diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 94f10e5..c2dcd35 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -75,6 +75,7 @@
 }
 
 AudioStreamInternal::~AudioStreamInternal() {
+    ALOGD("%s() %p called", __func__, this);
 }
 
 aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) {
@@ -270,21 +271,21 @@
     return result;
 
 error:
-    releaseCloseFinal();
+    safeReleaseClose();
     return result;
 }
 
 // This must be called under mStreamLock.
 aaudio_result_t AudioStreamInternal::release_l() {
     aaudio_result_t result = AAUDIO_OK;
-    ALOGV("%s(): mServiceStreamHandle = 0x%08X", __func__, mServiceStreamHandle);
+    ALOGD("%s(): mServiceStreamHandle = 0x%08X", __func__, mServiceStreamHandle);
     if (mServiceStreamHandle != AAUDIO_HANDLE_INVALID) {
         aaudio_stream_state_t currentState = getState();
         // Don't release a stream while it is running. Stop it first.
         // If DISCONNECTED then we should still try to stop in case the
         // error callback is still running.
         if (isActive() || currentState == AAUDIO_STREAM_STATE_DISCONNECTED) {
-            requestStop();
+            requestStop_l();
         }
 
         logReleaseBufferState();
@@ -330,7 +331,7 @@
  * The processing code will then save the current offset
  * between client and server and apply that to any position given to the app.
  */
-aaudio_result_t AudioStreamInternal::requestStart()
+aaudio_result_t AudioStreamInternal::requestStart_l()
 {
     int64_t startTime;
     if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
@@ -373,7 +374,7 @@
                               * AAUDIO_NANOS_PER_SECOND
                               / getSampleRate();
         mCallbackEnabled.store(true);
-        result = createThread(periodNanos, aaudio_callback_thread_proc, this);
+        result = createThread_l(periodNanos, aaudio_callback_thread_proc, this);
     }
     if (result != AAUDIO_OK) {
         setState(originalState);
@@ -399,26 +400,29 @@
 }
 
 // This must be called under mStreamLock.
-aaudio_result_t AudioStreamInternal::stopCallback()
+aaudio_result_t AudioStreamInternal::stopCallback_l()
 {
     if (isDataCallbackSet()
             && (isActive() || getState() == AAUDIO_STREAM_STATE_DISCONNECTED)) {
         mCallbackEnabled.store(false);
-        aaudio_result_t result = joinThread(NULL); // may temporarily unlock mStreamLock
+        aaudio_result_t result = joinThread_l(NULL); // may temporarily unlock mStreamLock
         if (result == AAUDIO_ERROR_INVALID_HANDLE) {
             ALOGD("%s() INVALID_HANDLE, stream was probably stolen", __func__);
             result = AAUDIO_OK;
         }
         return result;
     } else {
+        ALOGD("%s() skipped, isDataCallbackSet() = %d, isActive() = %d, getState()  = %d", __func__,
+            isDataCallbackSet(), isActive(), getState());
         return AAUDIO_OK;
     }
 }
 
 // This must be called under mStreamLock.
-aaudio_result_t AudioStreamInternal::requestStop() {
-    aaudio_result_t result = stopCallback();
+aaudio_result_t AudioStreamInternal::requestStop_l() {
+    aaudio_result_t result = stopCallback_l();
     if (result != AAUDIO_OK) {
+        ALOGW("%s() stop callback returned %d, returning early", __func__, result);
         return result;
     }
     // The stream may have been unlocked temporarily to let a callback finish
@@ -426,6 +430,7 @@
     // Check to make sure the stream still needs to be stopped.
     // See also AudioStream::safeStop().
     if (!(isActive() || getState() == AAUDIO_STREAM_STATE_DISCONNECTED)) {
+        ALOGD("%s() returning early, not active or disconnected", __func__);
         return AAUDIO_OK;
     }
 
@@ -805,11 +810,15 @@
     return mBufferCapacityInFrames;
 }
 
-// This must be called under mStreamLock.
 aaudio_result_t AudioStreamInternal::joinThread(void** returnArg) {
     return AudioStream::joinThread(returnArg, calculateReasonableTimeout(getFramesPerBurst()));
 }
 
+// This must be called under mStreamLock.
+aaudio_result_t AudioStreamInternal::joinThread_l(void** returnArg) {
+    return AudioStream::joinThread_l(returnArg, calculateReasonableTimeout(getFramesPerBurst()));
+}
+
 bool AudioStreamInternal::isClockModelInControl() const {
     return isActive() && mAudioEndpoint->isFreeRunning() && mClockModel.isRunning();
 }
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index d7024cf..1838b53 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -44,9 +44,9 @@
     AudioStreamInternal(AAudioServiceInterface  &serviceInterface, bool inService);
     virtual ~AudioStreamInternal();
 
-    aaudio_result_t requestStart() override;
+    aaudio_result_t requestStart_l() override;
 
-    aaudio_result_t requestStop() override;
+    aaudio_result_t requestStop_l() override;
 
     aaudio_result_t getTimestamp(clockid_t clockId,
                                        int64_t *framePosition,
@@ -117,7 +117,9 @@
 
     aaudio_result_t processCommands();
 
-    aaudio_result_t stopCallback();
+    aaudio_result_t stopCallback_l();
+
+    aaudio_result_t joinThread_l(void** returnArg);
 
     virtual void prepareBuffersForStart() {}
 
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index 980592c..b81e5e4 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -56,7 +56,7 @@
                              getDeviceChannelCount());
 
         if (result != AAUDIO_OK) {
-            releaseCloseFinal();
+            safeReleaseClose();
         }
         // Sample rate is constrained to common values by now and should not overflow.
         int32_t numFrames = kRampMSec * getSampleRate() / AAUDIO_MILLIS_PER_SECOND;
@@ -66,9 +66,9 @@
 }
 
 // This must be called under mStreamLock.
-aaudio_result_t AudioStreamInternalPlay::requestPause()
+aaudio_result_t AudioStreamInternalPlay::requestPause_l()
 {
-    aaudio_result_t result = stopCallback();
+    aaudio_result_t result = stopCallback_l();
     if (result != AAUDIO_OK) {
         return result;
     }
@@ -83,7 +83,7 @@
     return mServiceInterface.pauseStream(mServiceStreamHandle);
 }
 
-aaudio_result_t AudioStreamInternalPlay::requestFlush() {
+aaudio_result_t AudioStreamInternalPlay::requestFlush_l() {
     if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
         ALOGW("%s() mServiceStreamHandle invalid", __func__);
         return AAUDIO_ERROR_INVALID_STATE;
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.h b/media/libaaudio/src/client/AudioStreamInternalPlay.h
index 7b1cddc..03c957d 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.h
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.h
@@ -35,9 +35,9 @@
 
     aaudio_result_t open(const AudioStreamBuilder &builder) override;
 
-    aaudio_result_t requestