
     virtual     status_t    releaseAudioPatch_l(const audio_patch_handle_t handle);
 
-                bool        usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL)
+    bool usesHwAvSync() const final { return mType == DIRECT && mOutput != nullptr
                                     && mHwSupportsPause
                                     && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); }
 
                 uint32_t    trackCountForUid_l(uid_t uid) const;
 
                 void        invalidateTracksForAudioSession_l(
-                                    audio_session_t sessionId) const override {
+    audio_session_t sessionId) const override {
                                 ThreadBase::invalidateTracksForAudioSession_l(sessionId, mTracks);
                             }
 
-private:
-
-    friend class AudioFlinger;      // for numerous
-
     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) final;
+    bool destroyTrack_l(const sp<IAfTrack>& track) final;
+
+    void        removeTrack_l(const sp<IAfTrack>& track);
 
     void        readOutputParameters_l();
     MetadataUpdate          updateMetadata_l() final;
@@ -1366,9 +1287,10 @@
         SortedVector<sp<T>> mTracks; // wrapped SortedVector.
     };
 
-    Tracks<Track>                   mTracks;
+    Tracks<IAfTrack>                   mTracks;
 
     stream_type_t                   mStreamTypes[AUDIO_STREAM_CNT];
+
     AudioStreamOut                  *mOutput;
 
     float                           mMasterVolume;
@@ -1423,19 +1345,20 @@
     // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence
     // callbacks are ignored.
     uint32_t                        mDrainSequence;
+
     sp<AsyncCallbackThread>         mCallbackThread;
 
     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
     sp<NBAIO_Sink>          mOutputSink;
     // If a fast mixer is present, the blocking pipe sink, otherwise clear
     sp<NBAIO_Sink>          mPipeSink;
     // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink
     sp<NBAIO_Sink>          mNormalSink;
+
     uint32_t                mScreenState;   // cached copy of gScreenState
     // TODO: add comment and adjust size as needed
     static const size_t     kFastMixerLogSize = 8 * 1024;
@@ -1453,14 +1376,14 @@
     int64_t                  mKernelPositionOnStandby = 0;
 
 public:
-    virtual     bool        hasFastMixer() const = 0;
-    virtual     FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const
-                                { FastTrackUnderruns dummy; return dummy; }
-                const std::atomic<int64_t>& framesWritten() const { return mFramesWritten; }
+    FastTrackUnderruns getFastTrackUnderruns(size_t /* fastIndex */) const override
+        { return {}; }
+    const std::atomic<int64_t>& framesWritten() const final { return mFramesWritten; }
 
 protected:
                 // accessed by both binder threads and within threadLoop(), lock on mutex needed
-                unsigned    mFastTrackAvailMask;    // bit i set if fast track [i] is available
+     uint32_t& fastTrackAvailMask_l() final { return mFastTrackAvailMask; }
+     uint32_t mFastTrackAvailMask;  // bit i set if fast track [i] is available
                 bool        mHwSupportsPause;
                 bool        mHwPaused;
                 bool        mFlushPending;
@@ -1511,36 +1434,35 @@
 class MixerThread : public PlaybackThread,
                     public StreamOutHalInterfaceLatencyModeCallback  {
 public:
-    MixerThread(const sp<AudioFlinger>& audioFlinger,
+    MixerThread(const sp<IAfThreadCallback>& afThreadCallback,
                 AudioStreamOut* output,
                 audio_io_handle_t id,
                 bool systemReady,
                 type_t type = MIXER,
                 audio_config_base_t *mixerConfig = nullptr);
-    virtual             ~MixerThread();
+    ~MixerThread() override;
 
     // RefBase
-    virtual     void        onFirstRef();
+    void onFirstRef() override;
 
                 // StreamOutHalInterfaceLatencyModeCallback
                 void        onRecommendedLatencyModeChanged(
-                                    std::vector<audio_latency_mode_t> modes) override;
+            std::vector<audio_latency_mode_t> modes) final;
 
     // Thread virtuals
 
-    virtual     bool        checkForNewParameter_l(const String8& keyValuePair,
-                                                   status_t& status);
+    bool checkForNewParameter_l(const String8& keyValuePair, status_t& status) final;
 
-    virtual     bool        isTrackAllowed_l(
+    bool isTrackAllowed_l(
                                     audio_channel_mask_t channelMask, audio_format_t format,
-                                    audio_session_t sessionId, uid_t uid) const override;
+                                    audio_session_t sessionId, uid_t uid) const final;
 protected:
-    virtual     mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
-    virtual     uint32_t    idleSleepTimeUs() const;
-    virtual     uint32_t    suspendSleepTimeUs() const;
-    virtual     void        cacheParameters_l();
+    mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove) override;
+    uint32_t idleSleepTimeUs() const final;
+    uint32_t suspendSleepTimeUs() const final;
+    void cacheParameters_l() override;
 
-    virtual void acquireWakeLock_l() {
+    void acquireWakeLock_l() final {
         PlaybackThread::acquireWakeLock_l();
         if (hasFastMixer()) {
             mFastMixer->setBoottimeOffset(
@@ -1551,15 +1473,15 @@
                 void        dumpInternals_l(int fd, const Vector<String16>& args) override;
 
     // threadLoop snippets
-    virtual     ssize_t     threadLoop_write();
-    virtual     void        threadLoop_standby();
-    virtual     void        threadLoop_mix();
-    virtual     void        threadLoop_sleepTime();
-    virtual     uint32_t    correctLatency_l(uint32_t latency) const;
+    ssize_t threadLoop_write() override;
+    void threadLoop_standby() override;
+    void threadLoop_mix() override;
+    void threadLoop_sleepTime() override;
+    uint32_t correctLatency_l(uint32_t latency) const final;
 
-    virtual     status_t    createAudioPatch_l(const struct audio_patch *patch,
-                                   audio_patch_handle_t *handle);
-    virtual     status_t    releaseAudioPatch_l(const audio_patch_handle_t handle);
+    status_t createAudioPatch_l(
+            const struct audio_patch* patch, audio_patch_handle_t* handle) final;
+    status_t releaseAudioPatch_l(const audio_patch_handle_t handle) final;
 
                 AudioMixer* mAudioMixer;    // normal mixer
 
@@ -1635,17 +1557,21 @@
                 void       setHalLatencyMode_l() override;
 };
 
-class DirectOutputThread : public PlaybackThread {
+class DirectOutputThread : public PlaybackThread, public virtual IAfDirectOutputThread {
 public:
 
-    DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+    sp<IAfDirectOutputThread> asIAfDirectOutputThread() final {
+        return sp<IAfDirectOutputThread>::fromExisting(this);
+    }
+
+    DirectOutputThread(const sp<IAfThreadCallback>& afThreadCallback, AudioStreamOut* output,
                        audio_io_handle_t id, bool systemReady,
                        const audio_offload_info_t& offloadInfo)
-        : DirectOutputThread(audioFlinger, output, id, DIRECT, systemReady, offloadInfo) { }
+        : DirectOutputThread(afThreadCallback, output, id, DIRECT, systemReady, offloadInfo) { }
 
     virtual                 ~DirectOutputThread();
 
-                status_t    selectPresentation(int presentationId, int programId);
+    status_t selectPresentation(int presentationId, int programId) final;
 
     // Thread virtuals
 
@@ -1665,7 +1591,7 @@
                 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();
@@ -1678,16 +1604,16 @@
     audioflinger::MonotonicFrameCounter mMonotonicFrameCounter;  // for VolumeShaper
     bool mVolumeShaperActive = false;
 
-    DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+    DirectOutputThread(const sp<IAfThreadCallback>& afThreadCallback, 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;
@@ -1719,7 +1645,7 @@
 class OffloadThread : public DirectOutputThread {
 public:
 
-    OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+    OffloadThread(const sp<IAfThreadCallback>& afThreadCallback, AudioStreamOut* output,
                   audio_io_handle_t id, bool systemReady,
                   const audio_offload_info_t& offloadInfo);
     virtual                 ~OffloadThread() {};
@@ -1727,7 +1653,7 @@
 
 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();
@@ -1745,11 +1671,8 @@
 
 class AsyncCallbackThread : public Thread {
 public:
-
     explicit AsyncCallbackThread(const wp<PlaybackThread>& playbackThread);
 
-    virtual             ~AsyncCallbackThread();
-
     // Thread virtuals
     virtual bool        threadLoop();
 
@@ -1778,16 +1701,21 @@
     bool                       mAsyncError;
 };
 
-class DuplicatingThread : public MixerThread {
+class DuplicatingThread : public MixerThread, public IAfDuplicatingThread {
 public:
-    DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread,
+    DuplicatingThread(const sp<IAfThreadCallback>& afThreadCallback,
+            IAfPlaybackThread* mainThread,
                       audio_io_handle_t id, bool systemReady);
-    virtual                 ~DuplicatingThread();
+    ~DuplicatingThread() override;
+
+    sp<IAfDuplicatingThread> asIAfDuplicatingThread() final {
+        return sp<IAfDuplicatingThread>::fromExisting(this);
+    }
 
     // Thread virtuals
-                void        addOutputTrack(MixerThread* thread);
-                void        removeOutputTrack(MixerThread* thread);
-                uint32_t    waitTimeMs() const { return mWaitTimeMs; }
+    void addOutputTrack(IAfPlaybackThread* thread) final;
+    void removeOutputTrack(IAfPlaybackThread* thread) final;
+    uint32_t waitTimeMs() const final { return mWaitTimeMs; }
 
                 void        sendMetadataToBackend_l(
                         const StreamOutHalInterface::SourceMetadata& metadata) override;
@@ -1814,8 +1742,8 @@
 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(
@@ -1838,23 +1766,22 @@
 
 class SpatializerThread : public MixerThread {
 public:
-    SpatializerThread(const sp<AudioFlinger>& audioFlinger,
+    SpatializerThread(const sp<IAfThreadCallback>& afThreadCallback,
                            AudioStreamOut* output,
                            audio_io_handle_t id,
                            bool systemReady,
                            audio_config_base_t *mixerConfig);
-            ~SpatializerThread() override {}
 
-            bool hasFastMixer() const override { return false; }
+    bool hasFastMixer() const final { return false; }
 
             // RefBase
-            virtual void        onFirstRef();
+    void onFirstRef() final;
 
-            status_t setRequestedLatencyMode(audio_latency_mode_t mode) override;
+    status_t setRequestedLatencyMode(audio_latency_mode_t mode) final;
 
 protected:
-            void checkOutputStageEffects() override;
-            void setHalLatencyMode_l() override;
+    void checkOutputStageEffects() final;
+    void setHalLatencyMode_l() final;
 
 private:
             // Do not request a specific mode by default
@@ -1864,83 +1791,39 @@
 };
 
 // record thread
-class RecordThread : public ThreadBase
+class RecordThread : public IAfRecordThread, public ThreadBase
 {
+    friend class ResamplerBufferProvider;
 public:
+    sp<IAfRecordThread> asIAfRecordThread() final {
+        return sp<IAfRecordThread>::fromExisting(this);
+    }
 
-    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,
+            RecordThread(const sp<IAfThreadCallback>& afThreadCallback,
                     AudioStreamIn *input,
                     audio_io_handle_t id,
                     bool systemReady
                     );
-            virtual     ~RecordThread();
+    ~RecordThread() override;
 
     // no addTrack_l ?
-    void        destroyTrack_l(const sp<RecordTrack>& track);
-    void        removeTrack_l(const sp<RecordTrack>& track);
+    void destroyTrack_l(const sp<IAfRecordTrack>& track) final;
+    void removeTrack_l(const sp<IAfRecordTrack>& track) final;
 
     // Thread virtuals
-    virtual bool        threadLoop();
-    virtual void        preExit();
+    bool threadLoop() final;
+    void preExit() final;
 
     // RefBase
-    virtual void        onFirstRef();
+    void onFirstRef() final;
 
-    virtual status_t    initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; }
+    status_t initCheck() const final { return mInput == nullptr ? NO_INIT : NO_ERROR; }
 
-    virtual sp<MemoryDealer>    readOnlyHeap() const { return mReadOnlyHeap; }
+    sp<MemoryDealer> readOnlyHeap() const final { return mReadOnlyHeap; }
 
-    virtual sp<IMemory> pipeMemory() const { return mPipeMemory; }
+    sp<IMemory> pipeMemory() const final { return mPipeMemory; }
 
-            sp<AudioFlinger::RecordThread::RecordTrack>  createRecordTrack_l(
+    sp<IAfRecordTrack> createRecordTrack_l(
                     const sp<Client>& client,
                     const audio_attributes_t& attr,
                     uint32_t *pSampleRate,
@@ -1955,17 +1838,19 @@
                     pid_t tid,
                     status_t *status /*non-NULL*/,
                     audio_port_handle_t portId,
-                    int32_t maxSharedAudioHistoryMs);
+                    int32_t maxSharedAudioHistoryMs) final;
 
-            status_t    start(RecordTrack* recordTrack,
+            status_t start(IAfRecordTrack* recordTrack,
                               AudioSystem::sync_event_t event,
-                              audio_session_t triggerSession);
+                              audio_session_t triggerSession) final;
 
             // 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) final;
+    AudioStreamIn* getInput() const final { return mInput; }
+    AudioStreamIn* clearInput() final;
 
-            AudioStreamIn* clearInput();
+            // TODO(b/291317898) Unify with IAfThreadBase
             virtual sp<StreamHalInterface> stream() const;
 
 
@@ -1973,19 +1858,19 @@
                                                status_t& status);
     virtual void        cacheParameters_l() {}
     virtual String8     getParameters(const String8& keys);
-    virtual void        ioConfigChanged(audio_io_config_event_t event, pid_t pid = 0,
-                                        audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
+    void ioConfigChanged(audio_io_config_event_t event, pid_t pid = 0,
+            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final;
     virtual status_t    createAudioPatch_l(const struct audio_patch *patch,
                                            audio_patch_handle_t *handle);
     virtual status_t    releaseAudioPatch_l(const audio_patch_handle_t handle);
             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) final;
+    void deletePatchTrack(const sp<IAfPatchRecord>& record) final;
 
             void        readInputParameters_l();
-    virtual uint32_t    getInputFramesLost();
+    uint32_t getInputFramesLost() const final;
 
     virtual status_t addEffectChain_l(const sp<IAfEffectChain>& chain);
     virtual size_t removeEffectChain_l(const sp<IAfEffectChain>& chain);
@@ -2004,7 +1889,7 @@
     static void syncStartEventCallback(const wp<audioflinger::SyncEvent>& event);
 
     virtual size_t      frameCount() const { return mFrameCount; }
-            bool        hasFastCapture() const { return mFastCapture != 0; }
+    bool hasFastCapture() const final { return mFastCapture != 0; }
     virtual void        toAudioPortConfig(struct audio_port_config *config);
 
     virtual status_t    checkEffectCompatibility_l(const effect_descriptor_t *desc,
@@ -2015,20 +1900,20 @@
                             mActiveTracks.updatePowerState(this, true /* force */);
                         }
 
-            void        checkBtNrec();
+    void checkBtNrec() final;
 
             // Sets the UID records silence
-            void        setRecordSilenced(audio_port_handle_t portId, bool silenced);
+    void setRecordSilenced(audio_port_handle_t portId, bool silenced) final;
 
-            status_t    getActiveMicrophones(
-                    std::vector<media::MicrophoneInfoFw>* activeMicrophones);
-
-            status_t    setPreferredMicrophoneDirection(audio_microphone_direction_t direction);
-            status_t    setPreferredMicrophoneFieldDimension(float zoom);
+    status_t getActiveMicrophones(
+            std::vector<media::MicrophoneInfoFw>* activeMicrophones) const final;
+    status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction) final;
+    status_t setPreferredMicrophoneFieldDimension(float zoom) final;
 
             MetadataUpdate        updateMetadata_l() override;
 
-            bool        fastTrackAvailable() const { return mFastTrackAvail; }
+    bool fastTrackAvailable() const final { return mFastTrackAvail; }
+    void setFastTrackAvailable(bool available) final { mFastTrackAvail = available; }
 
             bool        isTimestampCorrectionEnabled() const override {
                             // checks popcount for exactly one device.
@@ -2038,15 +1923,15 @@
                                     && inDeviceType() == mTimestampCorrectedDevice;
                         }
 
-            status_t    shareAudioHistory(const std::string& sharedAudioPackageName,
+    status_t shareAudioHistory(const std::string& sharedAudioPackageName,
                                           audio_session_t sharedSessionId = AUDIO_SESSION_NONE,
-                                          int64_t sharedAudioStartMs = -1);
+            int64_t sharedAudioStartMs = -1) final;
             status_t    shareAudioHistory_l(const std::string& sharedAudioPackageName,
                                           audio_session_t sharedSessionId = AUDIO_SESSION_NONE,
                                           int64_t sharedAudioStartMs = -1);
-            void        resetAudioHistory_l();
+    void resetAudioHistory_l() final;
 
-    virtual bool        isStreamInitialized() {
+    bool isStreamInitialized() const final {
                             return !(mInput == nullptr || mInput->stream == nullptr);
                         }
 
@@ -2068,10 +1953,10 @@
 
             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;
 
@@ -2138,90 +2023,85 @@
             audio_session_t                     mSharedAudioSessionId = AUDIO_SESSION_NONE;
 };
 
-class MmapThread : public ThreadBase
+class MmapThread : public ThreadBase, public virtual IAfMmapThread
 {
  public:
-
-#include "MmapTracks.h"
-
-    MmapThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+    MmapThread(const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
                AudioHwDevice *hwDev, const sp<StreamHalInterface>& stream, bool systemReady,
                bool isOut);
-    virtual     ~MmapThread();
 
-    virtual     void        configure(const audio_attributes_t *attr,
+    void configure(const audio_attributes_t* attr,
                                       audio_stream_type_t streamType,
                                       audio_session_t sessionId,
                                       const sp<MmapStreamCallback>& callback,
                                       audio_port_handle_t deviceId,
-                                      audio_port_handle_t portId);
+                                      audio_port_handle_t portId) override;
 
-                void        disconnect();
+    void disconnect() final;
 
-    // MmapStreamInterface
-    status_t createMmapBuffer(int32_t minSizeFrames,
-                                      struct audio_mmap_buffer_info *info);
-    status_t getMmapPosition(struct audio_mmap_position *position);
+    // MmapStreamInterface for adapter.
+    status_t createMmapBuffer(int32_t minSizeFrames, struct audio_mmap_buffer_info* info) final;
+    status_t getMmapPosition(struct audio_mmap_position* position) const override;
     status_t start(const AudioClient& client,
                    const audio_attributes_t *attr,
-                   audio_port_handle_t *handle);
-    status_t stop(audio_port_handle_t handle);
-    status_t standby();
-    virtual status_t getExternalPosition(uint64_t *position, int64_t *timeNaos) = 0;
-    virtual status_t reportData(const void* buffer, size_t frameCount);
+            audio_port_handle_t* handle) final;
+    status_t stop(audio_port_handle_t handle) final;
+    status_t standby() final;
+    status_t getExternalPosition(uint64_t* position, int64_t* timeNanos) const = 0;
+    status_t reportData(const void* buffer, size_t frameCount) override;
 
     // RefBase
-    virtual     void        onFirstRef();
+    void onFirstRef() final;
 
     // Thread virtuals
-    virtual     bool        threadLoop();
+    bool threadLoop() final;
 
-    virtual     void        threadLoop_exit();
-    virtual     void        threadLoop_standby();
-    virtual     bool        shouldStandby_l() { return false; }
-    virtual     status_t    exitStandby_l() REQUIRES(mLock);
+    // Not in ThreadBase
+    virtual void threadLoop_exit() final;
+    virtual void threadLoop_standby() final;
+    virtual bool shouldStandby_l() final { return false; }
+    virtual status_t exitStandby_l() REQUIRES(mLock);
 
-    virtual     status_t    initCheck() const { return (mHalStream == 0) ? NO_INIT : NO_ERROR; }
-    virtual     size_t      frameCount() const { return mFrameCount; }
-    virtual     bool        checkForNewParameter_l(const String8& keyValuePair,
-                                                    status_t& status);
-    virtual     String8     getParameters(const String8& keys);
-    virtual     void        ioConfigChanged(audio_io_config_event_t event, pid_t pid = 0,
-                                            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
+    status_t initCheck() const final { return mHalStream == nullptr ? NO_INIT : NO_ERROR; }
+    size_t frameCount() const final { return mFrameCount; }
+    bool checkForNewParameter_l(const String8& keyValuePair, status_t& status) final;
+    String8 getParameters(const String8& keys) final;
+    void ioConfigChanged(audio_io_config_event_t event, pid_t pid = 0,
+            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final;
                 void        readHalParameters_l();
-    virtual     void        cacheParameters_l() {}
-    virtual     status_t    createAudioPatch_l(const struct audio_patch *patch,
-                                               audio_patch_handle_t *handle);
-    virtual     status_t    releaseAudioPatch_l(const audio_patch_handle_t handle);
-    virtual     void        toAudioPortConfig(struct audio_port_config *config);
+    void cacheParameters_l() final {}
+    status_t createAudioPatch_l(
+            const struct audio_patch* patch, audio_patch_handle_t* handle) final;
+    status_t releaseAudioPatch_l(const audio_patch_handle_t handle) final;
+    void toAudioPortConfig(struct audio_port_config* config) override;
 
-    virtual     sp<StreamHalInterface> stream() const { return mHalStream; }
-    virtual     status_t    addEffectChain_l(const sp<IAfEffectChain>& chain);
-    virtual     size_t      removeEffectChain_l(const sp<IAfEffectChain>& chain);
-    virtual     status_t    checkEffectCompatibility_l(const effect_descriptor_t *desc,
-                                                               audio_session_t sessionId);
+    sp<StreamHalInterface> stream() const final { return mHalStream; }
+    status_t addEffectChain_l(const sp<IAfEffectChain>& chain) final;
+    size_t removeEffectChain_l(const sp<IAfEffectChain>& chain) final;
+    status_t checkEffectCompatibility_l(
+            const effect_descriptor_t *desc, audio_session_t sessionId) final;
 
-                uint32_t    hasAudioSession_l(audio_session_t sessionId) const override {
+    uint32_t hasAudioSession_l(audio_session_t sessionId) const override {
                                 // Note: using mActiveTracks as no mTracks here.
                                 return ThreadBase::hasAudioSession_l(sessionId, mActiveTracks);
                             }
-    virtual     status_t    setSyncEvent(const sp<audioflinger::SyncEvent>& event);
-    virtual     bool        isValidSyncEvent(const sp<audioflinger::SyncEvent>& event) const;
+    status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) final;
+    bool isValidSyncEvent(const sp<audioflinger::SyncEvent>& event) const final;
 
-    virtual     void        checkSilentMode_l() {}
-    virtual     void        processVolume_l() {}
+    virtual void checkSilentMode_l() {} // cannot be const (RecordThread)
+    virtual void processVolume_l() {}
                 void        checkInvalidTracks_l();
 
-    virtual     audio_stream_type_t streamType() { return AUDIO_STREAM_DEFAULT; }
-
-    virtual     void        invalidateTracks(audio_stream_type_t streamType __unused) {}
-    virtual     void        invalidateTracks(std::set<audio_port_handle_t>& portIds __unused) {}
+    // Not in ThreadBase
+    virtual audio_stream_type_t streamType() const { return AUDIO_STREAM_DEFAULT; }
+    virtual void invalidateTracks(audio_stream_type_t /* streamType */) {}
+    void invalidateTracks(std::set<audio_port_handle_t>& /* portIds */) override {}
 
                 // Sets the UID records silence
-    virtual     void        setRecordSilenced(audio_port_handle_t portId __unused,
-                                              bool silenced __unused) {}
+    void setRecordSilenced(
+            audio_port_handle_t /* portId */, bool /* silenced */) override {}
 
-    virtual     bool        isStreamInitialized() { return false; }
+    bool isStreamInitialized() const override { return false; }
 
                 void        setClientSilencedState_l(audio_port_handle_t portId, bool silenced) {
                                 mClientSilencedStates[portId] = silenced;
@@ -2244,8 +2124,8 @@
                             }
 
  protected:
-                void        dumpInternals_l(int fd, const Vector<String16>& args) override;
-                void        dumpTracks_l(int fd, const Vector<String16>& args) override;
+    void dumpInternals_l(int fd, const Vector<String16>& args) override;
+    void dumpTracks_l(int fd, const Vector<String16>& args) final;
 
                 /**
                  * @brief mDeviceId  current device port unique identifier
@@ -2260,7 +2140,7 @@
                 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;
 
@@ -2268,56 +2148,59 @@
      static     constexpr int32_t       kMaxNoCallbackWarnings = 5;
 };
 
-class MmapPlaybackThread : public MmapThread, public VolumeInterface
-{
-
+class MmapPlaybackThread : public MmapThread, public IAfMmapPlaybackThread,
+        public virtual VolumeInterface {
 public:
-    MmapPlaybackThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+    MmapPlaybackThread(const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
                        AudioHwDevice *hwDev, AudioStreamOut *output, bool systemReady);
-    virtual     ~MmapPlaybackThread() {}
 
-    virtual     void        configure(const audio_attributes_t *attr,
+    sp<IAfMmapPlaybackThread> asIAfMmapPlaybackThread() final {
+        return sp<IAfMmapPlaybackThread>::fromExisting(this);
+    }
+
+    void configure(const audio_attributes_t* attr,
                                       audio_stream_type_t streamType,
                                       audio_session_t sessionId,
                                       const sp<MmapStreamCallback>& callback,
                                       audio_port_handle_t deviceId,
-                                      audio_port_handle_t portId);
+                                      audio_port_handle_t portId) final;
 
-                AudioStreamOut* clearOutput();
+    AudioStreamOut* clearOutput() final;
 
                 // VolumeInterface
-    virtual     void        setMasterVolume(float value);
-    virtual     void        setMasterMute(bool muted);
-    virtual     void        setStreamVolume(audio_stream_type_t stream, float value);
-    virtual     void        setStreamMute(audio_stream_type_t stream, bool muted);
-    virtual     float       streamVolume(audio_stream_type_t stream) const;
+    void setMasterVolume(float value) final;
+    void setMasterBalance(float /* value */) final {}  // Needs implementation?
+    void setMasterMute(bool muted) final;
+    void setStreamVolume(audio_stream_type_t stream, float value) final;
+    void setStreamMute(audio_stream_type_t stream, bool muted) final;
+    float streamVolume(audio_stream_type_t stream) const final;
 
                 void        setMasterMute_l(bool muted) { mMasterMute = muted; }
 
-    virtual     void        invalidateTracks(audio_stream_type_t streamType);
-                void        invalidateTracks(std::set<audio_port_handle_t>& portIds) override;
+    void invalidateTracks(audio_stream_type_t streamType) final;
+    void invalidateTracks(std::set<audio_port_handle_t>& portIds) final;
 
-    virtual     audio_stream_type_t streamType() { return mStreamType; }
-    virtual     void        checkSilentMode_l();
-                void        processVolume_l() override;
+    audio_stream_type_t streamType() const final { return mStreamType; }
+    void checkSilentMode_l() final;
+    void processVolume_l() final;
 
-                MetadataUpdate        updateMetadata_l() override;
+    MetadataUpdate updateMetadata_l() final;
 
-    virtual     void        toAudioPortConfig(struct audio_port_config *config);
+    void toAudioPortConfig(struct audio_port_config* config) final;
 
-                status_t    getExternalPosition(uint64_t *position, int64_t *timeNanos) override;
+    status_t getExternalPosition(uint64_t* position, int64_t* timeNanos) const final;
 
-    virtual     bool        isStreamInitialized() {
+    bool isStreamInitialized() const final {
                                 return !(mOutput == nullptr || mOutput->stream == nullptr);
                             }
 
-                status_t    reportData(const void* buffer, size_t frameCount) override;
+    status_t reportData(const void* buffer, size_t frameCount) final;
 
-                void startMelComputation_l(const sp<audio_utils::MelProcessor>& processor) override;
-                void stopMelComputation_l() override;
+    void startMelComputation_l(const sp<audio_utils::MelProcessor>& processor) final;
+    void stopMelComputation_l() final;
 
 protected:
-                void        dumpInternals_l(int fd, const Vector<String16>& args) override;
+    void dumpInternals_l(int fd, const Vector<String16>& args) final;
 
                 audio_stream_type_t         mStreamType;
                 float                       mMasterVolume;
@@ -2329,28 +2212,29 @@
                 mediautils::atomic_sp<audio_utils::MelProcessor> mMelProcessor;
 };
 
-class MmapCaptureThread : public MmapThread
+class MmapCaptureThread : public MmapThread, public IAfMmapCaptureThread
 {
-
 public:
-    MmapCaptureThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+    MmapCaptureThread(const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
                       AudioHwDevice *hwDev, AudioStreamIn *input, bool systemReady);
-    virtual     ~MmapCaptureThread() {}
 
-                AudioStreamIn* clearInput();
+    sp<IAfMmapCaptureThread> asIAfMmapCaptureThread() final {
+        return sp<IAfMmapCaptureThread>::fromExisting(this);
+    }
 
-                status_t       exitStandby_l() REQUIRES(mLock) override;
+    AudioStreamIn* clearInput() final;
 
-                MetadataUpdate           updateMetadata_l() override;
-                void           processVolume_l() override;
-                void           setRecordSilenced(audio_port_handle_t portId,
-                                                 bool silenced) override;
+    status_t exitStandby_l() REQUIRES(mLock) final;
 
-    virtual     void           toAudioPortConfig(struct audio_port_config *config);
+    MetadataUpdate updateMetadata_l() final;
+    void processVolume_l() final;
+    void setRecordSilenced(audio_port_handle_t portId, bool silenced) final;
 
-                status_t       getExternalPosition(uint64_t *position, int64_t *timeNanos) override;
+    void toAudioPortConfig(struct audio_port_config* config) final;
 
-    virtual     bool           isStreamInitialized() {
+    status_t getExternalPosition(uint64_t* position, int64_t* timeNanos) const final;
+
+    bool isStreamInitialized() const final {
                                    return !(mInput == nullptr || mInput->stream == nullptr);
                                }
 
@@ -2361,12 +2245,12 @@
 
 class BitPerfectThread : public MixerThread {
 public:
-    BitPerfectThread(const sp<AudioFlinger>& audioflinger, AudioStreamOut *output,
+    BitPerfectThread(const sp<IAfThreadCallback>& afThreadCallback, AudioStreamOut *output,
                      audio_io_handle_t id, bool systemReady);
 
 protected:
-    mixer_state prepareTracks_l(Vector<sp<Track>> *tracksToRemove) override;
-    void threadLoop_mix() override;
+    mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove) final;
+    void threadLoop_mix() final;
 
 private:
     bool mIsBitPerfect;
@@ -2374,4 +2258,4 @@
     float mVolumeRight = 0.f;
 };
 
-private:
+} // namespace android
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index d5b6a98..5708c61 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -15,46 +15,26 @@
 ** limitations under the License.
 */
 
-#ifndef INCLUDING_FROM_AUDIOFLINGER_H
-    #error This header file should only be included from AudioFlinger.h
-#endif
+#pragma once
+
+#include "Configuration.h"  // TEE_SINK
+#include "IAfTrack.h"
+
+#include <afutils/NBAIO_Tee.h>
+#include <android-base/macros.h>  // DISALLOW_COPY_AND_ASSIGN
+#include <datapath/TrackMetrics.h>
+#include <mediautils/BatteryNotifier.h>
+
+#include <atomic>    // avoid transitive dependency
+#include <list>      // avoid transitive dependency
+#include <optional>  // avoid transitive dependency
+
+namespace android {
 
 // base for record and playback
-class TrackBase : public ExtendedAudioBufferProvider, public RefBase {
-
+class TrackBase : public ExtendedAudioBufferProvider, public virtual IAfTrackBase {
 public:
-    enum track_state : int32_t {
-        IDLE,
-        FLUSHED,        // for PlaybackTracks only
-        STOPPED,
-        // next 2 states are currently used for fast tracks
-        // and offloaded tracks only
-        STOPPING_1,     // waiting for first underrun
-        STOPPING_2,     // waiting for presentation complete
-        RESUMING,       // for PlaybackTracks only
-        ACTIVE,
-        PAUSING,
-        PAUSED,
-        STARTING_1,     // for RecordTrack only
-        STARTING_2,     // for RecordTrack only
-    };
-
-    // where to allocate the data buffer
-    enum alloc_type {
-        ALLOC_CBLK,     // allocate immediately after control block
-        ALLOC_READONLY, // allocate from a separate read-only heap per thread
-        ALLOC_PIPE,     // do not allocate; use the pipe buffer
-        ALLOC_LOCAL,    // allocate a local buffer
-        ALLOC_NONE,     // do not allocate:use the buffer passed to TrackBase constructor
-    };
-
-    enum track_type {
-        TYPE_DEFAULT,
-        TYPE_OUTPUT,
-        TYPE_PATCH,
-    };
-
-                        TrackBase(ThreadBase *thread,
+    TrackBase(IAfThreadBase* thread,
                                 const sp<Client>& client,
                                 const audio_attributes_t& mAttr,
                                 uint32_t sampleRate,
@@ -71,87 +51,79 @@
                                 track_type type = TYPE_DEFAULT,
                                 audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE,
                                 std::string metricsId = {});
-    virtual             ~TrackBase();
-    virtual status_t    initCheck() const;
+    ~TrackBase() override;
+    status_t initCheck() const override;
+    sp<IMemory> getCblk() const final { return mCblkMemory; }
+    audio_track_cblk_t* cblk() const final { return mCblk; }
+    audio_session_t sessionId() const final { return mSessionId; }
+    uid_t uid() const final { return mUid; }
+    pid_t creatorPid() const final { return mCreatorPid; }
+    audio_port_handle_t portId() const final { return mPortId; }
+    status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) override;
+    track_state state() const final { return mState; }
+    void setState(track_state state) final { mState = state; }
+    sp<IMemory> getBuffers() const final { return mBufferMemory; }
+    void* buffer() const final { return mBuffer; }
+    size_t bufferSize() const final { return mBufferSize; }
 
-    virtual status_t    start(AudioSystem::sync_event_t event,
-                             audio_session_t triggerSession) = 0;
-    virtual void        stop() = 0;
-            sp<IMemory> getCblk() const { return mCblkMemory; }
-            audio_track_cblk_t* cblk() const { return mCblk; }
-            audio_session_t sessionId() const { return mSessionId; }
-            uid_t       uid() const { return mUid; }
-            pid_t       creatorPid() const { return mCreatorPid; }
-
-            audio_port_handle_t portId() const { return mPortId; }
-    virtual status_t    setSyncEvent(const sp<audioflinger::SyncEvent>& event);
-
-            sp<IMemory> getBuffers() const { return mBufferMemory; }
-            void*       buffer() const { return mBuffer; }
-            size_t      bufferSize() const { return mBufferSize; }
-    virtual bool        isFastTrack() const = 0;
-    virtual bool        isDirect() const = 0;
-            bool        isOutputTrack() const { return (mType == TYPE_OUTPUT); }
-            bool        isPatchTrack() const { return (mType == TYPE_PATCH); }
-            bool        isExternalTrack() const { return !isOutputTrack() && !isPatchTrack(); }
-
-    virtual void        invalidate() {
+    bool isOutputTrack() const final { return (mType == TYPE_OUTPUT); }
+    bool isPatchTrack() const final { return (mType == TYPE_PATCH); }
+    bool isExternalTrack() const final { return !isOutputTrack() && !isPatchTrack(); }
+    void invalidate() override {
                             if (mIsInvalid) return;
                             mTrackMetrics.logInvalidate();
                             mIsInvalid = true;
                         }
-            bool        isInvalid() const { return mIsInvalid; }
+    bool isInvalid() const final { return mIsInvalid; }
+    void terminate() final { mTerminated = true; }
+    bool isTerminated() const final { return mTerminated; }
+    audio_attributes_t attributes() const final { return mAttr; }
+    bool isSpatialized() const override { return false; }
+    bool isBitPerfect() const override { return false; }
 
-            void        terminate() { mTerminated = true; }
-            bool        isTerminated() const { return mTerminated; }
+    wp<IAfThreadBase> thread() const final { return mThread; }
 
-    audio_attributes_t  attributes() const { return mAttr; }
-
-    virtual bool        isSpatialized() const { return false; }
-
-    virtual bool        isBitPerfect() const { return false; }
+    const sp<ServerProxy>& serverProxy() const final { return mServerProxy; }
 
 #ifdef TEE_SINK
-           void         dumpTee(int fd, const std::string &reason) const {
-                                mTee.dump(fd, reason);
-                        }
+    void dumpTee(int fd, const std::string &reason) const final {
+        mTee.dump(fd, reason);
+    }
 #endif
-
-            /** returns the buffer contents size converted to time in milliseconds
-             * for PCM Playback or Record streaming tracks. The return value is zero for
-             * PCM static tracks and not defined for non-PCM tracks.
-             *
-             * This may be called without the thread lock.
-             */
-    virtual double      bufferLatencyMs() const {
+    /** returns the buffer contents size converted to time in milliseconds
+     * for PCM Playback or Record streaming tracks. The return value is zero for
+     * PCM static tracks and not defined for non-PCM tracks.
+     *
+     * This may be called without the thread lock.
+     */
+    double bufferLatencyMs() const override {
                             return mServerProxy->framesReadySafe() * 1000. / sampleRate();
                         }
 
-            /** returns whether the track supports server latency computation.
-             * This is set in the constructor and constant throughout the track lifetime.
-             */
+    /** returns whether the track supports server latency computation.
+     * This is set in the constructor and constant throughout the track lifetime.
+     */
+    bool isServerLatencySupported() const final { return mServerLatencySupported; }
 
-            bool        isServerLatencySupported() const { return mServerLatencySupported; }
-
-            /** computes the server latency for PCM Playback or Record track
-             * to the device sink/source.  This is the time for the next frame in the track buffer
-             * written or read from the server thread to the device source or sink.
-             *
-             * This may be called without the thread lock, but latencyMs and fromTrack
-             * may be not be synchronized. For example PatchPanel may not obtain the
-             * thread lock before calling.
-             *
-             * \param latencyMs on success is set to the latency in milliseconds of the
-             *        next frame written/read by the server thread to/from the track buffer
-             *        from the device source/sink.
-             * \param fromTrack on success is set to true if latency was computed directly
-             *        from the track timestamp; otherwise set to false if latency was
-             *        estimated from the server timestamp.
-             *        fromTrack may be nullptr or omitted if not required.
-             *
-             * \returns OK or INVALID_OPERATION on failure.
-             */
-            status_t    getServerLatencyMs(double *latencyMs, bool *fromTrack = nullptr) const {
+    /** computes the server latency for PCM Playback or Record track
+     * to the device sink/source.  This is the time for the next frame in the track buffer
+     * written or read from the server thread to the device source or sink.
+     *
+     * This may be called without the thread lock, but latencyMs and fromTrack
+     * may be not be synchronized. For example PatchPanel may not obtain the
+     * thread lock before calling.
+     *
+     * \param latencyMs on success is set to the latency in milliseconds of the
+     *        next frame written/read by the server thread to/from the track buffer
+     *        from the device source/sink.
+     * \param fromTrack on success is set to true if latency was computed directly
+     *        from the track timestamp; otherwise set to false if latency was
+     *        estimated from the server timestamp.
+     *        fromTrack may be nullptr or omitted if not required.
+     *
+     * \returns OK or INVALID_OPERATION on failure.
+     */
+    status_t getServerLatencyMs(double* latencyMs, bool* fromTrack = nullptr) const final {
                             if (!isServerLatencySupported()) {
                                 return INVALID_OPERATION;
                             }
@@ -170,25 +142,25 @@
                             return OK;
                         }
 
-            /** computes the total client latency for PCM Playback or Record tracks
-             * for the next client app access to the device sink/source; i.e. the
-             * server latency plus the buffer latency.
-             *
-             * This may be called without the thread lock, but latencyMs and fromTrack
-             * may be not be synchronized. For example PatchPanel may not obtain the
-             * thread lock before calling.
-             *
-             * \param latencyMs on success is set to the latency in milliseconds of the
-             *        next frame written/read by the client app to/from the track buffer
-             *        from the device sink/source.
-             * \param fromTrack on success is set to true if latency was computed directly
-             *        from the track timestamp; otherwise set to false if latency was
-             *        estimated from the server timestamp.
-             *        fromTrack may be nullptr or omitted if not required.
-             *
-             * \returns OK or INVALID_OPERATION on failure.
-             */
-            status_t    getTrackLatencyMs(double *latencyMs, bool *fromTrack = nullptr) const {
+    /** computes the total client latency for PCM Playback or Record tracks
+     * for the next client app access to the device sink/source; i.e. the
+     * server latency plus the buffer latency.
+     *
+     * This may be called without the thread lock, but latencyMs and fromTrack
+     * may be not be synchronized. For example PatchPanel may not obtain the
+     * thread lock before calling.
+     *
+     * \param latencyMs on success is set to the latency in milliseconds of the
+     *        next frame written/read by the client app to/from the track buffer
+     *        from the device sink/source.
+     * \param fromTrack on success is set to true if latency was computed directly
+     *        from the track timestamp; otherwise set to false if latency was
+     *        estimated from the server timestamp.
+     *        fromTrack may be nullptr or omitted if not required.
+     *
+     * \returns OK or INVALID_OPERATION on failure.
+     */
+    status_t getTrackLatencyMs(double* latencyMs, bool* fromTrack = nullptr) const {
                             double serverLatencyMs;
                             status_t status = getServerLatencyMs(&serverLatencyMs, fromTrack);
                             if (status == OK) {
@@ -197,21 +169,15 @@
                             return status;
                         }
 
-           // TODO: Consider making this external.
-           struct FrameTime {
-               int64_t frames;
-               int64_t timeNs;
-           };
-
-           // KernelFrameTime is updated per "mix" period even for non-pcm tracks.
-           void         getKernelFrameTime(FrameTime *ft) const {
+    // KernelFrameTime is updated per "mix" period even for non-pcm tracks.
+    void getKernelFrameTime(FrameTime* ft) const final {
                            *ft = mKernelFrameTime.load();
                         }
 
-           audio_format_t format() const { return mFormat; }
-           int id() const { return mId; }
+    audio_format_t format() const final { return mFormat; }
+    int id() const final { return mId; }
 
-    const char *getTrackStateAsString() const {
+    const char* getTrackStateAsString() const final {
         if (isTerminated()) {
             return "TERMINATED";
         }
@@ -245,19 +211,19 @@
 
     // Called by the PlaybackThread to indicate that the track is becoming active
     // and a new interval should start with a given device list.
-    void logBeginInterval(const std::string& devices) {
+    void logBeginInterval(const std::string& devices) final {
         mTrackMetrics.logBeginInterval(devices);
     }
 
     // Called by the PlaybackThread to indicate the track is no longer active.
-    void logEndInterval() {
+    void logEndInterval() final {
         mTrackMetrics.logEndInterval();
     }
 
     // Called to tally underrun frames in playback.
-    virtual void tallyUnderrunFrames(size_t /* frames */) {}
+    void tallyUnderrunFrames(size_t /* frames */) override {}
 
-    audio_channel_mask_t channelMask() const { return mChannelMask; }
+    audio_channel_mask_t channelMask() const final { return mChannelMask; }
 
     /** @return true if the track has changed (metadata or volume) since
      *          the last time this function was called,
@@ -265,10 +231,26 @@
      *          false otherwise.
      *  Thread safe.
      */
-    bool readAndClearHasChanged() { return !mChangeNotified.test_and_set(); }
+    bool readAndClearHasChanged() final { return !mChangeNotified.test_and_set(); }
 
     /** Set that a metadata has changed and needs to be notified to backend. Thread safe. */
-    void setMetadataHasChanged() { mChangeNotified.clear(); }
+    void setMetadataHasChanged() final { mChangeNotified.clear(); }
+
+    /**
+     * Called when a track moves to active state to record its contribution to battery usage.
+     * Track state transitions should eventually be handled within the track class.
+     */
+    void beginBatteryAttribution() final {
+        mBatteryStatsHolder.emplace(uid());
+    }
+
+    /**
+     * Called when a track moves out of the active state to record its contribution
+     * to battery usage.
+     */
+    void endBatteryAttribution() final {
+        mBatteryStatsHolder.reset();
+    }
 
 protected:
     DISALLOW_COPY_AND_ASSIGN(TrackBase);
@@ -285,31 +267,31 @@
     }
 
     // AudioBufferProvider interface
-    virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0;
-    virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
+    // status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) override;
+    void releaseBuffer(AudioBufferProvider::Buffer* buffer) override;
 
     // ExtendedAudioBufferProvider interface is only needed for Track,
     // but putting it in TrackBase avoids the complexity of virtual inheritance
-    virtual size_t  framesReady() const { return SIZE_MAX; }
+    size_t framesReady() const override { return SIZE_MAX; } // MmapTrack doesn't implement.
 
     uint32_t channelCount() const { return mChannelCount; }
 
-    size_t frameSize() const { return mFrameSize; }
+    size_t frameSize() const final { return mFrameSize; }
 
-    virtual uint32_t sampleRate() const { return mSampleRate; }
+    uint32_t sampleRate() const override { return mSampleRate; }
 
-    bool isStopped() const {
+    bool isStopped() const final {
         return (mState == STOPPED || mState == FLUSHED);
     }
 
     // for fast tracks and offloaded tracks only
-    bool isStopping() const {
+    bool isStopping() const final {
         return mState == STOPPING_1 || mState == STOPPING_2;
     }
-    bool isStopping_1() const {
+    bool isStopping_1() const final {
         return mState == STOPPING_1;
     }
-    bool isStopping_2() const {
+    bool isStopping_2() const final {
         return mState == STOPPING_2;
     }
 
@@ -351,7 +333,7 @@
                                     // true for Track, false for RecordTrack,
                                     // this could be a track type if needed later
 
-    const wp<ThreadBase> mThread;
+    const wp<IAfThreadBase> mThread;
     const alloc_type     mAllocType;
     /*const*/ sp<Client> mClient;   // see explanation at ~TrackBase() why not const
     sp<IMemory>         mCblkMemory;
@@ -413,39 +395,32 @@
 
     // If the last track change was notified to the client with readAndClearHasChanged
     std::atomic_flag    mChangeNotified = ATOMIC_FLAG_INIT;
+    // RAII object for battery stats book-keeping
+    std::optional<mediautils::BatteryStatsAudioHandle> mBatteryStatsHolder;
 };
 
-// PatchProxyBufferProvider interface is implemented by PatchTrack and PatchRecord.
-// it provides buffer access methods that map those of a ClientProxy (see AudioTrackShared.h)
-class PatchProxyBufferProvider
+class PatchTrackBase : public PatchProxyBufferProvider, public virtual IAfPatchTrackBase
 {
 public:
-
-    virtual ~PatchProxyBufferProvider() {}
-
-    virtual bool        producesBufferOnDemand() const = 0;
-    virtual status_t    obtainBuffer(Proxy::Buffer* buffer,
-                                     const struct timespec *requested = NULL) = 0;
-    virtual void        releaseBuffer(Proxy::Buffer* buffer) = 0;
-};
-
-class PatchTrackBase : public PatchProxyBufferProvider
-{
-public:
-    using Timeout = std::optional<std::chrono::nanoseconds>;
-                        PatchTrackBase(const sp<ClientProxy>& proxy, const ThreadBase& thread,
+                        PatchTrackBase(const sp<ClientProxy>& proxy,
+                                       IAfThreadBase* thread,
                                        const Timeout& timeout);
-            void        setPeerTimeout(std::chrono::nanoseconds timeout);
-            template <typename T>
-            void        setPeerProxy(const sp<T> &proxy, bool holdReference) {
-                            mPeerReferenceHold = holdReference ? proxy : nullptr;
-                            mPeerProxy = proxy.get();
-                        }
-            void        clearPeerProxy() {
+            void setPeerTimeout(std::chrono::nanoseconds timeout) final;
+            void setPeerProxy(const sp<IAfPatchTrackBase>& proxy, bool holdReference) final {
+                if (proxy) {
+                    mPeerReferenceHold = holdReference ? proxy : nullptr;
+                    mPeerProxy = proxy->asPatchProxyBufferProvider();
+                } else {
+                    clearPeerProxy();
+                }
+            }
+            void clearPeerProxy() final {
                             mPeerReferenceHold.clear();
                             mPeerProxy = nullptr;
                         }
 
+            PatchProxyBufferProvider* asPatchProxyBufferProvider() final { return this; }
+
             bool        producesBufferOnDemand() const override { return false; }
 
 protected:
@@ -453,5 +428,6 @@
     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 9ec3ee3..ecea9eb 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -15,28 +15,33 @@
 ** limitations under the License.
 */
 
-
 #define LOG_TAG "AudioFlinger"
 //#define LOG_NDEBUG 0
 #define ATRACE_TAG ATRACE_TAG_AUDIO
 
-#include "Configuration.h"
-#include <linux/futex.h>
-#include <math.h>
-#include <sys/syscall.h>
+#include "MmapTracks.h"
+#include "PlaybackTracks.h"
+#include "RecordTracks.h"
+
+#include "Client.h"
+#include "IAfEffect.h"
+#include "IAfThread.h"
+#include "ResamplerBufferProvider.h"
+
+#include <audio_utils/minifloat.h>
+#include <media/AudioValidator.h>
+#include <media/RecordBufferConverter.h>
+#include <media/nbaio/Pipe.h>
+#include <media/nbaio/PipeReader.h>
+#include <mediautils/ServiceUtilities.h>
+#include <mediautils/SharedMemoryAllocator.h>
+#include <private/media/AudioTrackShared.h>
 #include <utils/Log.h>
 #include <utils/Trace.h>
 
-#include <private/media/AudioTrackShared.h>
-
-#include "AudioFlinger.h"
-
-#include <media/nbaio/Pipe.h>
-#include <media/nbaio/PipeReader.h>
-#include <media/AudioValidator.h>
-#include <media/RecordBufferConverter.h>
-#include <mediautils/ServiceUtilities.h>
-#include <audio_utils/minifloat.h>
+#include <linux/futex.h>
+#include <math.h>
+#include <sys/syscall.h>
 
 // ----------------------------------------------------------------------------
 
@@ -76,8 +81,8 @@
 static volatile int32_t nextTrackId = 55;
 
 // TrackBase constructor must be called with AudioFlinger::mLock held
-AudioFlinger::ThreadBase::TrackBase::TrackBase(
-            ThreadBase *thread,
+TrackBase::TrackBase(
+        IAfThreadBase *thread,
             const sp<Client>& client,
             const audio_attributes_t& attr,
             uint32_t sampleRate,
@@ -94,7 +99,7 @@
             track_type type,
             audio_port_handle_t portId,
             std::string metricsId)
-    :   RefBase(),
+    :
         mThread(thread),
         mAllocType(alloc),
         mClient(client),
@@ -253,7 +258,7 @@
    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 +269,7 @@
     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();
@@ -272,7 +277,7 @@
     mCblkMemory.clear();    // free the shared memory before releasing the heap it belongs to
     if (mClient != 0) {
         // Client destructor must run with AudioFlinger client mutex locked
-        Mutex::Autolock _l(mClient->audioFlinger()->mClientLock);
+        Mutex::Autolock _l(mClient->afClientCallback()->clientMutex());
         // If the client's reference count drops to zero, the associated destructor
         // must run with AudioFlinger lock held. Thus the explicit clear() rather than
         // relying on the automatic clear() at end of scope.
@@ -289,7 +294,7 @@
 // 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);
@@ -303,29 +308,28 @@
     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,
+        IAfThreadBase* thread, const Timeout& timeout)
     : mProxy(proxy)
 {
     if (timeout) {
         setPeerTimeout(*timeout);
     } else {
         // Double buffer mixer
-        uint64_t mixBufferNs = ((uint64_t)2 * thread.frameCount() * 1000000000) /
-                                              thread.sampleRate();
+        uint64_t mixBufferNs = ((uint64_t)2 * thread->frameCount() * 1000000000) /
+                                              thread->sampleRate();
         setPeerTimeout(std::chrono::nanoseconds{mixBufferNs});
     }
 }
 
-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;
 }
@@ -339,7 +343,7 @@
 
 class TrackHandle : public android::media::BnAudioTrack {
 public:
-    explicit TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track);
+    explicit TrackHandle(const sp<IAfTrack>& track);
     ~TrackHandle() override;
 
     binder::Status getCblk(std::optional<media::SharedFileRegion>* _aidl_return) final;
@@ -373,20 +377,18 @@
             const media::audio::common::AudioPlaybackRate& playbackRate) final;
 
 private:
-    const sp<AudioFlinger::PlaybackThread::Track> mTrack;
+    const sp<IAfTrack> mTrack;
 };
 
 /* static */
-sp<media::IAudioTrack> AudioFlinger::PlaybackThread::Track::createIAudioTrackAdapter(
-        const sp<Track>& track) {
+sp<media::IAudioTrack> IAfTrack::createIAudioTrackAdapter(const sp<IAfTrack>& track) {
     return sp<TrackHandle>::make(track);
 }
 
-TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
+TrackHandle::TrackHandle(const sp<IAfTrack>& track)
     : BnAudioTrack(),
       mTrack(track)
 {
-    // TODO(b/288339104) binder thread priority change not needed.
     setMinSchedulerPolicy(SCHED_NORMAL, ANDROID_PRIORITY_AUDIO);
 }
 
@@ -556,9 +558,8 @@
 // -------------------------------
 
 // static
-sp<AudioFlinger::PlaybackThread::OpPlayAudioMonitor>
-AudioFlinger::PlaybackThread::OpPlayAudioMonitor::createIfNeeded(
-            AudioFlinger::ThreadBase* thread,
+sp<OpPlayAudioMonitor> OpPlayAudioMonitor::createIfNeeded(
+            IAfThreadBase* thread,
             const AttributionSourceState& attributionSource, const audio_attributes_t& attr, int id,
             audio_stream_type_t streamType)
 {
@@ -588,11 +589,10 @@
     return sp<OpPlayAudioMonitor>::make(thread, attributionSource, attr.usage, id, uid);
 }
 
-AudioFlinger::PlaybackThread::OpPlayAudioMonitor::OpPlayAudioMonitor(
-        AudioFlinger::ThreadBase* thread,
-        const AttributionSourceState& attributionSource,
-        audio_usage_t usage, int id, uid_t uid)
-    : mThread(wp<AudioFlinger::ThreadBase>::fromExisting(thread)),
+OpPlayAudioMonitor::OpPlayAudioMonitor(IAfThreadBase* thread,
+                                       const AttributionSourceState& attributionSource,
+                                       audio_usage_t usage, int id, uid_t uid)
+    : mThread(wp<IAfThreadBase>::fromExisting(thread)),
       mHasOpPlayAudio(true),
       mAttributionSource(attributionSource),
       mUsage((int32_t)usage),
@@ -601,7 +601,7 @@
       mPackageName(VALUE_OR_FATAL(aidl2legacy_string_view_String16(
                   attributionSource.packageName.value_or("")))) {}
 
-AudioFlinger::PlaybackThread::OpPlayAudioMonitor::~OpPlayAudioMonitor()
+OpPlayAudioMonitor::~OpPlayAudioMonitor()
 {
     if (mOpCallback != 0) {
         mAppOpsManager.stopWatchingMode(mOpCallback);
@@ -609,7 +609,7 @@
     mOpCallback.clear();
 }
 
-void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::onFirstRef()
+void OpPlayAudioMonitor::onFirstRef()
 {
     // make sure not to broadcast the initial state since it is not needed and could
     // cause a deadlock since this method can be called with the mThread->mLock held
@@ -621,14 +621,14 @@
     }
 }
 
-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(bool doBroadcast)
+void OpPlayAudioMonitor::checkPlayAudioForUsage(bool doBroadcast)
 {
     const bool hasAppOps = mAttributionSource.packageName.has_value()
         && mAppOpsManager.checkAudioOpNoThrow(
@@ -640,20 +640,20 @@
         ALOGD("OpPlayAudio: track:%d usage:%d %smuted", mId, mUsage, hasAppOps ? "not " : "");
         if (doBroadcast) {
             auto thread = mThread.promote();
-            if (thread != nullptr && thread->type() == AudioFlinger::ThreadBase::OFFLOAD) {
+            if (thread != nullptr && thread->type() == IAfThreadBase::OFFLOAD) {
                 // Wake up Thread if offloaded, otherwise it may be several seconds for update.
-                Mutex::Autolock _l(thread->mLock);
+                Mutex::Autolock _l(thread->mutex());
                 thread->broadcast_l();
             }
         }
     }
 }
 
-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);
@@ -667,7 +667,7 @@
 }
 
 // static
-void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::getPackagesForUid(
+void OpPlayAudioMonitor::getPackagesForUid(
     uid_t uid, Vector<String16>& packages)
 {
     PermissionController permissionController;
@@ -678,9 +678,57 @@
 #undef LOG_TAG
 #define LOG_TAG "AF::Track"
 
+/* static */
+sp<IAfTrack> IAfTrack::create(
+        IAfPlaybackThread* 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(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(
+        IAfPlaybackThread* thread,
             const sp<Client>& client,
             audio_stream_type_t streamType,
             const audio_attributes_t& attr,
@@ -714,7 +762,7 @@
                   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),
@@ -771,15 +819,15 @@
         // race with setSyncEvent(). However, if we call it, we cannot properly start
         // static fast tracks (SoundPool) immediately after stopping.
         //mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads();
-        ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
-        int i = __builtin_ctz(thread->mFastTrackAvailMask);
+        ALOG_ASSERT(thread->fastTrackAvailMask_l() != 0);
+        const int i = __builtin_ctz(thread->fastTrackAvailMask_l());
         ALOG_ASSERT(0 < i && i < (int)FastMixerState::sMaxFastTracks);
         // FIXME This is too eager.  We allocate a fast track index before the
         //       fast track becomes active.  Since fast tracks are a scarce resource,
         //       this means we are potentially denying other more important fast tracks from
         //       being created.  It would be better to allocate the index dynamically.
         mFastIndex = i;
-        thread->mFastTrackAvailMask &= ~(1 << i);
+        thread->fastTrackAvailMask_l() &= ~(1 << i);
     }
 
     mServerLatencySupported = checkServerLatencySupported(format, flags);
@@ -804,7 +852,7 @@
     mTrackMetrics.logConstructor(creatorPid, uid, id(), traits, streamType);
 }
 
-AudioFlinger::PlaybackThread::Track::~Track()
+Track::~Track()
 {
     ALOGV("%s(%d)", __func__, mId);
 
@@ -817,7 +865,7 @@
     }
 }
 
-status_t AudioFlinger::PlaybackThread::Track::initCheck() const
+status_t Track::initCheck() const
 {
     status_t status = TrackBase::initCheck();
     if (status == NO_ERROR && mCblk == nullptr) {
@@ -826,7 +874,7 @@
     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
@@ -839,20 +887,20 @@
     sp<Track> keep(this);
     { // scope for mLock
         bool wasActive = false;
-        sp<ThreadBase> thread = mThread.promote();
+        const sp<IAfThreadBase> thread = mThread.promote();
         if (thread != 0) {
-            Mutex::Autolock _l(thread->mLock);
-            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+            Mutex::Autolock _l(thread->mutex());
+            auto* const playbackThread = thread->asIAfPlaybackThread().get();
             wasActive = playbackThread->destroyTrack_l(this);
+            forEachTeePatchTrack_l([](const auto& patchTrack) { patchTrack->destroy(); });
         }
         if (isExternalTrack() && !wasActive) {
             AudioSystem::releaseOutput(mPortId);
         }
     }
-    forEachTeePatchTrack([](auto patchTrack) { patchTrack->destroy(); });
 }
 
-void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
+void Track::appendDumpHeader(String8& result) const
 {
     result.appendFormat("Type     Id Active Client Session Port Id S  Flags "
                         "  Format Chn mask  SRate "
@@ -863,7 +911,7 @@
                         isServerLatencySupported() ? "   Latency" : "");
 }
 
-void AudioFlinger::PlaybackThread::Track::appendDump(String8& result, bool active)
+void Track::appendDump(String8& result, bool active) const
 {
     char trackType;
     switch (mType) {
@@ -905,7 +953,7 @@
     }
 
     char fillingStatus;
-    switch (mFillingUpStatus) {
+    switch (mFillingStatus) {
     case FS_INVALID:
         fillingStatus = 'I';
         break;
@@ -991,12 +1039,12 @@
     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;
@@ -1014,14 +1062,14 @@
     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;
@@ -1031,12 +1079,12 @@
         // 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);
@@ -1052,7 +1100,7 @@
 // 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.
@@ -1061,12 +1109,12 @@
     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.
@@ -1083,14 +1131,14 @@
 }
 
 // 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;
     }
@@ -1104,33 +1152,33 @@
     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();
+    const sp<IAfThreadBase> thread = mThread.promote();
     if (thread != 0) {
         if (isOffloaded()) {
-            Mutex::Autolock _laf(thread->mAudioFlinger->mLock);
-            Mutex::Autolock _lth(thread->mLock);
+            Mutex::Autolock _laf(thread->afThreadCallback()->mutex());
+            Mutex::Autolock _lth(thread->mutex());
             sp<IAfEffectChain> ec = thread->getEffectChain_l(mSessionId);
-            if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() ||
+            if (thread->afThreadCallback()->isNonOffloadableGlobalEffectEnabled_l() ||
                     (ec != 0 && ec->isNonOffloadableEnabled())) {
                 invalidate();
                 return PERMISSION_DENIED;
             }
         }
-        Mutex::Autolock _lth(thread->mLock);
+        Mutex::Autolock _lth(thread->mutex());
         track_state state = mState;
         // here the track could be either new, or restarted
         // in both cases "unstop" the track
@@ -1162,7 +1210,7 @@
                     __func__, mId, (int)mThreadIoHandle);
         }
 
-        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+        auto* const playbackThread = thread->asIAfPlaybackThread().get();
 
         // states to reset position info for pcm tracks
         if (audio_is_linear_pcm(mFormat)
@@ -1222,14 +1270,16 @@
             buffer.mFrameCount = 1;
             (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/);
         }
+        if (status == NO_ERROR) {
+            forEachTeePatchTrack_l([](const auto& patchTrack) { patchTrack->start(); });
+        }
     } else {
         status = BAD_VALUE;
     }
     if (status == NO_ERROR) {
-        forEachTeePatchTrack([](auto patchTrack) { patchTrack->start(); });
-
         // send format to AudioManager for playback activity monitoring
-        sp<IAudioManager> audioManager = thread->mAudioFlinger->getOrCreateAudioManager();
+        const sp<IAudioManager> audioManager =
+                thread->afThreadCallback()->getOrCreateAudioManager();
         if (audioManager && mPortId != AUDIO_PORT_HANDLE_NONE) {
             std::unique_ptr<os::PersistableBundle> bundle =
                     std::make_unique<os::PersistableBundle>();
@@ -1248,17 +1298,17 @@
     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();
+    const sp<IAfThreadBase> thread = mThread.promote();
     if (thread != 0) {
-        Mutex::Autolock _l(thread->mLock);
+        Mutex::Autolock _l(thread->mutex());
         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();
-            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
+            auto* const playbackThread = thread->asIAfPlaybackThread().get();
+            if (!playbackThread->isTrackActive(this)) {
                 reset();
                 mState = STOPPED;
             } else if (!isFastTrack() && !isOffloaded() && !isDirect()) {
@@ -1270,24 +1320,24 @@
                 // move to STOPPING_2 when drain completes and then STOPPED
                 mState = STOPPING_1;
                 if (isOffloaded()) {
-                    mRetryCount = PlaybackThread::kMaxTrackStopRetriesOffload;
+                    mRetryCount = IAfPlaybackThread::kMaxTrackStopRetriesOffload;
                 }
             }
             playbackThread->broadcast_l();
             ALOGV("%s(%d): not stopping/stopped => stopping/stopped on thread %d",
                     __func__, mId, (int)mThreadIoHandle);
         }
+        forEachTeePatchTrack_l([](const auto& patchTrack) { patchTrack->stop(); });
     }
-    forEachTeePatchTrack([](auto patchTrack) { patchTrack->stop(); });
 }
 
-void AudioFlinger::PlaybackThread::Track::pause()
+void Track::pause()
 {
     ALOGV("%s(%d): calling pid %d", __func__, mId, IPCThreadState::self()->getCallingPid());
-    sp<ThreadBase> thread = mThread.promote();
+    const sp<IAfThreadBase> thread = mThread.promote();
     if (thread != 0) {
-        Mutex::Autolock _l(thread->mLock);
-        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+        Mutex::Autolock _l(thread->mutex());
+        auto* const playbackThread = thread->asIAfPlaybackThread().get();
         switch (mState) {
         case STOPPING_1:
         case STOPPING_2:
@@ -1313,23 +1363,23 @@
         default:
             break;
         }
+        // Pausing the TeePatch to avoid a glitch on underrun, at the cost of buffered audio loss.
+        forEachTeePatchTrack_l([](const auto& patchTrack) { patchTrack->pause(); });
     }
-    // Pausing the TeePatch to avoid a glitch on underrun, at the cost of buffered audio loss.
-    forEachTeePatchTrack([](auto patchTrack) { patchTrack->pause(); });
 }
 
-void AudioFlinger::PlaybackThread::Track::flush()
+void Track::flush()
 {
     ALOGV("%s(%d)", __func__, mId);
-    sp<ThreadBase> thread = mThread.promote();
+    const sp<IAfThreadBase> thread = mThread.promote();
     if (thread != 0) {
-        Mutex::Autolock _l(thread->mLock);
-        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+        Mutex::Autolock _l(thread->mutex());
+        auto* const playbackThread = thread->asIAfPlaybackThread().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.
         // Requires FastTrack removal be BLOCK_UNTIL_ACKED
-        if (playbackThread->mActiveTracks.indexOf(this) < 0) {
+        if (!playbackThread->isTrackActive(this)) {
             (void)mServerProxy->flushBufferIfNeeded();
         }
 
@@ -1368,7 +1418,7 @@
             if (isDirect()) {
                 mFlushHwPending = true;
             }
-            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
+            if (!playbackThread->isTrackActive(this)) {
                 reset();
             }
         }
@@ -1376,13 +1426,14 @@
         // before mixer thread can run. This is important when offloading
         // because the hardware buffer could hold a large amount of audio
         playbackThread->broadcast_l();
+        // Flush the Tee to avoid on resume playing old data and glitching on the transition to
+        // new data
+        forEachTeePatchTrack_l([](const auto& patchTrack) { patchTrack->flush(); });
     }
-    // Flush the Tee to avoid on resume playing old data and glitching on the transition to new data
-    forEachTeePatchTrack([](auto patchTrack) { patchTrack->flush(); });
 }
 
 // must be called with thread lock held
-void AudioFlinger::PlaybackThread::Track::flushAck()
+void Track::flushAck()
 {
     if (!isOffloaded() && !isDirect()) {
         return;
@@ -1395,12 +1446,12 @@
     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.
@@ -1408,7 +1459,7 @@
         // 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;
@@ -1416,34 +1467,35 @@
     }
 }
 
-status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs)
+status_t Track::setParameters(const String8& keyValuePairs)
 {
-    sp<ThreadBase> thread = mThread.promote();
+    const sp<IAfThreadBase> 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() == IAfThreadBase::DIRECT
+            || thread->type() == IAfThreadBase::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();
+    const sp<IAfThreadBase> 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() == IAfThreadBase::DIRECT
+            || thread->type() == IAfThreadBase::OFFLOAD) {
+        auto directOutputThread = thread->asIAfDirectOutputThread().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)
 {
@@ -1451,16 +1503,16 @@
 
     if (isOffloadedOrDirect()) {
         // Signal thread to fetch new volume.
-        sp<ThreadBase> thread = mThread.promote();
+        const sp<IAfThreadBase> thread = mThread.promote();
         if (thread != 0) {
-            Mutex::Autolock _l(thread->mLock);
+            Mutex::Autolock _l(thread->mutex());
             thread->broadcast_l();
         }
     }
     return status;
 }
 
-sp<VolumeShaper::State> AudioFlinger::PlaybackThread::Track::getVolumeShaperState(int id)
+sp<VolumeShaper::State> Track::getVolumeShaperState(int id) const
 {
     // Note: We don't check if Thread exists.
 
@@ -1468,7 +1520,7 @@
     return mVolumeHandler->getVolumeShaperState(id);
 }
 
-void AudioFlinger::PlaybackThread::Track::setFinalVolume(float volumeLeft, float volumeRight)
+void Track::setFinalVolume(float volumeLeft, float volumeRight)
 {
     mFinalVolumeLeft = volumeLeft;
     mFinalVolumeRight = volumeRight;
@@ -1484,7 +1536,7 @@
     }
 }
 
-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) {
@@ -1556,26 +1608,26 @@
     *backInserter++ = metadata;
 }
 
-void AudioFlinger::PlaybackThread::Track::updateTeePatches() {
+void Track::updateTeePatches_l() {
     if (mTeePatchesToUpdate.has_value()) {
-        forEachTeePatchTrack([](auto patchTrack) { patchTrack->destroy(); });
+        forEachTeePatchTrack_l([](const auto& patchTrack) { patchTrack->destroy(); });
         mTeePatches = mTeePatchesToUpdate.value();
         if (mState == TrackBase::ACTIVE || mState == TrackBase::RESUMING ||
                 mState == TrackBase::STOPPING_1) {
-            forEachTeePatchTrack([](auto patchTrack) { patchTrack->start(); });
+            forEachTeePatchTrack_l([](const auto& patchTrack) { patchTrack->start(); });
         }
         mTeePatchesToUpdate.reset();
     }
 }
 
-void AudioFlinger::PlaybackThread::Track::setTeePatchesToUpdate(TeePatches teePatchesToUpdate) {
+void Track::setTeePatchesToUpdate_l(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) {
@@ -1607,31 +1659,32 @@
     }
 }
 
-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();
+    const sp<IAfThreadBase> thread = mThread.promote();
     if (thread == 0) {
         return INVALID_OPERATION;
     }
 
-    Mutex::Autolock _l(thread->mLock);
-    PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+    Mutex::Autolock _l(thread->mutex());
+    auto* const playbackThread = thread->asIAfPlaybackThread().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();
+    const sp<IAfThreadBase> 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()
-    sp<AudioFlinger> af = mClient->audioFlinger();
+    auto dstThread = thread->asIAfPlaybackThread();
+    // srcThread is initialized by call to moveAuxEffectToIo()
+    sp<IAfPlaybackThread> srcThread;
+    const auto& af = mClient->afClientCallback();
     status_t status = af->moveAuxEffectToIo(EffectId, dstThread, &srcThread);
 
     if (EffectId != 0 && status == NO_ERROR) {
@@ -1647,14 +1700,14 @@
     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.
@@ -1697,7 +1750,7 @@
 }
 
 // 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.
 
@@ -1729,14 +1782,14 @@
     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) {
@@ -1751,7 +1804,7 @@
 
 // implement VolumeBufferProvider interface
 
-gain_minifloat_packed_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
+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));
@@ -1776,7 +1829,7 @@
     return vlr;
 }
 
-status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(
+status_t Track::setSyncEvent(
         const sp<audioflinger::SyncEvent>& event)
 {
     if (isTerminated() || mState == PAUSED ||
@@ -1792,19 +1845,19 @@
     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;
@@ -1814,25 +1867,25 @@
     (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX);
 }
 
-void AudioFlinger::PlaybackThread::Track::signal()
+void Track::signal()
 {
-    sp<ThreadBase> thread = mThread.promote();
+    const sp<IAfThreadBase> thread = mThread.promote();
     if (thread != 0) {
-        PlaybackThread *t = (PlaybackThread *)thread.get();
-        Mutex::Autolock _l(t->mLock);
+        auto* const t = thread->asIAfPlaybackThread().get();
+        Mutex::Autolock _l(t->mutex());
         t->broadcast_l();
     }
 }
 
-status_t AudioFlinger::PlaybackThread::Track::getDualMonoMode(audio_dual_mono_mode_t* mode)
+status_t Track::getDualMonoMode(audio_dual_mono_mode_t* mode) const
 {
     status_t status = INVALID_OPERATION;
     if (isOffloadedOrDirect()) {
-        sp<ThreadBase> thread = mThread.promote();
+        const sp<IAfThreadBase> thread = mThread.promote();
         if (thread != nullptr) {
-            PlaybackThread *t = (PlaybackThread *)thread.get();
-            Mutex::Autolock _l(t->mLock);
-            status = t->mOutput->stream->getDualMonoMode(mode);
+            auto* const t = thread->asIAfPlaybackThread().get();
+            Mutex::Autolock _l(t->mutex());
+            status = t->getOutput_l()->stream->getDualMonoMode(mode);
             ALOGD_IF((status == NO_ERROR) && (mDualMonoMode != *mode),
                     "%s: mode %d inconsistent", __func__, mDualMonoMode);
         }
@@ -1840,15 +1893,15 @@
     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();
+        const sp<IAfThreadBase> thread = mThread.promote();
         if (thread != nullptr) {
-            auto t = static_cast<PlaybackThread *>(thread.get());
-            Mutex::Autolock lock(t->mLock);
-            status = t->mOutput->stream->setDualMonoMode(mode);
+            auto* const t = thread->asIAfPlaybackThread().get();
+            Mutex::Autolock lock(t->mutex());
+            status = t->getOutput_l()->stream->setDualMonoMode(mode);
             if (status == NO_ERROR) {
                 mDualMonoMode = mode;
             }
@@ -1857,15 +1910,15 @@
     return status;
 }
 
-status_t AudioFlinger::PlaybackThread::Track::getAudioDescriptionMixLevel(float* leveldB)
+status_t Track::getAudioDescriptionMixLevel(float* leveldB) const
 {
     status_t status = INVALID_OPERATION;
     if (isOffloadedOrDirect()) {
-        sp<ThreadBase> thread = mThread.promote();
+        sp<IAfThreadBase> thread = mThread.promote();
         if (thread != nullptr) {
-            auto t = static_cast<PlaybackThread *>(thread.get());
-            Mutex::Autolock lock(t->mLock);
-            status = t->mOutput->stream->getAudioDescriptionMixLevel(leveldB);
+            auto* const t = thread->asIAfPlaybackThread().get();
+            Mutex::Autolock lock(t->mutex());
+            status = t->getOutput_l()->stream->getAudioDescriptionMixLevel(leveldB);
             ALOGD_IF((status == NO_ERROR) && (mAudioDescriptionMixLevel != *leveldB),
                     "%s: level %.3f inconsistent", __func__, mAudioDescriptionMixLevel);
         }
@@ -1873,15 +1926,15 @@
     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();
+        const sp<IAfThreadBase> thread = mThread.promote();
         if (thread != nullptr) {
-            auto t = static_cast<PlaybackThread *>(thread.get());
-            Mutex::Autolock lock(t->mLock);
-            status = t->mOutput->stream->setAudioDescriptionMixLevel(leveldB);
+            auto* const t = thread->asIAfPlaybackThread().get();
+            Mutex::Autolock lock(t->mutex());
+            status = t->getOutput_l()->stream->setAudioDescriptionMixLevel(leveldB);
             if (status == NO_ERROR) {
                 mAudioDescriptionMixLevel = leveldB;
             }
@@ -1890,16 +1943,16 @@
     return status;
 }
 
-status_t AudioFlinger::PlaybackThread::Track::getPlaybackRateParameters(
-        audio_playback_rate_t* playbackRate)
+status_t Track::getPlaybackRateParameters(
+        audio_playback_rate_t* playbackRate) const
 {
     status_t status = INVALID_OPERATION;
     if (isOffloadedOrDirect()) {
-        sp<ThreadBase> thread = mThread.promote();
+        const sp<IAfThreadBase> thread = mThread.promote();
         if (thread != nullptr) {
-            auto t = static_cast<PlaybackThread *>(thread.get());
-            Mutex::Autolock lock(t->mLock);
-            status = t->mOutput->stream->getPlaybackRateParameters(playbackRate);
+            auto* const t = thread->asIAfPlaybackThread().get();
+            Mutex::Autolock lock(t->mutex());
+            status = t->getOutput_l()->stream->getPlaybackRateParameters(playbackRate);
             ALOGD_IF((status == NO_ERROR) &&
                     !isAudioPlaybackRateEqual(mPlaybackRateParameters, *playbackRate),
                     "%s: playbackRate inconsistent", __func__);
@@ -1908,16 +1961,16 @@
     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();
+        const sp<IAfThreadBase> thread = mThread.promote();
         if (thread != nullptr) {
-            auto t = static_cast<PlaybackThread *>(thread.get());
-            Mutex::Autolock lock(t->mLock);
-            status = t->mOutput->stream->setPlaybackRateParameters(playbackRate);
+            auto* const t = thread->asIAfPlaybackThread().get();
+            Mutex::Autolock lock(t->mutex());
+            status = t->getOutput_l()->stream->setPlaybackRateParameters(playbackRate);
             if (status == NO_ERROR) {
                 mPlaybackRateParameters = playbackRate;
             }
@@ -1927,7 +1980,7 @@
 }
 
 //To be called with thread lock held
-bool AudioFlinger::PlaybackThread::Track::isResumePending() {
+bool Track::isResumePending() const {
     if (mState == RESUMING) {
         return true;
     }
@@ -1941,7 +1994,7 @@
 }
 
 //To be called with thread lock held
-void AudioFlinger::PlaybackThread::Track::resumeAck() {
+void Track::resumeAck() {
     if (mState == RESUMING) {
         mState = ACTIVE;
     }
@@ -1955,7 +2008,7 @@
 }
 
 //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.
@@ -2035,14 +2088,14 @@
     }
 }
 
-bool AudioFlinger::PlaybackThread::Track::AudioVibrationController::setMute(bool muted) {
-    sp<ThreadBase> thread = mTrack->mThread.promote();
+bool Track::AudioVibrationController::setMute(bool muted) {
+    const sp<IAfThreadBase> thread = mTrack->mThread.promote();
     if (thread != 0) {
         // Lock for updating mHapticPlaybackEnabled.
-        Mutex::Autolock _l(thread->mLock);
-        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+        Mutex::Autolock _l(thread->mutex());
+        auto* const playbackThread = thread->asIAfPlaybackThread().get();
         if ((mTrack->channelMask() & AUDIO_CHANNEL_HAPTIC_ALL) != AUDIO_CHANNEL_NONE
-                && playbackThread->mHapticChannelCount > 0) {
+                && playbackThread->hapticChannelCount() > 0) {
             ALOGD("%s, haptic playback was %s for track %d",
                     __func__, muted ? "muted" : "unmuted", mTrack->id());
             mTrack->setHapticPlaybackEnabled(!muted);
@@ -2052,13 +2105,13 @@
     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();
@@ -2068,9 +2121,28 @@
 #undef LOG_TAG
 #define LOG_TAG "AF::OutputTrack"
 
-AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
-            PlaybackThread *playbackThread,
-            DuplicatingThread *sourceThread,
+/* static */
+sp<IAfOutputTrack> IAfOutputTrack::create(
+        IAfPlaybackThread* playbackThread,
+        IAfDuplicatingThread* sourceThread,
+        uint32_t sampleRate,
+        audio_format_t format,
+        audio_channel_mask_t channelMask,
+        size_t frameCount,
+        const AttributionSourceState& attributionSource) {
+    return sp<OutputTrack>::make(
+            playbackThread,
+            sourceThread,
+            sampleRate,
+            format,
+            channelMask,
+            frameCount,
+            attributionSource);
+}
+
+OutputTrack::OutputTrack(
+            IAfPlaybackThread* playbackThread,
+            IAfDuplicatingThread* sourceThread,
             uint32_t sampleRate,
             audio_format_t format,
             audio_channel_mask_t channelMask,
@@ -2087,7 +2159,7 @@
 
     if (mCblk != NULL) {
         mOutBuffer.frameCount = 0;
-        playbackThread->mTracks.add(this);
+        playbackThread->addOutputTrack_l(this);
         ALOGV("%s(): mCblk %p, mBuffer %p, "
                 "frameCount %zu, mChannelMask 0x%08x",
                 __func__, mCblk, mBuffer,
@@ -2105,13 +2177,13 @@
     }
 }
 
-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);
@@ -2124,7 +2196,7 @@
     return status;
 }
 
-void AudioFlinger::PlaybackThread::OutputTrack::stop()
+void OutputTrack::stop()
 {
     Track::stop();
     clearBufferQueue();
@@ -2132,11 +2204,11 @@
     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();
-        if (thread != nullptr && thread->standby()) {
+        const sp<IAfThreadBase> thread = mThread.promote();
+        if (thread != nullptr && thread->inStandby()) {
             // preload one silent buffer to trigger mixer on start()
             ClientProxy::Buffer buf { .mFrameCount = mClientProxy->getStartThresholdInFrames() };
             status_t status = mClientProxy->obtainBuffer(&buf);
@@ -2154,7 +2226,7 @@
             // 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 = thread->asIAfPlaybackThread().get();
             if (!pt->waitForHalStart()) {
                 ALOGW("%s(%d): timeout waiting for thread to exit standby", __func__, mId);
                 stop();
@@ -2243,8 +2315,8 @@
 
     // If we could not write all frames, allocate a buffer and queue it for next time.
     if (inBuffer.frameCount) {
-        sp<ThreadBase> thread = mThread.promote();
-        if (thread != 0 && !thread->standby()) {
+        const sp<IAfThreadBase> thread = mThread.promote();
+        if (thread != nullptr && !thread->inStandby()) {
             queueBuffer(inBuffer);
         }
     }
@@ -2258,7 +2330,7 @@
     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;
@@ -2281,13 +2353,13 @@
     }
 }
 
-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;
@@ -2296,7 +2368,7 @@
     setMetadataHasChanged();
 }
 
-status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
+status_t OutputTrack::obtainBuffer(
         AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
 {
     ClientProxy::Buffer buf;
@@ -2310,7 +2382,7 @@
     return status;
 }
 
-void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
+void OutputTrack::clearBufferQueue()
 {
     size_t size = mBufferQueue.size();
 
@@ -2322,7 +2394,7 @@
     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)) {
@@ -2334,7 +2406,38 @@
 #undef LOG_TAG
 #define LOG_TAG "AF::PatchTrack"
 
-AudioFlinger::PlaybackThread::PatchTrack::PatchTrack(PlaybackThread *playbackThread,
+/* static */
+sp<IAfPatchTrack> IAfPatchTrack::create(
+        IAfPlaybackThread* playbackThread,
+        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(
+            playbackThread,
+            streamType,
+            sampleRate,
+            channelMask,
+            format,
+            frameCount,
+            buffer,
+            bufferSize,
+            flags,
+            timeout,
+            frameCountToBeReady);
+}
+
+PatchTrack::PatchTrack(IAfPlaybackThread* playbackThread,
                                                      audio_stream_type_t streamType,
                                                      uint32_t sampleRate,
                                                      audio_channel_mask_t channelMask,
@@ -2353,7 +2456,7 @@
               TYPE_PATCH, AUDIO_PORT_HANDLE_NONE, frameCountToBeReady),
         PatchTrackBase(mCblk ? new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true)
                         : nullptr,
-                       *playbackThread, timeout)
+                       playbackThread, timeout)
 {
     ALOGV("%s(%d): sampleRate %d mPeerTimeout %d.%03d sec",
                                       __func__, mId, sampleRate,
@@ -2361,12 +2464,12 @@
                                       (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();
@@ -2375,7 +2478,7 @@
     }
 }
 
-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);
@@ -2387,7 +2490,7 @@
 }
 
 // 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);
@@ -2413,7 +2516,7 @@
     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;
@@ -2423,7 +2526,7 @@
     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;
@@ -2440,7 +2543,7 @@
     return status;
 }
 
-void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(Proxy::Buffer* buffer)
+void PatchTrack::releaseBuffer(Proxy::Buffer* buffer)
 {
     mProxy->releaseBuffer(buffer);
     restartIfDisabled();
@@ -2449,23 +2552,23 @@
     // 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 (const sp<IAfThreadBase> thread = mThread.promote();
             thread != 0) {
-            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+            auto* const playbackThread = thread->asIAfPlaybackThread().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);
@@ -2483,7 +2586,7 @@
 
 class RecordHandle : public android::media::BnAudioRecord {
 public:
-    explicit RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack);
+    explicit RecordHandle(const sp<IAfRecordTrack>& recordTrack);
     ~RecordHandle() override;
     binder::Status start(int /*AudioSystem::sync_event_t*/ event,
             int /*audio_session_t*/ triggerSession) final;
@@ -2497,24 +2600,23 @@
             const std::string& sharedAudioPackageName, int64_t sharedAudioStartMs) final;
 
 private:
-    const sp<AudioFlinger::RecordThread::RecordTrack> mRecordTrack;
+    const sp<IAfRecordTrack> mRecordTrack;
 
     // for use from destructor
     void stop_nonvirtual();
 };
 
 /* static */
-sp<media::IAudioRecord> AudioFlinger::RecordThread::RecordTrack::createIAudioRecordAdapter(
-        const sp<RecordTrack>& recordTrack) {
+sp<media::IAudioRecord> IAfRecordTrack::createIAudioRecordAdapter(
+        const sp<IAfRecordTrack>& recordTrack) {
     return sp<RecordHandle>::make(recordTrack);
 }
 
 RecordHandle::RecordHandle(
-        const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
+        const sp<IAfRecordTrack>& recordTrack)
     : BnAudioRecord(),
     mRecordTrack(recordTrack)
 {
-    // TODO(b/288339104) binder thread priority change not needed.
     setMinSchedulerPolicy(SCHED_NORMAL, ANDROID_PRIORITY_AUDIO);
 }
 
@@ -2568,9 +2670,47 @@
 #undef LOG_TAG
 #define LOG_TAG "AF::RecordTrack"
 
+
+/* static */
+sp<IAfRecordTrack> IAfRecordTrack::create(IAfRecordThread* 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(
+        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(
+            IAfRecordThread* thread,
             const sp<Client>& client,
             const audio_attributes_t& attr,
             uint32_t sampleRate,
@@ -2609,7 +2749,7 @@
 
     if (!isDirect()) {
         mRecordBufferConverter = new RecordBufferConverter(
-                thread->mChannelMask, thread->mFormat, thread->mSampleRate,
+                thread->channelMask(), thread->format(), thread->sampleRate(),
                 channelMask, format, sampleRate);
         // Check if the RecordBufferConverter construction was successful.
         // If not, don't continue with construction.
@@ -2629,8 +2769,8 @@
     mResamplerBufferProvider = new ResamplerBufferProvider(this);
 
     if (flags & AUDIO_INPUT_FLAG_FAST) {
-        ALOG_ASSERT(thread->mFastTrackAvail);
-        thread->mFastTrackAvail = false;
+        ALOG_ASSERT(thread->fastTrackAvailable());
+        thread->setFastTrackAvailable(false);
     } else {
         // TODO: only Normal Record has timestamps (Fast Record does not).
         mServerLatencySupported = checkServerLatencySupported(mFormat, flags);
@@ -2645,14 +2785,14 @@
     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) {
@@ -2662,7 +2802,7 @@
 }
 
 // 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;
@@ -2676,12 +2816,12 @@
     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();
+    const sp<IAfThreadBase> thread = mThread.promote();
     if (thread != 0) {
-        RecordThread *recordThread = (RecordThread *)thread.get();
+        auto* const recordThread = thread->asIAfRecordThread().get();
         return recordThread->start(this, event, triggerSession);
     } else {
         ALOGW("%s track %d: thread was destroyed", __func__, portId());
@@ -2689,27 +2829,27 @@
     }
 }
 
-void AudioFlinger::RecordThread::RecordTrack::stop()
+void RecordTrack::stop()
 {
-    sp<ThreadBase> thread = mThread.promote();
+    const sp<IAfThreadBase> thread = mThread.promote();
     if (thread != 0) {
-        RecordThread *recordThread = (RecordThread *)thread.get();
+        auto* const recordThread = thread->asIAfRecordThread().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();
+        const sp<IAfThreadBase> thread = mThread.promote();
         if (thread != 0) {
-            Mutex::Autolock _l(thread->mLock);
-            RecordThread *recordThread = (RecordThread *) thread.get();
+            Mutex::Autolock _l(thread->mutex());
+            auto* const recordThread = thread->asIAfRecordThread().get();
             priorState = mState;
             if (!mSharedAudioPackageName.empty()) {
                 recordThread->resetAudioHistory_l();
@@ -2740,7 +2880,7 @@
     }
 }
 
-void AudioFlinger::RecordThread::RecordTrack::invalidate()
+void RecordTrack::invalidate()
 {
     TrackBase::invalidate();
     // FIXME should use proxy, and needs work
@@ -2752,7 +2892,7 @@
 }
 
 
-void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
+void RecordTrack::appendDumpHeader(String8& result) const
 {
     result.appendFormat("Active     Id Client Session Port Id  S  Flags  "
                         " Format Chn mask  SRate Source  "
@@ -2760,7 +2900,7 @@
                         isServerLatencySupported() ? "   Latency" : "");
 }
 
-void AudioFlinger::RecordThread::RecordTrack::appendDump(String8& result, bool active)
+void RecordTrack::appendDump(String8& result, bool active) const
 {
     result.appendFormat("%c%5s %6d %6u %7u %7u  %2s 0x%03X "
             "%08X %08X %6u %6X "
@@ -2799,26 +2939,26 @@
 }
 
 // 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();
+    const sp<IAfThreadBase> threadBase = mThread.promote();
     if (threadBase != 0) {
         // TODO: use actual buffer filling status instead of 2 buffers when info is available
         // from audio HAL
-        framesToDrop = threadBase->mFrameCount * 2;
+        framesToDrop = threadBase->frameCount() * 2;
     }
 
     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)
 {
@@ -2858,40 +2998,40 @@
     mServerLatencyMs.store(latencyMs);
 }
 
-status_t AudioFlinger::RecordThread::RecordTrack::getActiveMicrophones(
-        std::vector<media::MicrophoneInfoFw>* activeMicrophones)
+status_t RecordTrack::getActiveMicrophones(
+        std::vector<media::MicrophoneInfoFw>* activeMicrophones) const
 {
-    sp<ThreadBase> thread = mThread.promote();
+    const sp<IAfThreadBase> thread = mThread.promote();
     if (thread != 0) {
-        RecordThread *recordThread = (RecordThread *)thread.get();
+        auto* const recordThread = thread->asIAfRecordThread().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();
+    const sp<IAfThreadBase> thread = mThread.promote();
     if (thread != 0) {
-        RecordThread *recordThread = (RecordThread *)thread.get();
+        auto* const recordThread = thread->asIAfRecordThread().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) {
+    const sp<IAfThreadBase> thread = mThread.promote();
     if (thread != 0) {
-        RecordThread *recordThread = (RecordThread *)thread.get();
+        auto* const recordThread = thread->asIAfRecordThread().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();
@@ -2908,9 +3048,9 @@
         return PERMISSION_DENIED;
     }
 
-    sp<ThreadBase> thread = mThread.promote();
+    const sp<IAfThreadBase> thread = mThread.promote();
     if (thread != 0) {
-        RecordThread *recordThread = (RecordThread *)thread.get();
+        auto* const recordThread = thread->asIAfRecordThread().get();
         status_t status = recordThread->shareAudioHistory(
                 sharedAudioPackageName, mSessionId, sharedAudioStartMs);
         if (status == NO_ERROR) {
@@ -2922,7 +3062,7 @@
     }
 }
 
-void AudioFlinger::RecordThread::RecordTrack::copyMetadataTo(MetadataInserter& backInserter) const
+void RecordTrack::copyMetadataTo(MetadataInserter& backInserter) const
 {
 
     // Do not forward PatchRecord metadata with unspecified audio source
@@ -2946,7 +3086,33 @@
 #undef LOG_TAG
 #define LOG_TAG "AF::PatchRecord"
 
-AudioFlinger::RecordThread::PatchRecord::PatchRecord(RecordThread *recordThread,
+/* static */
+sp<IAfPatchRecord> IAfPatchRecord::create(
+        IAfRecordThread* recordThread,
+        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(
+            recordThread,
+            sampleRate,
+            channelMask,
+            format,
+            frameCount,
+            buffer,
+            bufferSize,
+            flags,
+            timeout,
+            source);
+}
+
+PatchRecord::PatchRecord(IAfRecordThread* recordThread,
                                                      uint32_t sampleRate,
                                                      audio_channel_mask_t channelMask,
                                                      audio_format_t format,
@@ -2963,7 +3129,7 @@
                 audioServerAttributionSource(getpid()), flags, TYPE_PATCH),
         PatchTrackBase(mCblk ? new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, false, true)
                         : nullptr,
-                       *recordThread, timeout)
+                       recordThread, timeout)
 {
     ALOGV("%s(%d): sampleRate %d mPeerTimeout %d.%03d sec",
                                       __func__, mId, sampleRate,
@@ -2971,7 +3137,7 @@
                                       (int)(mPeerTimeout.tv_nsec / 1000000));
 }
 
-AudioFlinger::RecordThread::PatchRecord::~PatchRecord()
+PatchRecord::~PatchRecord()
 {
     ALOGV("%s(%d)", __func__, mId);
 }
@@ -2995,7 +3161,7 @@
 }
 
 // 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);
@@ -3010,7 +3176,7 @@
 }
 
 // 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);
@@ -3032,7 +3198,7 @@
     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;
@@ -3042,13 +3208,13 @@
     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);
 }
@@ -3063,8 +3229,28 @@
     return {ptr, free};
 }
 
-AudioFlinger::RecordThread::PassthruPatchRecord::PassthruPatchRecord(
-        RecordThread *recordThread,
+/* static */
+sp<IAfPatchRecord> IAfPatchRecord::createPassThru(
+        IAfRecordThread* recordThread,
+        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(
+            recordThread,
+            sampleRate,
+            channelMask,
+            format,
+            frameCount,
+            flags,
+            source);
+}
+
+PassthruPatchRecord::PassthruPatchRecord(
+        IAfRecordThread* recordThread,
         uint32_t sampleRate,
         audio_channel_mask_t channelMask,
         audio_format_t format,
@@ -3080,18 +3266,18 @@
     memset(mStubBuffer.get(), 0, mFrameCount * mFrameSize);
 }
 
-sp<StreamInHalInterface> AudioFlinger::RecordThread::PassthruPatchRecord::obtainStream(
-        sp<ThreadBase>* thread)
+sp<StreamInHalInterface> PassthruPatchRecord::obtainStream(
+        sp<IAfThreadBase>* thread)
 {
     *thread = mThread.promote();
     if (!*thread) return nullptr;
-    RecordThread *recordThread = static_cast<RecordThread*>((*thread).get());
-    Mutex::Autolock _l(recordThread->mLock);
-    return recordThread->mInput ? recordThread->mInput->stream : nullptr;
+    auto* const recordThread = (*thread)->asIAfRecordThread().get();
+    Mutex::Autolock _l(recordThread->mutex());
+    return recordThread->getInput() ? recordThread->getInput()->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) {
@@ -3109,7 +3295,7 @@
     const size_t framesToRead = std::min(buffer->mFrameCount, mFrameCount);
     buffer->mFrameCount = 0;
     buffer->mRaw = nullptr;
-    sp<ThreadBase> thread;
+    sp<IAfThreadBase> thread;
     sp<StreamInHalInterface> stream = obtainStream(&thread);
     if (!stream) return NO_INIT;  // If there is no stream, RecordThread is not reading.
 
@@ -3157,7 +3343,7 @@
     return result;
 }
 
-void AudioFlinger::RecordThread::PassthruPatchRecord::releaseBuffer(Proxy::Buffer* buffer)
+void PassthruPatchRecord::releaseBuffer(Proxy::Buffer* buffer)
 {
     if (buffer->mFrameCount <= mUnconsumedFrames) {
         mUnconsumedFrames -= buffer->mFrameCount;
@@ -3174,7 +3360,7 @@
 // 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);
@@ -3193,15 +3379,15 @@
     return 0;
 }
 
-status_t AudioFlinger::RecordThread::PassthruPatchRecord::getCapturePosition(
+status_t PassthruPatchRecord::getCapturePosition(
         int64_t* frames, int64_t* time)
 {
-    sp<ThreadBase> thread;
+    sp<IAfThreadBase> 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'.
@@ -3213,7 +3399,7 @@
 }
 
 // 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;
@@ -3221,7 +3407,7 @@
     return NO_ERROR;
 }
 
-void AudioFlinger::RecordThread::PassthruPatchRecord::releaseBuffer(
+void PassthruPatchRecord::releaseBuffer(
         AudioBufferProvider::Buffer* buffer)
 {
     buffer->frameCount = 0;
@@ -3232,7 +3418,32 @@
 #undef LOG_TAG
 #define LOG_TAG "AF::MmapTrack"
 
-AudioFlinger::MmapThread::MmapTrack::MmapTrack(ThreadBase *thread,
+/* static */
+sp<IAfMmapTrack> IAfMmapTrack::create(IAfThreadBase* 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(
+            thread,
+            attr,
+            sampleRate,
+            format,
+            channelMask,
+            sessionId,
+            isOut,
+            attributionSource,
+            creatorPid,
+            portId);
+}
+
+MmapTrack::MmapTrack(IAfThreadBase* thread,
         const audio_attributes_t& attr,
         uint32_t sampleRate,
         audio_format_t format,
@@ -3258,27 +3469,27 @@
     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;
@@ -3286,21 +3497,20 @@
 }
 
 // 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
@@ -3331,13 +3541,13 @@
     }
 }
 
-void AudioFlinger::MmapThread::MmapTrack::appendDumpHeader(String8& result)
+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)
+void MmapTrack::appendDump(String8& result, bool active __unused) const
 {
     result.appendFormat("%6u %7u %7u %08X %08X %6u 0x%03X ",
             mPid,
diff --git a/services/audioflinger/afutils/Android.bp b/services/audioflinger/afutils/Android.bp
index 1580b8f..b5131eb 100644
--- a/services/audioflinger/afutils/Android.bp
+++ b/services/audioflinger/afutils/Android.bp
@@ -39,13 +39,17 @@
         "AudioWatchdog.cpp",
         "BufLog.cpp",
         "NBAIO_Tee.cpp",
+        "Permission.cpp",
         "PropertyUtils.cpp",
         "TypedLogger.cpp",
     ],
 
     shared_libs: [
+        "framework-permission-aidl-cpp",
+        "libaudioclient_aidl_conversion",
         "libaudioutils",
         "libbase",
+        "libbinder",
         "libcutils", // property_get_int32
         "liblog",
         "libnbaio",
diff --git a/services/audioflinger/afutils/DumpTryLock.h b/services/audioflinger/afutils/DumpTryLock.h
new file mode 100644
index 0000000..c185a68
--- /dev/null
+++ b/services/audioflinger/afutils/DumpTryLock.h
@@ -0,0 +1,31 @@
+/*
+ *
+ * Copyright 2023, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *     http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include <utils/Mutex.h>
+
+namespace android::afutils {
+
+inline bool dumpTryLock(Mutex& mutex)
+{
+    static constexpr int kDumpLockTimeoutNs = 1'000'000'000;
+    const status_t err = mutex.timedLock(kDumpLockTimeoutNs);
+    return err == NO_ERROR;
+}
+
+}  // android::afutils
\ No newline at end of file
diff --git a/services/audioflinger/afutils/Permission.cpp b/services/audioflinger/afutils/Permission.cpp
new file mode 100644
index 0000000..35448e3
--- /dev/null
+++ b/services/audioflinger/afutils/Permission.cpp
@@ -0,0 +1,54 @@
+/*
+ * 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.
+ */
+
+#define LOG_TAG "Permission"
+//#define LOG_NDEBUG 0
+
+#include "Permission.h"
+
+#include <binder/PermissionController.h>
+#include <media/AidlConversionCppNdk.h>
+#include <utils/Log.h>
+
+namespace android::afutils {
+
+// TODO b/182392769: use attribution source util
+content::AttributionSourceState checkAttributionSourcePackage(
+        const content::AttributionSourceState& attributionSource) {
+    Vector<String16> packages;
+    PermissionController{}.getPackagesForUid(attributionSource.uid, packages);
+
+    content::AttributionSourceState checkedAttributionSource = attributionSource;
+    if (!attributionSource.packageName.has_value()
+            || attributionSource.packageName.value().size() == 0) {
+        if (!packages.isEmpty()) {
+            checkedAttributionSource.packageName =
+                std::move(legacy2aidl_String16_string(packages[0]).value());
+        }
+    } else {
+        const String16 opPackageLegacy = VALUE_OR_FATAL(
+                aidl2legacy_string_view_String16(attributionSource.packageName.value_or("")));
+        if (std::find_if(packages.begin(), packages.end(),
+                [&opPackageLegacy](const auto& package) {
+                return opPackageLegacy == package; }) == packages.end()) {
+            ALOGW("The package name(%s) provided does not correspond to the uid %d",
+                    attributionSource.packageName.value_or("").c_str(), attributionSource.uid);
+        }
+    }
+    return checkedAttributionSource;
+}
+
+}  // namespace android::afutils
diff --git a/services/audioflinger/afutils/Permission.h b/services/audioflinger/afutils/Permission.h
new file mode 100644
index 0000000..97c7ff9
--- /dev/null
+++ b/services/audioflinger/afutils/Permission.h
@@ -0,0 +1,26 @@
+/*
+ * 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
+
+#include <android/content/AttributionSourceState.h>
+
+namespace android::afutils {
+
+content::AttributionSourceState checkAttributionSourcePackage(
+        const content::AttributionSourceState& attributionSource);
+
+}  // namespace android::afutils
diff --git a/services/audioflinger/datapath/AudioStreamIn.h b/services/audioflinger/datapath/AudioStreamIn.h
new file mode 100644
index 0000000..604a4e4
--- /dev/null
+++ b/services/audioflinger/datapath/AudioStreamIn.h
@@ -0,0 +1,60 @@
+/*
+ * 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
+
+#include "AudioHwDevice.h"
+#include <media/audiohal/DeviceHalInterface.h>
+#include <media/audiohal/StreamHalInterface.h>
+
+namespace android {
+
+// Abstraction for the Audio Source for the RecordThread (HAL or PassthruPatchRecord).
+struct Source {
+    virtual ~Source() = default;
+    // The following methods have the same signatures as in StreamHalInterface.
+    virtual status_t read(void* buffer, size_t bytes, size_t* read) = 0;
+    virtual status_t getCapturePosition(int64_t* frames, int64_t* time) = 0;
+    virtual status_t standby() = 0;
+};
+
+// AudioStreamIn is immutable, so its fields are const.
+// The methods must not be const to match StreamHalInterface signature.
+
+struct AudioStreamIn : public Source {
+    const AudioHwDevice* const audioHwDev;
+    const sp<StreamInHalInterface> stream;
+    const audio_input_flags_t flags;
+
+    AudioStreamIn(
+            const AudioHwDevice* dev, const sp<StreamInHalInterface>& in,
+            audio_input_flags_t flags)
+        : audioHwDev(dev), stream(in), flags(flags) {}
+
+    status_t read(void* buffer, size_t bytes, size_t* read) final {
+        return stream->read(buffer, bytes, read);
+    }
+
+    status_t getCapturePosition(int64_t* frames, int64_t* time) final {
+        return stream->getCapturePosition(frames, time);
+    }
+
+    status_t standby() final { return stream->standby(); }
+
+    sp<DeviceHalInterface> hwDev() const { return audioHwDev->hwDevice(); }
+};
+
+}  // namespace android
diff --git a/services/audioflinger/datapath/ThreadMetrics.h b/services/audioflinger/datapath/ThreadMetrics.h
index 5493b3c..c643a57 100644
--- a/services/audioflinger/datapath/ThreadMetrics.h
+++ b/services/audioflinger/datapath/ThreadMetrics.h
@@ -17,6 +17,8 @@
 #ifndef ANDROID_AUDIO_THREADMETRICS_H
 #define ANDROID_AUDIO_THREADMETRICS_H
 
+#include <media/MediaMetricsItem.h>
+
 #include <mutex>
 
 namespace android {
diff --git a/services/audioflinger/datapath/TrackMetrics.h b/services/audioflinger/datapath/TrackMetrics.h
index f3425df..2b44acb 100644
--- a/services/audioflinger/datapath/TrackMetrics.h
+++ b/services/audioflinger/datapath/TrackMetrics.h
@@ -20,6 +20,8 @@
 #include <binder/IActivityManager.h>
 #include <binder/IPCThreadState.h>
 #include <binder/IServiceManager.h>
+#include <media/MediaMetricsItem.h>
+
 #include <mutex>
 
 namespace android {
diff --git a/services/audioflinger/datapath/VolumeInterface.h b/services/audioflinger/datapath/VolumeInterface.h
new file mode 100644
index 0000000..1564fe1
--- /dev/null
+++ b/services/audioflinger/datapath/VolumeInterface.h
@@ -0,0 +1,34 @@
+/*
+ * 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
+
+#include <system/audio.h>
+
+namespace android {
+
+class VolumeInterface : public virtual RefBase {
+public:
+    virtual void setMasterVolume(float value) = 0;
+    virtual void setMasterBalance(float balance) = 0;
+    virtual void setMasterMute(bool muted) = 0;
+    virtual void setStreamVolume(audio_stream_type_t stream, float value) = 0;
+    virtual void setStreamMute(audio_stream_type_t stream, bool muted) = 0;
+    // TODO(b/290699744) add "get" prefix for getter below.
+    virtual float streamVolume(audio_stream_type_t stream) const = 0;
+};
+
+}  // namespace android
diff --git a/services/audioflinger/fastpath/FastMixerState.h b/services/audioflinger/fastpath/FastMixerState.h
index c70e42a..8ab6d25 100644
--- a/services/audioflinger/fastpath/FastMixerState.h
+++ b/services/audioflinger/fastpath/FastMixerState.h
@@ -35,7 +35,7 @@
 class VolumeProvider {
 public:
     // The provider implementation is responsible for validating that the return value is in range.
-    virtual gain_minifloat_packed_t getVolumeLR() = 0;
+    virtual gain_minifloat_packed_t getVolumeLR() const = 0;
 protected:
     VolumeProvider() = default;
     virtual ~VolumeProvider() = default;
diff --git a/services/audioflinger/sounddose/SoundDoseManager.cpp b/services/audioflinger/sounddose/SoundDoseManager.cpp
index 21f346e..97470a2 100644
--- a/services/audioflinger/sounddose/SoundDoseManager.cpp
+++ b/services/audioflinger/sounddose/SoundDoseManager.cpp
@@ -301,6 +301,25 @@
     return binder::Status::ok();
 }
 
+binder::Status SoundDoseManager::SoundDose::initCachedAudioDeviceCategories(
+        const std::vector<media::ISoundDose::AudioDeviceCategory>& btDeviceCategories) {
+    ALOGV("%s", __func__);
+    auto soundDoseManager = mSoundDoseManager.promote();
+    if (soundDoseManager != nullptr) {
+        soundDoseManager->initCachedAudioDeviceCategories(btDeviceCategories);
+    }
+    return binder::Status::ok();
+}
+binder::Status SoundDoseManager::SoundDose::setAudioDeviceCategory(
+        const media::ISoundDose::AudioDeviceCategory& btAudioDevice) {
+    ALOGV("%s", __func__);
+    auto soundDoseManager = mSoundDoseManager.promote();
+    if (soundDoseManager != nullptr) {
+        soundDoseManager->setAudioDeviceCategory(btAudioDevice);
+    }
+    return binder::Status::ok();
+}
+
 binder::Status SoundDoseManager::SoundDose::getOutputRs2UpperBound(float* value) {
     ALOGV("%s", __func__);
     auto soundDoseManager = mSoundDoseManager.promote();
@@ -358,7 +377,9 @@
         auto melProcessor = mp.second.promote();
         if (melProcessor != nullptr) {
             auto deviceId = melProcessor->getDeviceId();
-            if (mActiveDeviceTypes[deviceId] == deviceType) {
+            const auto deviceTypeIt = mActiveDeviceTypes.find(deviceId);
+            if (deviceTypeIt != mActiveDeviceTypes.end() &&
+                deviceTypeIt->second == deviceType) {
                 ALOGV("%s: set attenuation for deviceId %d to %f",
                         __func__, deviceId, attenuationDB);
                 melProcessor->setAttenuation(attenuationDB);
@@ -390,6 +411,103 @@
     return mEnabledCsd;
 }
 
+void SoundDoseManager::initCachedAudioDeviceCategories(
+        const std::vector<media::ISoundDose::AudioDeviceCategory>& deviceCategories) {
+    ALOGV("%s", __func__);
+    {
+        const std::lock_guard _l(mLock);
+        mBluetoothDevicesWithCsd.clear();
+    }
+    for (const auto& btDeviceCategory : deviceCategories) {
+        setAudioDeviceCategory(btDeviceCategory);
+    }
+}
+
+void SoundDoseManager::setAudioDeviceCategory(
+        const media::ISoundDose::AudioDeviceCategory& audioDevice) {
+    ALOGV("%s: set BT audio device type with address %s to headphone %d", __func__,
+          audioDevice.address.c_str(), audioDevice.csdCompatible);
+
+    std::vector<audio_port_handle_t> devicesToStart;
+    std::vector<audio_port_handle_t> devicesToStop;
+    {
+        const std::lock_guard _l(mLock);
+        const auto deviceIt = mBluetoothDevicesWithCsd.find(
+                std::make_pair(audioDevice.address,
+                               static_cast<audio_devices_t>(audioDevice.internalAudioType)));
+        if (deviceIt != mBluetoothDevicesWithCsd.end()) {
+            deviceIt->second = audioDevice.csdCompatible;
+        } else {
+            mBluetoothDevicesWithCsd.emplace(
+                    std::make_pair(audioDevice.address,
+                                   static_cast<audio_devices_t>(audioDevice.internalAudioType)),
+                    audioDevice.csdCompatible);
+        }
+
+        for (const auto &activeDevice: mActiveDevices) {
+            if (activeDevice.first.address() == audioDevice.address &&
+                activeDevice.first.mType ==
+                static_cast<audio_devices_t>(audioDevice.internalAudioType)) {
+                if (audioDevice.csdCompatible) {
+                    devicesToStart.push_back(activeDevice.second);
+                } else {
+                    devicesToStop.push_back(activeDevice.second);
+                }
+            }
+        }
+    }
+
+    for (const auto& deviceToStart : devicesToStart) {
+        mMelReporterCallback->startMelComputationForDeviceId(deviceToStart);
+    }
+    for (const auto& deviceToStop : devicesToStop) {
+        mMelReporterCallback->stopMelComputationForDeviceId(deviceToStop);
+    }
+}
+
+bool SoundDoseManager::shouldComputeCsdForDeviceType(audio_devices_t device) {
+    if (!isCsdEnabled()) {
+        ALOGV("%s csd is disabled", __func__);
+        return false;
+    }
+    if (forceComputeCsdOnAllDevices()) {
+        return true;
+    }
+
+    switch (device) {
+        case AUDIO_DEVICE_OUT_WIRED_HEADSET:
+        case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
+        // TODO(b/278265907): enable A2DP when we can distinguish A2DP headsets
+        // case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
+        case AUDIO_DEVICE_OUT_USB_HEADSET:
+        case AUDIO_DEVICE_OUT_BLE_HEADSET:
+        case AUDIO_DEVICE_OUT_BLE_BROADCAST:
+            return true;
+        default:
+            return false;
+    }
+}
+
+bool SoundDoseManager::shouldComputeCsdForDeviceWithAddress(const audio_devices_t type,
+                                                            const std::string& deviceAddress) {
+    if (!isCsdEnabled()) {
+        ALOGV("%s csd is disabled", __func__);
+        return false;
+    }
+    if (forceComputeCsdOnAllDevices()) {
+        return true;
+    }
+
+    if (!audio_is_ble_out_device(type) && !audio_is_a2dp_device(type)) {
+        return shouldComputeCsdForDeviceType(type);
+    }
+
+    const std::lock_guard _l(mLock);
+    const auto deviceIt = mBluetoothDevicesWithCsd.find(std::make_pair(deviceAddress, type));
+    return deviceIt != mBluetoothDevicesWithCsd.end() && deviceIt->second;
+}
+
 void SoundDoseManager::setUseFrameworkMel(bool useFrameworkMel) {
     // invalidate any HAL sound dose interface used
     setHalSoundDoseInterface(nullptr);
diff --git a/services/audioflinger/sounddose/SoundDoseManager.h b/services/audioflinger/sounddose/SoundDoseManager.h
index 9ed0661..888561f 100644
--- a/services/audioflinger/sounddose/SoundDoseManager.h
+++ b/services/audioflinger/sounddose/SoundDoseManager.h
@@ -32,6 +32,15 @@
 
 using aidl::android::hardware::audio::core::sounddose::ISoundDose;
 
+class IMelReporterCallback : public virtual RefBase {
+public:
+    IMelReporterCallback() {};
+    virtual ~IMelReporterCallback() {};
+
+    virtual void stopMelComputationForDeviceId(audio_port_handle_t deviceId) = 0;
+    virtual void startMelComputationForDeviceId(audio_port_handle_t deviceId) = 0;
+};
+
 class SoundDoseManager : public audio_utils::MelProcessor::MelCallback {
 public:
     /** CSD is computed with a rolling window of 7 days. */
@@ -39,8 +48,9 @@
     /** Default RS2 upper bound in dBA as defined in IEC 62368-1 3rd edition. */
     static constexpr float kDefaultRs2UpperBound = 100.f;
 
-    SoundDoseManager()
-        : mMelAggregator(sp<audio_utils::MelAggregator>::make(kCsdWindowSeconds)),
+    explicit SoundDoseManager(const sp<IMelReporterCallback>& melReporterCallback)
+        : mMelReporterCallback(melReporterCallback),
+          mMelAggregator(sp<audio_utils::MelAggregator>::make(kCsdWindowSeconds)),
           mRs2UpperBound(kDefaultRs2UpperBound) {};
 
     /**
@@ -104,6 +114,21 @@
     /** Returns true if CSD is enabled. */
     bool isCsdEnabled();
 
+    void initCachedAudioDeviceCategories(
+            const std::vector<media::ISoundDose::AudioDeviceCategory>& deviceCategories);
+
+    void setAudioDeviceCategory(
+            const media::ISoundDose::AudioDeviceCategory& audioDevice);
+
+    /**
+     * Returns true if the type can compute CSD. For bluetooth devices we rely on whether we
+     * categorized the address as headphones/headsets, only in this case we return true.
+     */
+    bool shouldComputeCsdForDeviceWithAddress(const audio_devices_t type,
+                                              const std::string& deviceAddress);
+    /** Returns true for all device types which could support CSD computation. */
+    bool shouldComputeCsdForDeviceType(audio_devices_t device);
+
     std::string dump() const;
 
     // used for testing only
@@ -139,6 +164,13 @@
         binder::Status getOutputRs2UpperBound(float* value) override;
         binder::Status setCsdEnabled(bool enabled) override;
 
+        binder::Status initCachedAudioDeviceCategories(
+                const std::vector<media::ISoundDose::AudioDeviceCategory> &btDeviceCategories)
+                override;
+
+        binder::Status setAudioDeviceCategory(
+                const media::ISoundDose::AudioDeviceCategory& btAudioDevice) override;
+
         binder::Status getCsd(float* value) override;
         binder::Status forceUseFrameworkMel(bool useFrameworkMel) override;
         binder::Status forceComputeCsdOnAllDevices(bool computeCsdOnAllDevices) override;
@@ -179,6 +211,8 @@
 
     mutable std::mutex mLock;
 
+    const sp<IMelReporterCallback> mMelReporterCallback;
+
     // no need for lock since MelAggregator is thread-safe
     const sp<audio_utils::MelAggregator> mMelAggregator;
 
@@ -191,6 +225,17 @@
     std::map<AudioDeviceTypeAddr, audio_port_handle_t> mActiveDevices GUARDED_BY(mLock);
     std::unordered_map<audio_port_handle_t, audio_devices_t> mActiveDeviceTypes GUARDED_BY(mLock);
 
+    struct bt_device_type_hash {
+        std::size_t operator() (const std::pair<std::string, audio_devices_t> &deviceType) const {
+            return std::hash<std::string>()(deviceType.first) ^
+                   std::hash<audio_devices_t>()(deviceType.second);
+        }
+    };
+    // storing the BT cached information as received from the java side
+    // see SoundDoseManager::setCachedAudioDeviceCategories
+    std::unordered_map<std::pair<std::string, audio_devices_t>, bool, bt_device_type_hash>
+            mBluetoothDevicesWithCsd GUARDED_BY(mLock);
+
     float mRs2UpperBound GUARDED_BY(mLock);
     std::unordered_map<audio_devices_t, float> mMelAttenuationDB GUARDED_BY(mLock);
 
diff --git a/services/audioflinger/sounddose/tests/sounddosemanager_tests.cpp b/services/audioflinger/sounddose/tests/sounddosemanager_tests.cpp
index 9fab77d..7d0b3a7 100644
--- a/services/audioflinger/sounddose/tests/sounddosemanager_tests.cpp
+++ b/services/audioflinger/sounddose/tests/sounddosemanager_tests.cpp
@@ -39,10 +39,18 @@
                 (const std::shared_ptr<ISoundDose::IHalSoundDoseCallback>&), (override));
 };
 
+class MelReporterCallback : public IMelReporterCallback {
+public:
+    MOCK_METHOD(void, startMelComputationForDeviceId, (audio_port_handle_t), (override));
+    MOCK_METHOD(void, stopMelComputationForDeviceId, (audio_port_handle_t), (override));
+};
+
+
 class SoundDoseManagerTest : public ::testing::Test {
 protected:
     void SetUp() override {
-        mSoundDoseManager = sp<SoundDoseManager>::make();
+        mMelReporterCallback = sp<MelReporterCallback>::make();
+        mSoundDoseManager = sp<SoundDoseManager>::make(mMelReporterCallback);
         mHalSoundDose = ndk::SharedRefBase::make<HalSoundDoseMock>();
 
         ON_CALL(*mHalSoundDose.get(), setOutputRs2UpperBound)
@@ -52,6 +60,7 @@
             });
     }
 
+    sp<MelReporterCallback> mMelReporterCallback;
     sp<SoundDoseManager> mSoundDoseManager;
     std::shared_ptr<HalSoundDoseMock> mHalSoundDose;
 };
@@ -243,5 +252,53 @@
     EXPECT_TRUE(mSoundDoseManager->forceUseFrameworkMel());
 }
 
+TEST_F(SoundDoseManagerTest, SetAudioDeviceCategoryStopsNonHeadphone) {
+    media::ISoundDose::AudioDeviceCategory device1;
+    device1.address = "dev1";
+    device1.csdCompatible = false;
+    device1.internalAudioType = AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+    const AudioDeviceTypeAddr dev1Adt{AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, device1.address};
+
+    // this will mark the device as active
+    mSoundDoseManager->mapAddressToDeviceId(dev1Adt, /*deviceId=*/1);
+    EXPECT_CALL(*mMelReporterCallback.get(), stopMelComputationForDeviceId).Times(1);
+
+    mSoundDoseManager->setAudioDeviceCategory(device1);
+}
+
+TEST_F(SoundDoseManagerTest, SetAudioDeviceCategoryStartsHeadphone) {
+    media::ISoundDose::AudioDeviceCategory device1;
+    device1.address = "dev1";
+    device1.csdCompatible = true;
+    device1.internalAudioType = AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+    const AudioDeviceTypeAddr dev1Adt{AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, device1.address};
+
+        // this will mark the device as active
+    mSoundDoseManager->mapAddressToDeviceId(dev1Adt, /*deviceId=*/1);
+    EXPECT_CALL(*mMelReporterCallback.get(), startMelComputationForDeviceId).Times(1);
+
+    mSoundDoseManager->setAudioDeviceCategory(device1);
+}
+
+TEST_F(SoundDoseManagerTest, InitCachedAudioDevicesStartsOnlyActiveDevices) {
+    media::ISoundDose::AudioDeviceCategory device1;
+    media::ISoundDose::AudioDeviceCategory device2;
+    device1.address = "dev1";
+    device1.csdCompatible = true;
+    device1.internalAudioType = AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+    device2.address = "dev2";
+    device2.csdCompatible = true;
+    device2.internalAudioType = AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+    const AudioDeviceTypeAddr dev1Adt{AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, device1.address};
+    std::vector<media::ISoundDose::AudioDeviceCategory> btDevices = {device1, device2};
+
+    // this will mark the device as active
+    mSoundDoseManager->mapAddressToDeviceId(dev1Adt, /*deviceId=*/1);
+    EXPECT_CALL(*mMelReporterCallback.get(), startMelComputationForDeviceId).Times(1);
+
+    mSoundDoseManager->initCachedAudioDeviceCategories(btDevices);
+}
+
+
 }  // namespace
 }  // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
index 876911d..1e57edd 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -102,9 +102,13 @@
     void setVolume(float volumeDb) { mCurVolumeDb = volumeDb; }
     float getVolume() const { return mCurVolumeDb; }
 
+    void setIsVoice(bool isVoice) { mIsVoice = isVoice; }
+    bool isVoice() const { return mIsVoice; }
+
 private:
     int mMuteCount = 0; /**< mute request counter */
     float mCurVolumeDb = NAN; /**< current volume in dB. */
+    bool mIsVoice = false; /** true if this volume source is used for voice call volume */
 };
 /**
  * Note: volume activities shall be indexed by CurvesId if we want to allow multiple
@@ -162,7 +166,8 @@
                            VolumeSource volumeSource, const StreamTypeVector &streams,
                            const DeviceTypeSet& deviceTypes,
                            uint32_t delayMs,
-                           bool force);
+                           bool force,
+                           bool isVoiceVolSrc = false);
 
     /**
      * @brief setStopTime set the stop time due to the client stoppage or a re routing of this
@@ -222,17 +227,25 @@
     {
         return mVolumeActivities[vs].decMuteCount();
     }
-    void setCurVolume(VolumeSource vs, float volumeDb)
+    void setCurVolume(VolumeSource vs, float volumeDb, bool isVoiceVolSrc)
     {
         // Even if not activity for this source registered, need to create anyway
         mVolumeActivities[vs].setVolume(volumeDb);
+        mVolumeActivities[vs].setIsVoice(isVoiceVolSrc);
     }
     float getCurVolume(VolumeSource vs) const
     {
         return mVolumeActivities.find(vs) != std::end(mVolumeActivities) ?
                     mVolumeActivities.at(vs).getVolume() : NAN;
     }
-
+    VolumeSource getVoiceSource() {
+        for (const auto &iter : mVolumeActivities) {
+            if (iter.second.isVoice()) {
+                return iter.first;
+            }
+        }
+        return VOLUME_SOURCE_NONE;
+    }
     bool isStrategyActive(product_strategy_t ps, uint32_t inPastMs = 0, nsecs_t sysTime = 0) const
     {
         return mRoutingActivities.find(ps) != std::end(mRoutingActivities)?
@@ -381,7 +394,8 @@
                            VolumeSource volumeSource, const StreamTypeVector &streams,
                            const DeviceTypeSet& device,
                            uint32_t delayMs,
-                           bool force);
+                           bool force,
+                           bool isVoiceVolSrc = false);
 
     virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
                            const struct audio_port_config *srcConfig = NULL) const;
@@ -424,6 +438,15 @@
     bool supportsAllDevices(const DeviceVector &devices) const;
 
     /**
+     * @brief supportsAtLeastOne checks if any device in devices is currently supported
+     * @param devices to be checked against
+     * @return true if the device is weakly supported by type (e.g. for non bus / rsubmix devices),
+     *         true if the device is supported (both type and address) for bus / remote submix
+     *         false otherwise
+     */
+    bool supportsAtLeastOne(const DeviceVector &devices) const;
+
+    /**
      * @brief supportsDevicesForPlayback
      * @param devices to be checked against
      * @return true if the devices is a supported combo for playback
@@ -475,7 +498,8 @@
                            VolumeSource volumeSource, const StreamTypeVector &streams,
                            const DeviceTypeSet& deviceTypes,
                            uint32_t delayMs,
-                           bool force);
+                           bool force,
+                           bool isVoiceVolSrc = false);
 
     virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
                            const struct audio_port_config *srcConfig = NULL) const;
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
index 92292e1..7e29e10 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
@@ -138,7 +138,7 @@
      */
     status_t setUserIdDeviceAffinities(int userId, const AudioDeviceTypeAddrVector& devices);
     status_t removeUserIdDeviceAffinities(int userId);
-    status_t getDevicesForUserId(int userId, Vector<AudioDeviceTypeAddr>& devices) const;
+    status_t getDevicesForUserId(int userId, AudioDeviceTypeAddrVector& devices) const;
 
     void dump(String8 *dst) const;
 
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 4877166..2f424b8 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -163,7 +163,8 @@
                                       const StreamTypeVector &/*streams*/,
                                       const DeviceTypeSet& deviceTypes,
                                       uint32_t delayMs,
-                                      bool force)
+                                      bool force,
+                                      bool isVoiceVolSrc)
 {
 
     if (!supportedDevices().containsDeviceAmongTypes(deviceTypes)) {
@@ -176,7 +177,7 @@
     // - the force flag is set
     if (volumeDb != getCurVolume(volumeSource) || force) {
         ALOGV("%s for volumeSrc %d, volume %f, delay %d", __func__, volumeSource, volumeDb, delayMs);
-        setCurVolume(volumeSource, volumeDb);
+        setCurVolume(volumeSource, volumeDb, isVoiceVolSrc);
         return true;
     }
     return false;
@@ -389,6 +390,11 @@
     return supportedDevices().containsAllDevices(devices);
 }
 
+bool SwAudioOutputDescriptor::supportsAtLeastOne(const DeviceVector &devices) const
+{
+    return filterSupportedDevices(devices).size() > 0;
+}
+
 bool SwAudioOutputDescriptor::supportsDevicesForPlayback(const DeviceVector &devices) const
 {
     // No considering duplicated output
@@ -505,11 +511,12 @@
                                         VolumeSource vs, const StreamTypeVector &streamTypes,
                                         const DeviceTypeSet& deviceTypes,
                                         uint32_t delayMs,
-                                        bool force)
+                                        bool force,
+                                        bool isVoiceVolSrc)
 {
     StreamTypeVector streams = streamTypes;
     if (!AudioOutputDescriptor::setVolume(
-            volumeDb, muted, vs, streamTypes, deviceTypes, delayMs, force)) {
+            volumeDb, muted, vs, streamTypes, deviceTypes, delayMs, force, isVoiceVolSrc)) {
         return false;
     }
     if (streams.empty()) {
@@ -555,6 +562,10 @@
     float volumeAmpl = Volume::DbToAmpl(getCurVolume(vs));
     if (hasStream(streams, AUDIO_STREAM_BLUETOOTH_SCO)) {
         mClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volumeAmpl, mIoHandle, delayMs);
+        VolumeSource callVolSrc = getVoiceSource();
+        if (callVolSrc != VOLUME_SOURCE_NONE) {
+            setCurVolume(callVolSrc, getCurVolume(vs), true);
+        }
     }
     for (const auto &stream : streams) {
         ALOGV("%s output %d for volumeSource %d, volume %f, delay %d stream=%s", __func__,
@@ -783,10 +794,11 @@
                                         VolumeSource volumeSource, const StreamTypeVector &streams,
                                         const DeviceTypeSet& deviceTypes,
                                         uint32_t delayMs,
-                                        bool force)
+                                        bool force,
+                                        bool isVoiceVolSrc)
 {
     bool changed = AudioOutputDescriptor::setVolume(
-            volumeDb, muted, volumeSource, streams, deviceTypes, delayMs, force);
+            volumeDb, muted, volumeSource, streams, deviceTypes, delayMs, force, isVoiceVolSrc);
 
     if (changed) {
       // TODO: use gain controller on source device if any to adjust volume
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
index 6b9757d..bc2ba31 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
@@ -642,7 +642,7 @@
 }
 
 status_t AudioPolicyMixCollection::getDevicesForUserId(int userId,
-        Vector<AudioDeviceTypeAddr>& devices) const {
+        AudioDeviceTypeAddrVector& devices) const {
     // for each player mix:
     // find rules that don't exclude this userId, and add the device to the list
     for (size_t i = 0; i < size(); i++) {
@@ -660,7 +660,7 @@
             }
         }
         if (ruleAllowsUserId) {
-            devices.add(AudioDeviceTypeAddr(mix->mDeviceType, mix->mDeviceAddress.string()));
+            devices.push_back(AudioDeviceTypeAddr(mix->mDeviceType, mix->mDeviceAddress.string()));
         }
     }
     return NO_ERROR;
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioProfileVectorHelper.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioProfileVectorHelper.cpp
index 8ccb8b9..82f51ad 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioProfileVectorHelper.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioProfileVectorHelper.cpp
@@ -115,12 +115,22 @@
         profile->setDynamicFormat(true);
         profile->setDynamicChannels(dynamicFormatProfile->isDynamicChannels());
         profile->setDynamicRate(dynamicFormatProfile->isDynamicRate());
-        addAudioProfileAndSort(audioProfileVector, profile);
+        size_t profileIndex = 0;
+        for (; profileIndex < audioProfileVector.size(); profileIndex++) {
+            if (profile->equals(audioProfileVector.at(profileIndex))) {
+                // The dynamic profile is already there
+                break;
+            }
+        }
+        if (profileIndex >= audioProfileVector.size()) {
+            // Only add when the dynamic profile is not there
+            addAudioProfileAndSort(audioProfileVector, profile);
+        }
     }
 }
 
 void addDynamicAudioProfileAndSort(AudioProfileVector &audioProfileVector,
-                                      const sp<AudioProfile> &profileToAdd)
+                                   const sp<AudioProfile> &profileToAdd)
 {
     // Check valid profile to add:
     if (!profileToAdd->hasValidFormat()) {
@@ -143,11 +153,15 @@
                 audioProfileVector, profileToAdd->getChannels(), profileToAdd->getFormat());
         return;
     }
+    const bool originalIsDynamicFormat = profileToAdd->isDynamicFormat();
+    profileToAdd->setDynamicFormat(true); // set the format as dynamic to allow removal
     // Go through the list of profile to avoid duplicates
     for (size_t profileIndex = 0; profileIndex < audioProfileVector.size(); profileIndex++) {
         const sp<AudioProfile> &profile = audioProfileVector.at(profileIndex);
-        if (profile->isValid() && profile == profileToAdd) {
-            // Nothing to do
+        if (profile->isValid() && profile->equals(profileToAdd)) {
+            // The same profile is already there, no need to add.
+            // Reset `isDynamicProfile` as original value.
+            profileToAdd->setDynamicFormat(originalIsDynamicFormat);
             return;
         }
     }
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index fad89d9..44648de 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -1298,7 +1298,8 @@
         if (outputDevices.size() == 1) {
             info = getPreferredMixerAttributesInfo(
                     outputDevices.itemAt(0)->getId(),
-                    mEngine->getProductStrategyForAttributes(*resultAttr));
+                    mEngine->getProductStrategyForAttributes(*resultAttr),
+                    true /*activeBitPerfectPreferred*/);
             // Only use preferred mixer if the uid matches or the preferred mixer is bit-perfect
             // and it is currently active.
             if (info != nullptr && info->getUid() != uid &&
@@ -2152,6 +2153,26 @@
                 return DEAD_OBJECT;
             }
             info->increaseActiveClient();
+            if (info->getActiveClientCount() == 1 &&
+                (info->getFlags() & AUDIO_OUTPUT_FLAG_BIT_PERFECT) != AUDIO_OUTPUT_FLAG_NONE) {
+                // If it is first bit-perfect client, reroute all clients that will be routed to
+                // the bit-perfect sink so that it is guaranteed only bit-perfect stream is active.
+                PortHandleVector clientsToInvalidate;
+                for (size_t i = 0; i < mOutputs.size(); i++) {
+                    if (mOutputs[i] == outputDesc ||
+                        mOutputs[i]->devices().filter(outputDesc->devices()).isEmpty()) {
+                        continue;
+                    }
+                    for (const auto& c : mOutputs[i]->getClientIterable()) {
+                        clientsToInvalidate.push_back(c->portId());
+                    }
+                }
+                if (!clientsToInvalidate.empty()) {
+                    ALOGD("%s Invalidate clients due to first bit-perfect client started",
+                          __func__);
+                    mpClientInterface->invalidateTracks(clientsToInvalidate);
+                }
+            }
         }
     }
 
@@ -3787,6 +3808,44 @@
     return true;
 }
 
+void AudioPolicyManager::changeOutputDevicesMuteState(
+        const AudioDeviceTypeAddrVector& devices) {
+    ALOGVV("%s() num devices %zu", __func__, devices.size());
+
+    std::vector<sp<SwAudioOutputDescriptor>> outputs =
+            getSoftwareOutputsForDevices(devices);
+
+    for (size_t i = 0; i < outputs.size(); i++) {
+        sp<SwAudioOutputDescriptor> outputDesc = outputs[i];
+        DeviceVector prevDevices = outputDesc->devices();
+        checkDeviceMuteStrategies(outputDesc, prevDevices, 0 /* delayMs */);
+    }
+}
+
+std::vector<sp<SwAudioOutputDescriptor>> AudioPolicyManager::getSoftwareOutputsForDevices(
+        const AudioDeviceTypeAddrVector& devices) const
+{
+    std::vector<sp<SwAudioOutputDescriptor>> outputs;
+    DeviceVector deviceDescriptors;
+    for (size_t j = 0; j < devices.size(); j++) {
+        sp<DeviceDescriptor> desc = mHwModules.getDeviceDescriptor(
+                devices[j].mType, devices[j].getAddress(), String8(), AUDIO_FORMAT_DEFAULT);
+        if (desc == nullptr || !audio_is_output_device(devices[j].mType)) {
+            ALOGE("%s: device type %#x address %s not supported or not an output device",
+                __func__, devices[j].mType, devices[j].getAddress());
+                    continue;
+        }
+        deviceDescriptors.add(desc);
+    }
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        if (!mOutputs.valueAt(i)->supportsAtLeastOne(deviceDescriptors)) {
+            continue;
+        }
+        outputs.push_back(mOutputs.valueAt(i));
+    }
+    return outputs;
+}
+
 status_t AudioPolicyManager::setUidDeviceAffinities(uid_t uid,
         const AudioDeviceTypeAddrVector& devices) {
     ALOGV("%s() uid=%d num devices %zu", __FUNCTION__, uid, devices.size());
@@ -3853,7 +3912,8 @@
     return NO_ERROR;
 }
 
-void AudioPolicyManager::updateCallAndOutputRouting(bool forceVolumeReeval, uint32_t delayMs)
+void AudioPolicyManager::updateCallAndOutputRouting(bool forceVolumeReeval, uint32_t delayMs,
+    bool skipDelays)
 {
     uint32_t waitMs = 0;
     bool wasLeUnicastActive = isLeUnicastActive();
@@ -3879,8 +3939,8 @@
                 continue;
             }
             waitMs = setOutputDevices(outputDesc, newDevices, forceRouting, delayMs, nullptr,
-                                      true /*requiresMuteCheck*/,
-                                      !forceRouting /*requiresVolumeCheck*/);
+                                      !skipDelays /*requiresMuteCheck*/,
+                                      !forceRouting /*requiresVolumeCheck*/, skipDelays);
             // Only apply special touch sound delay once
             delayMs = 0;
         }
@@ -4065,13 +4125,18 @@
 
     // reevaluate outputs for all devices
     checkForDeviceAndOutputChanges();
-    updateCallAndOutputRouting();
+    changeOutputDevicesMuteState(devices);
+    updateCallAndOutputRouting(false /* forceVolumeReeval */, 0 /* delayMs */,
+        true /* skipDelays */);
+    changeOutputDevicesMuteState(devices);
 
     return NO_ERROR;
 }
 
 status_t AudioPolicyManager::removeUserIdDeviceAffinities(int userId) {
     ALOGV("%s() userId=%d", __FUNCTION__, userId);
+    AudioDeviceTypeAddrVector devices;
+    mPolicyMixes.getDevicesForUserId(userId, devices);
     status_t status = mPolicyMixes.removeUserIdDeviceAffinities(userId);
     if (status != NO_ERROR) {
         ALOGE("%s() Could not remove all device affinities fo userId = %d",
@@ -4081,7 +4146,10 @@
 
     // reevaluate outputs for all devices
     checkForDeviceAndOutputChanges();
-    updateCallAndOutputRouting();
+    changeOutputDevicesMuteState(devices);
+    updateCallAndOutputRouting(false /* forceVolumeReeval */, 0 /* delayMs */,
+        true /* skipDelays */);
+    changeOutputDevicesMuteState(devices);
 
     return NO_ERROR;
 }
@@ -4490,16 +4558,24 @@
 }
 
 sp<PreferredMixerAttributesInfo> AudioPolicyManager::getPreferredMixerAttributesInfo(
-        audio_port_handle_t devicePortId, product_strategy_t strategy) {
+        audio_port_handle_t devicePortId,
+        product_strategy_t strategy,
+        bool activeBitPerfectPreferred) {
     auto it = mPreferredMixerAttrInfos.find(devicePortId);
     if (it == mPreferredMixerAttrInfos.end()) {
         return nullptr;
     }
-    auto mixerAttrInfoIt = it->second.find(strategy);
-    if (mixerAttrInfoIt == it->second.end()) {
-        return nullptr;
+    if (activeBitPerfectPreferred) {
+        for (auto [strategy, info] : it->second) {
+            if ((info->getFlags() & AUDIO_OUTPUT_FLAG_BIT_PERFECT) != AUDIO_OUTPUT_FLAG_NONE
+                && info->getActiveClientCount() != 0) {
+                return info;
+            }
+        }
     }
-    return mixerAttrInfoIt->second;
+    auto strategyMatchedMixerAttrInfoIt = it->second.find(strategy);
+    return strategyMatchedMixerAttrInfoIt == it->second.end()
+            ? nullptr : strategyMatchedMixerAttrInfoIt->second;
 }
 
 status_t AudioPolicyManager::getPreferredMixerAttributes(
@@ -5841,22 +5917,26 @@
         }
     }
 
+    // The caller can have the audio config criteria ignored by either passing a null ptr or
+    // the AUDIO_CONFIG_INITIALIZER value.
+    // If an audio config is specified, current policy is to only allow spatialization for
+    // some positional channel masks and PCM format
+
+    if (config != nullptr && *config != AUDIO_CONFIG_INITIALIZER) {
+        if (!audio_is_channel_mask_spatialized(config->channel_mask)) {
+            return false;
+        }
+        if (!audio_is_linear_pcm(config->format)) {
+            return false;
+        }
+    }
+
     sp<IOProfile> profile =
             getSpatializerOutputProfile(config, devices);
     if (profile == nullptr) {
         return false;
     }
 
-    // The caller can have the audio config criteria ignored by either passing a null ptr or
-    // the AUDIO_CONFIG_INITIALIZER value.
-    // If an audio config is specified, current policy is to only allow spatialization for
-    // some positional channel masks.
-
-    if (config != nullptr && *config != AUDIO_CONFIG_INITIALIZER) {
-        if (!audio_is_channel_mask_spatialized(config->channel_mask)) {
-            return false;
-        }
-    }
     return true;
 }
 
@@ -7320,7 +7400,8 @@
                                               bool force,
                                               int delayMs,
                                               audio_patch_handle_t *patchHandle,
-                                              bool requiresMuteCheck, bool requiresVolumeCheck)
+                                              bool requiresMuteCheck, bool requiresVolumeCheck,
+                                              bool skipMuteDelay)
 {
     // TODO(b/262404095): Consider if the output need to be reopened.
     ALOGV("%s device %s delayMs %d", __func__, devices.toString().c_str(), delayMs);
@@ -7328,9 +7409,9 @@
 
     if (outputDesc->isDuplicated()) {
         muteWaitMs = setOutputDevices(outputDesc->subOutput1(), devices, force, delayMs,
-                nullptr /* patchHandle */, requiresMuteCheck);
+                nullptr /* patchHandle */, requiresMuteCheck, skipMuteDelay);
         muteWaitMs += setOutputDevices(outputDesc->subOutput2(), devices, force, delayMs,
-                nullptr /* patchHandle */, requiresMuteCheck);
+                nullptr /* patchHandle */, requiresMuteCheck, skipMuteDelay);
         return muteWaitMs;
     }
 
@@ -7396,12 +7477,16 @@
 
         // Add half reported latency to delayMs when muteWaitMs is null in order
         // to avoid disordered sequence of muting volume and changing devices.
-        installPatch(__func__, patchHandle, outputDesc.get(), patchBuilder.patch(),
-                muteWaitMs == 0 ? (delayMs + (outputDesc->latency() / 2)) : delayMs);
+        int actualDelayMs = !skipMuteDelay && muteWaitMs == 0
+                ? (delayMs + (outputDesc->latency() / 2)) : delayMs;
+        installPatch(__func__, patchHandle, outputDesc.get(), patchBuilder.patch(), actualDelayMs);
     }
 
-    // update stream volumes according to new device
-    applyStreamVolumes(outputDesc, filteredDevices.types(), delayMs);
+    // Since the mute is skip, also skip the apply stream volume as that will be applied externally
+    if (!skipMuteDelay) {
+        // update stream volumes according to new device
+        applyStreamVolumes(outputDesc, filteredDevices.types(), delayMs);
+    }
 
     return muteWaitMs;
 }
@@ -7759,8 +7844,8 @@
         volumeDb = 0.0f;
     }
     const bool muted = (index == 0) && (volumeDb != 0.0f);
-    outputDesc->setVolume(
-            volumeDb, muted, volumeSource, curves.getStreamTypes(), deviceTypes, delayMs, force);
+    outputDesc->setVolume(volumeDb, muted, volumeSource, curves.getStreamTypes(),
+            deviceTypes, delayMs, force, isVoiceVolSrc);
 
     if (outputDesc == mPrimaryOutput && (isVoiceVolSrc || isBtScoVolSrc)) {
         float voiceVolume;
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 88bafef..863c785 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -535,8 +535,9 @@
          *        and currently active, allow to have proper drain and avoid pops
          * @param requiresVolumeCheck true if called requires to reapply volume if the routing did
          * not change (but the output is still routed).
+         * @param skipMuteDelay if true will skip mute delay when installing audio patch
          * @return the number of ms we have slept to allow new routing to take effect in certain
-         * cases.
+         *        cases.
          */
         uint32_t setOutputDevices(const sp<SwAudioOutputDescriptor>& outputDesc,
                                   const DeviceVector &device,
@@ -544,7 +545,8 @@
                                   int delayMs = 0,
                                   audio_patch_handle_t *patchHandle = NULL,
                                   bool requiresMuteCheck = true,
-                                  bool requiresVolumeCheck = false);
+                                  bool requiresVolumeCheck = false,
+                                  bool skipMuteDelay = false);
         status_t resetOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
                                    int delayMs = 0,
                                    audio_patch_handle_t *patchHandle = NULL);
@@ -647,8 +649,10 @@
         /**
          * @brief updates routing for all outputs (including call if call in progress).
          * @param delayMs delay for unmuting if required
+         * @param skipDelays if true all the delays will be skip while updating routing
          */
-        void updateCallAndOutputRouting(bool forceVolumeReeval = true, uint32_t delayMs = 0);
+        void updateCallAndOutputRouting(bool forceVolumeReeval = true, uint32_t delayMs = 0,
+                bool skipDelays = false);
 
         bool isCallRxAudioSource(const sp<SourceClientDescriptor> &source) {
             return mCallRxSourceClient != nullptr && source == mCallRxSourceClient;
@@ -1241,6 +1245,21 @@
                 const char* context,
                 bool matchAddress = true);
 
+        /**
+         * @brief changeOutputDevicesMuteState mute/unmute devices using checkDeviceMuteStrategies
+         * @param devices devices to mute/unmute
+         */
+        void changeOutputDevicesMuteState(const AudioDeviceTypeAddrVector& devices);
+
+        /**
+         * @brief Returns a vector of software output descriptor that support the queried devices
+         * @param devices devices to query
+         * @param openOutputs open outputs where the devices are supported as determined by
+         *      SwAudioOutputDescriptor::supportsAtLeastOne
+         */
+        std::vector<sp<SwAudioOutputDescriptor>> getSoftwareOutputsForDevices(
+                const AudioDeviceTypeAddrVector& devices) const;
+
         bool isScoRequestedForComm() const;
 
         bool isHearingAidUsedForComm() const;
@@ -1298,8 +1317,15 @@
                                        uint32_t flags,
                                        bool isInput);
 
+        /**
+         * Returns the preferred mixer attributes info for the given device port id and strategy.
+         * Bit-perfect mixer attributes will be returned if it is active and
+         * `activeBitPerfectPreferred` is true.
+         */
         sp<PreferredMixerAttributesInfo> getPreferredMixerAttributesInfo(
-                audio_port_handle_t devicePortId, product_strategy_t strategy);
+                audio_port_handle_t devicePortId,
+                product_strategy_t strategy,
+                bool activeBitPerfectPreferred = false);
 
         sp<SwAudioOutputDescriptor> reopenOutput(
                 sp<SwAudioOutputDescriptor> outputDesc,
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index 5d22ed4..15aced0 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -25,6 +25,7 @@
 #include <sys/time.h>
 #include <dlfcn.h>
 
+#include <android/content/pm/IPackageManagerNative.h>
 #include <audio_utils/clock.h>
 #include <binder/IServiceManager.h>
 #include <utils/Log.h>
@@ -215,6 +216,27 @@
 {
     delete interface;
 }
+
+namespace {
+int getTargetSdkForPackageName(std::string_view packageName) {
+    const auto binder = defaultServiceManager()->checkService(String16{"package_native"});
+    int targetSdk = -1;
+    if (binder != nullptr) {
+        const auto pm = interface_cast<content::pm::IPackageManagerNative>(binder);
+        if (pm != nullptr) {
+            const auto status = pm->getTargetSdkVersionForPackage(
+                    String16{packageName.data(), packageName.size()}, &targetSdk);
+            ALOGI("Capy check package %s, sdk %d", packageName.data(), targetSdk);
+            return status.isOk() ? targetSdk : -1;
+        }
+    }
+    return targetSdk;
+}
+
+bool doesPackageTargetAtLeastU(std::string_view packageName) {
+    return getTargetSdkForPackageName(packageName) >= __ANDROID_API_U__;
+}
+} // anonymous
 // ----------------------------------------------------------------------------
 
 AudioPolicyService::AudioPolicyService()
@@ -1926,10 +1948,14 @@
     checkOp();
     mOpCallback = new RecordAudioOpCallback(this);
     ALOGV("start watching op %d for %s", mAppOp, mAttributionSource.toString().c_str());
+    int flags = doesPackageTargetAtLeastU(
+            mAttributionSource.packageName.value_or("")) ?
+            AppOpsManager::WATCH_FOREGROUND_CHANGES : 0;
     // TODO: We need to always watch AppOpsManager::OP_RECORD_AUDIO too
     // since it controls the mic permission for legacy apps.
     mAppOpsManager.startWatchingMode(mAppOp, VALUE_OR_FATAL(aidl2legacy_string_view_String16(
         mAttributionSource.packageName.value_or(""))),
+        flags,
         mOpCallback);
 }
 
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index 5e58dbb..15eae14 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -1232,6 +1232,19 @@
     EXPECT_FALSE(isBitPerfect);
     EXPECT_EQ(bitPerfectOutput, output);
 
+    const audio_attributes_t dtmfAttr = {
+            .content_type = AUDIO_CONTENT_TYPE_UNKNOWN,
+            .usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING,
+    };
+    audio_io_handle_t dtmfOutput = AUDIO_IO_HANDLE_NONE;
+    selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+    portId = AUDIO_PORT_HANDLE_NONE;
+    getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+            48000, AUDIO_OUTPUT_FLAG_NONE, &dtmfOutput, &portId, dtmfAttr,
+            AUDIO_SESSION_NONE, anotherUid, &isBitPerfect);
+    EXPECT_FALSE(isBitPerfect);
+    EXPECT_EQ(bitPerfectOutput, dtmfOutput);
+
     // When configuration matches preferred mixer attributes, which is bit-perfect, but the client
     // is not the owner of preferred mixer attributes, the playback will not be bit-perfect.
     getOutputForAttr(&selectedDeviceId, bitPerfectFormat, bitPerfectChannelMask,
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index bf85029..1b1662b 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -316,14 +316,24 @@
     for (auto& i : mListenerList) {
         if (shouldSkipStatusUpdates(systemCameraKind, i->isVendorListener(), i->getListenerPid(),
                 i->getListenerUid())) {
-            ALOGV("Skipping torch callback for system-only camera device %s",
-                    cameraId.c_str());
+            ALOGV("%s: Skipping torch callback for system-only camera device %s",
+                    __FUNCTION__, cameraId.c_str());
             continue;
         }
         auto ret = i->getListener()->onTorchStatusChanged(mapToInterface(status),
                 String16{cameraId});
         i->handleBinderStatus(ret, "%s: Failed to trigger onTorchStatusChanged for %d:%d: %d",
                 __FUNCTION__, i->getListenerUid(), i->getListenerPid(), ret.exceptionCode());
+        // Also trigger the torch callbacks for cameras that were remapped to the current cameraId
+        // for the specific package that this listener belongs to.
+        std::vector<String8> remappedCameraIds =
+                findOriginalIdsForRemappedCameraId(cameraId, i->getListenerUid());
+        for (auto& remappedCameraId : remappedCameraIds) {
+            ret = i->getListener()->onTorchStatusChanged(mapToInterface(status),
+                    String16(remappedCameraId));
+            i->handleBinderStatus(ret, "%s: Failed to trigger onTorchStatusChanged for %d:%d: %d",
+                    __FUNCTION__, i->getListenerUid(), i->getListenerPid(), ret.exceptionCode());
+        }
     }
 }
 
@@ -745,6 +755,156 @@
     return Status::ok();
 }
 
+Status CameraService::remapCameraIds(const hardware::CameraIdRemapping&
+      cameraIdRemapping) {
+    if (!checkCallingPermission(sCameraInjectExternalCameraPermission)) {
+        const int pid = CameraThreadState::getCallingPid();
+        const int uid = CameraThreadState::getCallingUid();
+        ALOGE("%s: Permission Denial: can't configure camera ID mapping pid=%d, uid=%d",
+                __FUNCTION__, pid, uid);
+        return STATUS_ERROR(ERROR_PERMISSION_DENIED,
+                "Permission Denial: no permission to configure camera id mapping");
+    }
+    TCameraIdRemapping cameraIdRemappingMap{};
+    binder::Status parseStatus = parseCameraIdRemapping(cameraIdRemapping, &cameraIdRemappingMap);
+    if (!parseStatus.isOk()) {
+        return parseStatus;
+    }
+    remapCameraIds(cameraIdRemappingMap);
+    return Status::ok();
+}
+
+Status CameraService::parseCameraIdRemapping(
+        const hardware::CameraIdRemapping& cameraIdRemapping,
+        /* out */ TCameraIdRemapping* cameraIdRemappingMap) {
+    String16 packageName;
+    String8 cameraIdToReplace, updatedCameraId;
+    for(const auto& packageIdRemapping: cameraIdRemapping.packageIdRemappings) {
+        packageName = packageIdRemapping.packageName;
+        if (packageName == String16("")) {
+            return STATUS_ERROR(ERROR_ILLEGAL_ARGUMENT,
+                    "CameraIdRemapping: Package name cannot be empty");
+        }
+
+        if (packageIdRemapping.cameraIdsToReplace.size()
+            != packageIdRemapping.updatedCameraIds.size()) {
+            return STATUS_ERROR_FMT(ERROR_ILLEGAL_ARGUMENT,
+                    "CameraIdRemapping: Mismatch in CameraId Remapping lists sizes for package %s",
+                     String8(packageName).c_str());
+        }
+        for(size_t i = 0; i < packageIdRemapping.cameraIdsToReplace.size(); i++) {
+            cameraIdToReplace = String8(packageIdRemapping.cameraIdsToReplace[i]);
+            updatedCameraId = String8(packageIdRemapping.updatedCameraIds[i]);
+            if (cameraIdToReplace == String8("") || updatedCameraId == String8("")) {
+                return STATUS_ERROR_FMT(ERROR_ILLEGAL_ARGUMENT,
+                        "CameraIdRemapping: Camera Id cannot be empty for package %s",
+                        String8(packageName).c_str());
+            }
+            if (cameraIdToReplace == updatedCameraId) {
+                return STATUS_ERROR_FMT(ERROR_ILLEGAL_ARGUMENT,
+                        "CameraIdRemapping: CameraIdToReplace cannot be the same"
+                        " as updatedCameraId for %s",
+                        String8(packageName).c_str());
+            }
+            (*cameraIdRemappingMap)[packageName][cameraIdToReplace] = updatedCameraId;
+        }
+    }
+    return Status::ok();
+}
+
+void CameraService::remapCameraIds(const TCameraIdRemapping& cameraIdRemapping) {
+    // Acquire mServiceLock and prevent other clients from connecting
+    std::unique_ptr<AutoConditionLock> serviceLockWrapper =
+            AutoConditionLock::waitAndAcquire(mServiceLockWrapper);
+
+    Mutex::Autolock lock(mCameraIdRemappingLock);
+    // This will disconnect all existing clients for camera Ids that are being
+    // remapped in cameraIdRemapping, but only if they were being used by an
+    // affected packageName.
+    std::vector<sp<BasicClient>> clientsToDisconnect;
+    std::vector<String8> cameraIdsToUpdate;
+    for (const auto& [packageName, injectionMap] : cameraIdRemapping) {
+        for (auto& [id0, id1] : injectionMap) {
+            ALOGI("%s: UPDATE:= %s: %s: %s", __FUNCTION__, String8(packageName).c_str(),
+                    id0.c_str(), id1.c_str());
+            auto clientDescriptor = mActiveClientManager.get(id0);
+            if (clientDescriptor != nullptr) {
+                sp<BasicClient> clientSp = clientDescriptor->getValue();
+                if (clientSp->getPackageName() == packageName) {
+                    // This camera ID is being used by the affected packageName.
+                    clientsToDisconnect.push_back(clientSp);
+                    cameraIdsToUpdate.push_back(id0);
+                }
+            }
+        }
+    }
+
+    // Update mCameraIdRemapping.
+    mCameraIdRemapping.clear();
+    mCameraIdRemapping.insert(cameraIdRemapping.begin(), cameraIdRemapping.end());
+
+    // Do not hold mServiceLock while disconnecting clients, but retain the condition
+    // blocking other clients from connecting in mServiceLockWrapper if held.
+    mServiceLock.unlock();
+
+    // Disconnect clients.
+    for (auto& clientSp : clientsToDisconnect) {
+        // We send up ERROR_CAMERA_DEVICE so that the app attempts to reconnect
+        // automatically.
+        clientSp->notifyError(hardware::camera2::ICameraDeviceCallbacks::ERROR_CAMERA_DEVICE,
+                CaptureResultExtras{});
+        // This also triggers the status updates
+        clientSp->disconnect();
+    }
+
+    mServiceLock.lock();
+}
+
+std::vector<String8> CameraService::findOriginalIdsForRemappedCameraId(
+    const String8& inputCameraId, int clientUid) {
+    String16 packageName = getPackageNameFromUid(clientUid);
+    std::vector<String8> cameraIds;
+    Mutex::Autolock lock(mCameraIdRemappingLock);
+    if (auto packageMapIter = mCameraIdRemapping.find(packageName);
+        packageMapIter != mCameraIdRemapping.end()) {
+        for (auto& [id0, id1]: packageMapIter->second) {
+            if (id1 == inputCameraId) {
+                cameraIds.push_back(id0);
+            }
+        }
+    }
+    return cameraIds;
+}
+
+String8 CameraService::resolveCameraId(const String8& inputCameraId) {
+  return resolveCameraId(inputCameraId, String16(""));
+}
+
+String8 CameraService::resolveCameraId(
+    const String8& inputCameraId,
+    const String16& packageName) {
+    String16 packageNameVal = packageName;
+    if (packageName == String16("")) {
+        int clientUid = CameraThreadState::getCallingUid();
+        packageNameVal = getPackageNameFromUid(clientUid);
+    }
+    Mutex::Autolock lock(mCameraIdRemappingLock);
+    if (auto packageMapIter = mCameraIdRemapping.find(packageNameVal);
+        packageMapIter != mCameraIdRemapping.end()) {
+        ALOGI("%s: resolveCameraId: packageName found %s",
+                __FUNCTION__, String8(packageNameVal).c_str());
+        auto packageMap = packageMapIter->second;
+        if (auto replacementIdIter = packageMap.find(inputCameraId);
+            replacementIdIter != packageMap.end()) {
+            ALOGI("%s: resolveCameraId: inputId found %s, replacing with %s",
+                    __FUNCTION__, inputCameraId.c_str(),
+                    replacementIdIter->second.c_str());
+            return replacementIdIter->second;
+        }
+    }
+    return inputCameraId;
+}
+
 Status CameraService::getCameraInfo(int cameraId, bool overrideToPortrait,
         CameraInfo* cameraInfo) {
     ATRACE_CALL();
@@ -815,9 +975,10 @@
     return String8(cameraIdIntToStrLocked(cameraIdInt).c_str());
 }
 
-Status CameraService::getCameraCharacteristics(const String16& cameraId,
+Status CameraService::getCameraCharacteristics(const String16& unresolvedCameraId,
         int targetSdkVersion, bool overrideToPortrait, CameraMetadata* cameraInfo) {
     ATRACE_CALL();
+    String8 cameraId = resolveCameraId(String8(unresolvedCameraId));
     if (!cameraInfo) {
         ALOGE("%s: cameraInfo is NULL", __FUNCTION__);
         return STATUS_ERROR(ERROR_ILLEGAL_ARGUMENT, "cameraInfo is NULL");
@@ -837,8 +998,7 @@
 
     Status ret{};
 
-
-    std::string cameraIdStr = String8(cameraId).string();
+    std::string cameraIdStr = cameraId.string();
     bool overrideForPerfClass =
             SessionConfigurationUtils::targetPerfClassPrimaryCamera(mPerfClassPrimaryCameraIds,
                     cameraIdStr, targetSdkVersion);
@@ -1028,7 +1188,7 @@
         int api1CameraId, int facing, int sensorOrientation, int clientPid, uid_t clientUid,
         int servicePid, std::pair<int, IPCTransport> deviceVersionAndTransport,
         apiLevel effectiveApiLevel, bool overrideForPerfClass, bool overrideToPortrait,
-        bool forceSlowJpegMode, /*out*/sp<BasicClient>* client) {
+        bool forceSlowJpegMode, const String8& originalCameraId, /*out*/sp<BasicClient>* client) {
     // For HIDL devices
     if (deviceVersionAndTransport.second == IPCTransport::HIDL) {
         // Create CameraClient based on device version reported by the HAL.
@@ -1071,7 +1231,7 @@
         *client = new CameraDeviceClient(cameraService, tmp,
                 cameraService->mCameraServiceProxyWrapper, packageName, systemNativeClient,
                 featureId, cameraId, facing, sensorOrientation, clientPid, clientUid, servicePid,
-                overrideForPerfClass, overrideToPortrait);
+                overrideForPerfClass, overrideToPortrait, originalCameraId);
         ALOGI("%s: Camera2 API, override to portrait %d", __FUNCTION__, overrideToPortrait);
     }
     return Status::ok();
@@ -1163,7 +1323,7 @@
             internalPackageName, /*systemNativeClient*/ false, {}, uid, USE_CALLING_PID,
             API_1, /*shimUpdateOnly*/ true, /*oomScoreOffset*/ 0,
             /*targetSdkVersion*/ __ANDROID_API_FUTURE__, /*overrideToPortrait*/ true,
-            /*forceSlowJpegMode*/false, /*out*/ tmp)
+            /*forceSlowJpegMode*/false, id, /*out*/ tmp)
             ).isOk()) {
         ALOGE("%s: Error initializing shim metadata: %s", __FUNCTION__, ret.toString8().string());
     }
@@ -1700,7 +1860,7 @@
     ret = connectHelper<ICameraClient,Client>(cameraClient, id, api1CameraId,
             clientPackageName,/*systemNativeClient*/ false, {}, clientUid, clientPid, API_1,
             /*shimUpdateOnly*/ false, /*oomScoreOffset*/ 0, targetSdkVersion,
-            overrideToPortrait, forceSlowJpegMode, /*out*/client);
+            overrideToPortrait, forceSlowJpegMode, id, /*out*/client);
 
     if(!ret.isOk()) {
         logRejected(id, CameraThreadState::getCallingPid(), String8(clientPackageName),
@@ -1780,7 +1940,7 @@
 
 Status CameraService::connectDevice(
         const sp<hardware::camera2::ICameraDeviceCallbacks>& cameraCb,
-        const String16& cameraId,
+        const String16& unresolvedCameraId,
         const String16& clientPackageName,
         const std::optional<String16>& clientFeatureId,
         int clientUid, int oomScoreOffset, int targetSdkVersion,
@@ -1790,7 +1950,7 @@
 
     ATRACE_CALL();
     Status ret = Status::ok();
-    String8 id = String8(cameraId);
+    String8 id = resolveCameraId(String8(unresolvedCameraId), clientPackageName);
     sp<CameraDeviceClient> client = nullptr;
     String16 clientPackageNameAdj = clientPackageName;
     int callingPid = CameraThreadState::getCallingPid();
@@ -1840,7 +2000,7 @@
             /*api1CameraId*/-1, clientPackageNameAdj, systemNativeClient,clientFeatureId,
             clientUid, USE_CALLING_PID, API_2, /*shimUpdateOnly*/ false, oomScoreOffset,
             targetSdkVersion, overrideToPortrait, /*forceSlowJpegMode*/false,
-            /*out*/client);
+            String8(unresolvedCameraId), /*out*/client);
 
     if(!ret.isOk()) {
         logRejected(id, callingPid, String8(clientPackageNameAdj), ret.toString8());
@@ -1909,7 +2069,7 @@
         int api1CameraId, const String16& clientPackageNameMaybe, bool systemNativeClient,
         const std::optional<String16>& clientFeatureId, int clientUid, int clientPid,
         apiLevel effectiveApiLevel, bool shimUpdateOnly, int oomScoreOffset, int targetSdkVersion,
-        bool overrideToPortrait, bool forceSlowJpegMode,
+        bool overrideToPortrait, bool forceSlowJpegMode, const String8& originalCameraId,
         /*out*/sp<CLIENT>& device) {
     binder::Status ret = binder::Status::ok();
 
@@ -2026,7 +2186,7 @@
                 clientFeatureId, cameraId, api1CameraId, facing, orientation,
                 clientPid, clientUid, getpid(),
                 deviceVersionAndTransport, effectiveApiLevel, overrideForPerfClass,
-                overrideToPortrait, forceSlowJpegMode,
+                overrideToPortrait, forceSlowJpegMode, originalCameraId,
                 /*out*/&tmp)).isOk()) {
             return ret;
         }
@@ -2040,6 +2200,9 @@
         if (err != OK) {
             ALOGE("%s: Could not initialize client from HAL.", __FUNCTION__);
             // Errors could be from the HAL module open call or from AppOpsManager
+            mServiceLock.unlock();
+            client->disconnect();
+            mServiceLock.lock();
             switch(err) {
                 case BAD_VALUE:
                     return STATUS_ERROR_FMT(ERROR_ILLEGAL_ARGUMENT,
@@ -2278,7 +2441,8 @@
     return OK;
 }
 
-Status CameraService::turnOnTorchWithStrengthLevel(const String16& cameraId, int32_t torchStrength,
+Status CameraService::turnOnTorchWithStrengthLevel(const String16& unresolvedCameraId,
+        int32_t torchStrength,
         const sp<IBinder>& clientBinder) {
     Mutex::Autolock lock(mServiceLock);
 
@@ -2289,7 +2453,7 @@
                 "Torch client binder in null.");
     }
 
-    String8 id = String8(cameraId.string());
+    String8 id = resolveCameraId(String8(unresolvedCameraId));
     int uid = CameraThreadState::getCallingUid();
 
     if (shouldRejectSystemCameraConnection(id)) {
@@ -2349,7 +2513,7 @@
 
     {
         Mutex::Autolock al(mTorchUidMapMutex);
-        updateTorchUidMapLocked(cameraId, uid);
+        updateTorchUidMapLocked(String16(id), uid);
     }
     // Check if the current torch strength level is same as the new one.
     bool shouldSkipTorchStrengthUpdates = mCameraProviderManager->shouldSkipTorchStrengthUpdate(
@@ -2408,7 +2572,8 @@
     return Status::ok();
 }
 
-Status CameraService::setTorchMode(const String16& cameraId, bool enabled,
+Status CameraService::setTorchMode(const String16& unresolvedCameraId,
+        bool enabled,
         const sp<IBinder>& clientBinder) {
     Mutex::Autolock lock(mServiceLock);
 
@@ -2419,7 +2584,7 @@
                 "Torch client Binder is null");
     }
 
-    String8 id = String8(cameraId.string());
+    String8 id = resolveCameraId(String8(unresolvedCameraId));
     int uid = CameraThreadState::getCallingUid();
 
     if (shouldRejectSystemCameraConnection(id)) {
@@ -2479,7 +2644,7 @@
         // Update UID map - this is used in the torch status changed callbacks, so must be done
         // before setTorchMode
         Mutex::Autolock al(mTorchUidMapMutex);
-        updateTorchUidMapLocked(cameraId, uid);
+        updateTorchUidMapLocked(String16(id), uid);
     }
 
     status_t err = mFlashlight->setTorchMode(id, enabled);
@@ -2949,11 +3114,20 @@
     return ret;
 }
 
-Status CameraService::supportsCameraApi(const String16& cameraId, int apiVersion,
+Status CameraService::supportsCameraApi(const String16& unresolvedCameraId, int apiVersion,
         /*out*/ bool *isSupported) {
     ATRACE_CALL();
 
-    const String8 id = String8(cameraId);
+    String8 resolvedId;
+    if (apiVersion == API_VERSION_2) {
+        resolvedId = resolveCameraId(String8(unresolvedCameraId));
+    } else { // if (apiVersion == API_VERSION_1)
+        // We don't support remapping for API 1.
+        // TODO(b/286287541): Also support remapping for API 1.
+        resolvedId = String8(unresolvedCameraId);
+    }
+
+    const String8 id = resolvedId;
 
     ALOGV("%s: for camera ID = %s", __FUNCTION__, id.string());
 
@@ -3013,11 +3187,11 @@
     return Status::ok();
 }
 
-Status CameraService::isHiddenPhysicalCamera(const String16& cameraId,
+Status CameraService::isHiddenPhysicalCamera(const String16& unresolvedCameraId,
         /*out*/ bool *isSupported) {
     ATRACE_CALL();
 
-    const String8 id = String8(cameraId);
+    const String8 id = resolveCameraId(String8(unresolvedCameraId));
 
     ALOGV("%s: for camera ID = %s", __FUNCTION__, id.string());
     *isSupported = mCameraProviderManager->isHiddenPhysicalCamera(id.string());
@@ -4960,7 +5134,6 @@
     state->updateStatus(status, cameraId, rejectSourceStates, [this, &deviceKind,
                         &logicalCameraIds]
             (const String8& cameraId, StatusInternal status) {
-
             if (status != StatusInternal::ENUMERATING) {
                 // Update torch status if it has a flash unit.
                 Mutex::Autolock al(mTorchStatusMutex);
@@ -4993,9 +5166,21 @@
                 auto ret = listener->getListener()->onStatusChanged(mapToInterface(status),
                         String16(cameraId));
                 listener->handleBinderStatus(ret,
-                        "%s: Failed to trigger onStatusChanged callback for %d:%d: %d",
+                         "%s: Failed to trigger onStatusChanged callback for %d:%d: %d",
                         __FUNCTION__, listener->getListenerUid(), listener->getListenerPid(),
                         ret.exceptionCode());
+                // Also trigger the callbacks for cameras that were remapped to the current
+                // cameraId for the specific package that this listener belongs to.
+                std::vector<String8> remappedCameraIds =
+                        findOriginalIdsForRemappedCameraId(cameraId, listener->getListenerUid());
+                for (auto& remappedCameraId : remappedCameraIds) {
+                    ret = listener->getListener()->onStatusChanged(
+                            mapToInterface(status), String16(remappedCameraId));
+                    listener->handleBinderStatus(ret,
+                             "%s: Failed to trigger onStatusChanged callback for %d:%d: %d",
+                            __FUNCTION__, listener->getListenerUid(), listener->getListenerPid(),
+                            ret.exceptionCode());
+                }
             }
         });
 }
@@ -5206,6 +5391,8 @@
         return handleWatchCommand(args, in, out);
     } else if (args.size() >= 2 && args[0] == String16("set-watchdog")) {
         return handleSetCameraServiceWatchdog(args);
+    } else if (args.size() >= 4 && args[0] == String16("remap-camera-id")) {
+        return handleCameraIdRemapping(args, err);
     } else if (args.size() == 1 && args[0] == String16("help")) {
         printHelp(out);
         return OK;
@@ -5214,6 +5401,23 @@
     return BAD_VALUE;
 }
 
+status_t CameraService::handleCameraIdRemapping(const Vector<String16>& args, int err) {
+    uid_t uid = IPCThreadState::self()->getCallingUid();
+    if (uid != AID_ROOT) {
+        dprintf(err, "Must be adb root\n");
+        return PERMISSION_DENIED;
+    }
+    if (args.size() != 4) {
+        dprintf(err, "Expected format: remap-camera-id <PACKAGE> <Id0> <Id1>\n");
+        return BAD_VALUE;
+    }
+    String16 packageName = args[1];
+    String8 cameraIdToReplace = String8(args[2]);
+    String8 cameraIdNew = String8(args[3]);
+    remapCameraIds({{packageName, {{cameraIdToReplace, cameraIdNew}}}});
+    return OK;
+}
+
 status_t CameraService::handleSetUidState(const Vector<String16>& args, int err) {
     String16 packageName = args[1];
 
@@ -5841,6 +6045,7 @@
         "  clear-stream-use-case-override clear the stream use case override\n"
         "  set-zoom-override <-1/0/1> enable or disable zoom override\n"
         "      Valid values -1: do not override, 0: override to OFF, 1: override to ZOOM\n"
+        "  remap-camera-id <PACKAGE> <Id0> <Id1> remaps camera ids. Must use adb root\n"
         "  watch <start|stop|dump|print|clear> manages tag monitoring in connected clients\n"
         "  help print this message\n");
 }
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index 3214d4c..65b11e7 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -20,6 +20,7 @@
 #include <android/hardware/BnCameraService.h>
 #include <android/hardware/BnSensorPrivacyListener.h>
 #include <android/hardware/ICameraServiceListener.h>
+#include <android/hardware/CameraIdRemapping.h>
 #include <android/hardware/camera2/BnCameraInjectionSession.h>
 #include <android/hardware/camera2/ICameraInjectionCallback.h>
 
@@ -61,6 +62,7 @@
 #include <utility>
 #include <unordered_map>
 #include <unordered_set>
+#include <vector>
 
 namespace android {
 
@@ -138,6 +140,9 @@
 
     /////////////////////////////////////////////////////////////////////
     // ICameraService
+    // IMPORTANT: All binder calls that deal with logicalCameraId should use
+    // resolveCameraId(logicalCameraId) to arrive at the correct cameraId to
+    // perform the operation on (in case of Id Remapping).
     virtual binder::Status     getNumberOfCameras(int32_t type, int32_t* numCameras);
 
     virtual binder::Status     getCameraInfo(int cameraId, bool overrideToPortrait,
@@ -221,6 +226,9 @@
     virtual binder::Status reportExtensionSessionStats(
             const hardware::CameraExtensionSessionStats& stats, String16* sessionKey /*out*/);
 
+    virtual binder::Status remapCameraIds(const hardware::CameraIdRemapping&
+        cameraIdRemapping);
+
     // Extra permissions checks
     virtual status_t    onTransact(uint32_t code, const Parcel& data,
                                    Parcel* reply, uint32_t flags);
@@ -916,7 +924,7 @@
             int api1CameraId, const String16& clientPackageNameMaybe, bool systemNativeClient,
             const std::optional<String16>& clientFeatureId, int clientUid, int clientPid,
             apiLevel effectiveApiLevel, bool shimUpdateOnly, int scoreOffset, int targetSdkVersion,
-            bool overrideToPortrait, bool forceSlowJpegMode,
+            bool overrideToPortrait, bool forceSlowJpegMode, const String8& originalCameraId,
             /*out*/sp<CLIENT>& device);
 
     // Lock guarding camera service state
@@ -943,6 +951,48 @@
     // Mutex guarding mCameraStates map
     mutable Mutex mCameraStatesLock;
 
+    /**
+     * Mapping from packageName -> {cameraIdToReplace -> newCameraIdtoUse}.
+     *
+     * This specifies that for packageName, for every binder operation targeting
+     * cameraIdToReplace, use newCameraIdToUse instead.
+     */
+    typedef std::map<String16, std::map<String8, String8>> TCameraIdRemapping;
+    TCameraIdRemapping mCameraIdRemapping{};
+    /** Mutex guarding mCameraIdRemapping. */
+    Mutex mCameraIdRemappingLock;
+
+    /** Parses cameraIdRemapping parcelable into the native cameraIdRemappingMap. */
+    binder::Status parseCameraIdRemapping(
+            const hardware::CameraIdRemapping& cameraIdRemapping,
+            /* out */ TCameraIdRemapping* cameraIdRemappingMap);
+
+    /**
+     * Resolve the (potentially remapped) camera Id to use for packageName.
+     *
+     * This returns the Camera Id to use in case inputCameraId was remapped to a
+     * different Id for the given packageName. Otherwise, it returns the inputCameraId.
+     */
+    String8 resolveCameraId(const String8& inputCameraId, const String16& packageName);
+    /**
+     * Resolve the (potentially remapped) camera Id to use.
+     *
+     * This returns the Camera Id to use in case inputCameraId was remapped to a
+     * different Id for the packageName of the client. Otherwise, it returns the inputCameraId.
+     */
+    String8 resolveCameraId(const String8& inputCameraId);
+
+    /**
+     * Updates the state of mCameraIdRemapping, while disconnecting active clients as necessary.
+     */
+    void remapCameraIds(const TCameraIdRemapping& cameraIdRemapping);
+
+    /**
+     * Finds the Camera Ids that were remapped to the inputCameraId for the given client.
+     */
+    std::vector<String8> findOriginalIdsForRemappedCameraId(
+        const String8& inputCameraId, int clientUid);
+
     // Circular buffer for storing event logging for dumps
     RingBuffer<String8> mEventLog;
     Mutex mLogLock;
@@ -1322,6 +1372,9 @@
     // Set or clear the zoom override flag
     status_t handleSetZoomOverride(const Vector<String16>& args);
 
+    // Set Camera Id remapping using 'cmd'
+    status_t handleCameraIdRemapping(const Vector<String16>& args, int errFd);
+
     // Handle 'watch' command as passed through 'cmd'
     status_t handleWatchCommand(const Vector<String16> &args, int inFd, int outFd);
 
@@ -1367,14 +1420,15 @@
      */
     static String8 getFormattedCurrentTime();
 
-    static binder::Status makeClient(const sp<CameraService>& cameraService,
-            const sp<IInterface>& cameraCb, const String16& packageName,
-            bool systemNativeClient, const std::optional<String16>& featureId,
-            const String8& cameraId, int api1CameraId, int facing, int sensorOrientation,
-            int clientPid, uid_t clientUid, int servicePid,
+    static binder::Status makeClient(
+            const sp<CameraService>& cameraService, const sp<IInterface>& cameraCb,
+            const String16& packageName, bool systemNativeClient,
+            const std::optional<String16>& featureId, const String8& cameraId, int api1CameraId,
+            int facing, int sensorOrientation, int clientPid, uid_t clientUid, int servicePid,
             std::pair<int, IPCTransport> deviceVersionAndIPCTransport, apiLevel effectiveApiLevel,
             bool overrideForPerfClass, bool overrideToPortrait, bool forceSlowJpegMode,
-            /*out*/sp<BasicClient>* client);
+            const String8& originalCameraId,
+            /*out*/ sp<BasicClient>* client);
 
     status_t checkCameraAccess(const String16& opPackageName);
 
diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp
index 8348cd9..5b5892a 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.cpp
+++ b/services/camera/libcameraservice/api1/Camera2Client.cpp
@@ -70,7 +70,9 @@
                 cameraDeviceId, api1CameraId, cameraFacing, sensorOrientation, clientPid,
                 clientUid, servicePid, overrideForPerfClass, overrideToPortrait,
                 /*legacyClient*/ true),
-        mParameters(api1CameraId, cameraFacing)
+        mParameters(api1CameraId, cameraFacing),
+        mLatestRequestIds(kMaxRequestIds),
+        mLatestFailedRequestIds(kMaxRequestIds)
 {
     ATRACE_CALL();
 
@@ -1843,7 +1845,7 @@
                     (hardware::camera2::ICameraDeviceCallbacks::ERROR_CAMERA_RESULT == errorCode)) {
                 Mutex::Autolock al(mLatestRequestMutex);
 
-                mLatestFailedRequestId = resultExtras.requestId;
+                mLatestFailedRequestIds.add(resultExtras.requestId);
                 mLatestRequestSignal.signal();
             }
             mCaptureSequencer->notifyError(errorCode, resultExtras);
@@ -2418,7 +2420,10 @@
 
 status_t Camera2Client::waitUntilRequestIdApplied(int32_t requestId, nsecs_t timeout) {
     Mutex::Autolock l(mLatestRequestMutex);
-    while ((mLatestRequestId != requestId) && (mLatestFailedRequestId != requestId)) {
+    while ((std::find(mLatestRequestIds.begin(), mLatestRequestIds.end(), requestId) ==
+            mLatestRequestIds.end()) &&
+           (std::find(mLatestFailedRequestIds.begin(), mLatestFailedRequestIds.end(), requestId) ==
+            mLatestFailedRequestIds.end())) {
         nsecs_t startTime = systemTime();
 
         auto res = mLatestRequestSignal.waitRelative(mLatestRequestMutex, timeout);
@@ -2427,13 +2432,14 @@
         timeout -= (systemTime() - startTime);
     }
 
-    return (mLatestRequestId == requestId) ? OK : DEAD_OBJECT;
+    return (std::find(mLatestRequestIds.begin(), mLatestRequestIds.end(), requestId) !=
+             mLatestRequestIds.end()) ? OK : DEAD_OBJECT;
 }
 
 void Camera2Client::notifyRequestId(int32_t requestId) {
     Mutex::Autolock al(mLatestRequestMutex);
 
-    mLatestRequestId = requestId;
+    mLatestRequestIds.add(requestId);
     mLatestRequestSignal.signal();
 }
 
diff --git a/services/camera/libcameraservice/api1/Camera2Client.h b/services/camera/libcameraservice/api1/Camera2Client.h
index 6bdb644..a7ea823 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.h
+++ b/services/camera/libcameraservice/api1/Camera2Client.h
@@ -22,11 +22,7 @@
 #include "common/Camera2ClientBase.h"
 #include "api1/client2/Parameters.h"
 #include "api1/client2/FrameProcessor.h"
-//#include "api1/client2/StreamingProcessor.h"
-//#include "api1/client2/JpegProcessor.h"
-//#include "api1/client2/ZslProcessor.h"
-//#include "api1/client2/CaptureSequencer.h"
-//#include "api1/client2/CallbackProcessor.h"
+#include <media/RingBuffer.h>
 
 namespace android {
 
@@ -263,8 +259,8 @@
 
     mutable Mutex mLatestRequestMutex;
     Condition mLatestRequestSignal;
-    int32_t mLatestRequestId = -1;
-    int32_t mLatestFailedRequestId = -1;
+    static constexpr size_t kMaxRequestIds = BufferQueueDefs::NUM_BUFFER_SLOTS;
+    RingBuffer<int32_t> mLatestRequestIds, mLatestFailedRequestIds;
     status_t waitUntilRequestIdApplied(int32_t requestId, nsecs_t timeout);
     status_t waitUntilCurrentRequestIdLocked();
 };
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
index 38c615d..1720b55 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
@@ -98,7 +98,8 @@
         uid_t clientUid,
         int servicePid,
         bool overrideForPerfClass,
-        bool overrideToPortrait) :
+        bool overrideToPortrait,
+        const String8& originalCameraId) :
     Camera2ClientBase(cameraService, remoteCallback, cameraServiceProxyWrapper, clientPackageName,
             systemNativeClient, clientFeatureId, cameraId, /*API1 camera ID*/ -1, cameraFacing,
             sensorOrientation, clientPid, clientUid, servicePid, overrideForPerfClass,
@@ -106,8 +107,8 @@
     mInputStream(),
     mStreamingRequestId(REQUEST_ID_NONE),
     mRequestIdCounter(0),
-    mOverrideForPerfClass(overrideForPerfClass) {
-
+    mOverrideForPerfClass(overrideForPerfClass),
+    mOriginalCameraId(originalCameraId) {
     ATRACE_CALL();
     ALOGI("CameraDeviceClient %s: Opened", cameraId.string());
 }
@@ -322,7 +323,7 @@
 
         //The first capture settings should always match the logical camera id
         String8 logicalId(request.mPhysicalCameraSettings.begin()->id.c_str());
-        if (mDevice->getId() != logicalId) {
+        if (mDevice->getId() != logicalId && mOriginalCameraId != logicalId) {
             ALOGE("%s: Camera %s: Invalid camera request settings.", __FUNCTION__,
                     mCameraIdStr.string());
             return STATUS_ERROR(CameraService::ERROR_ILLEGAL_ARGUMENT,
@@ -437,6 +438,8 @@
 
         CameraDeviceBase::PhysicalCameraSettingsList physicalSettingsList;
         for (const auto& it : request.mPhysicalCameraSettings) {
+            std::string resolvedId = (
+                mOriginalCameraId.string() == it.id) ? mDevice->getId().string() : it.id;
             if (it.settings.isEmpty()) {
                 ALOGE("%s: Camera %s: Sent empty metadata packet. Rejecting request.",
                         __FUNCTION__, mCameraIdStr.string());
@@ -447,7 +450,7 @@
             // Check whether the physical / logical stream has settings
             // consistent with the sensor pixel mode(s) it was configured with.
             // mCameraIdToStreamSet will only have ids that are high resolution
-            const auto streamIdSetIt = mHighResolutionCameraIdToStreamIdSet.find(it.id);
+            const auto streamIdSetIt = mHighResolutionCameraIdToStreamIdSet.find(resolvedId);
             if (streamIdSetIt != mHighResolutionCameraIdToStreamIdSet.end()) {
                 std::list<int> streamIdsUsedInRequest = getIntersection(streamIdSetIt->second,
                         outputStreamIds);
@@ -455,14 +458,14 @@
                         !isSensorPixelModeConsistent(streamIdsUsedInRequest, it.settings)) {
                      ALOGE("%s: Camera %s: Request settings CONTROL_SENSOR_PIXEL_MODE not "
                             "consistent with configured streams. Rejecting request.",
-                            __FUNCTION__, it.id.c_str());
+                            __FUNCTION__, resolvedId.c_str());
                     return STATUS_ERROR(CameraService::ERROR_ILLEGAL_ARGUMENT,
                         "Request settings CONTROL_SENSOR_PIXEL_MODE are not consistent with "
                         "streams configured");
                 }
             }
 
-            String8 physicalId(it.id.c_str());
+            String8 physicalId(resolvedId.c_str());
             bool hasTestPatternModePhysicalKey = std::find(mSupportedPhysicalRequestKeys.begin(),
                     mSupportedPhysicalRequestKeys.end(), ANDROID_SENSOR_TEST_PATTERN_MODE) !=
                     mSupportedPhysicalRequestKeys.end();
@@ -471,7 +474,7 @@
                     mSupportedPhysicalRequestKeys.end();
             if (physicalId != mDevice->getId()) {
                 auto found = std::find(requestedPhysicalIds.begin(), requestedPhysicalIds.end(),
-                        it.id);
+                        resolvedId);
                 if (found == requestedPhysicalIds.end()) {
                     ALOGE("%s: Camera %s: Physical camera id: %s not part of attached outputs.",
                             __FUNCTION__, mCameraIdStr.string(), physicalId.string());
@@ -494,11 +497,11 @@
                         }
                     }
 
-                    physicalSettingsList.push_back({it.id, filteredParams,
+                    physicalSettingsList.push_back({resolvedId, filteredParams,
                             hasTestPatternModePhysicalKey, hasTestPatternDataPhysicalKey});
                 }
             } else {
-                physicalSettingsList.push_back({it.id, it.settings});
+                physicalSettingsList.push_back({resolvedId, it.settings});
             }
         }
 
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.h b/services/camera/libcameraservice/api2/CameraDeviceClient.h
index 1533cf5..95563ee 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.h
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.h
@@ -191,7 +191,8 @@
             uid_t clientUid,
             int servicePid,
             bool overrideForPerfClass,
-            bool overrideToPortrait);
+            bool overrideToPortrait,
+            const String8& originalCameraId);
     virtual ~CameraDeviceClient();
 
     virtual status_t      initialize(sp<CameraProviderManager> manager,
@@ -368,6 +369,9 @@
     std::string mUserTag;
     // The last set video stabilization mode
     int mVideoStabilizationMode = -1;
+
+    // This only exists in case of camera ID Remapping.
+    String8 mOriginalCameraId;
 };
 
 }; // namespace android
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index 61c3298..71e49fd 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -3030,6 +3030,7 @@
         mNotifyPipelineDrain(false),
         mFrameNumber(0),
         mLatestRequestId(NAME_NOT_FOUND),
+        mLatestFailedRequestId(NAME_NOT_FOUND),
         mCurrentAfTriggerId(0),
         mCurrentPreCaptureTriggerId(0),
         mRotateAndCropOverride(ANDROID_SCALER_ROTATE_AND_CROP_NONE),
@@ -3281,7 +3282,7 @@
     ATRACE_CALL();
     Mutex::Autolock l(mLatestRequestMutex);
     status_t res;
-    while (mLatestRequestId != requestId) {
+    while (mLatestRequestId != requestId && mLatestFailedRequestId != requestId) {
         nsecs_t startTime = systemTime();
 
         res = mLatestRequestSignal.waitRelative(mLatestRequestMutex, timeout);
@@ -4010,8 +4011,11 @@
                 sp<Camera3Device> parent = mParent.promote();
                 if (parent != nullptr) {
                     const String8& streamCameraId = outputStream->getPhysicalCameraId();
+                    // Consider the case where clients are sending a single logical camera request
+                    // to physical output/outputs
+                    bool singleRequest = captureRequest->mSettingsList.size() == 1;
                     for (const auto& settings : captureRequest->mSettingsList) {
-                        if ((streamCameraId.isEmpty() &&
+                        if (((streamCameraId.isEmpty() || singleRequest) &&
                                 parent->getId() == settings.cameraId.c_str()) ||
                                 streamCameraId == settings.cameraId.c_str()) {
                             outputStream->fireBufferRequestForFrameNumber(
@@ -4356,6 +4360,12 @@
                         hardware::camera2::ICameraDeviceCallbacks::ERROR_CAMERA_REQUEST,
                         captureRequest->mResultExtras);
             }
+            {
+                Mutex::Autolock al(mLatestRequestMutex);
+
+                mLatestFailedRequestId = captureRequest->mResultExtras.requestId;
+                mLatestRequestSignal.signal();
+            }
         }
 
         // Remove yet-to-be submitted inflight request from inflightMap
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index b1dd135..aa1d55a 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -1142,6 +1142,7 @@
         Condition          mLatestRequestSignal;
         // android.request.id for latest process_capture_request
         int32_t            mLatestRequestId;
+        int32_t            mLatestFailedRequestId;
         CameraMetadata     mLatestRequest;
         std::unordered_map<std::string, CameraMetadata> mLatestPhysicalRequest;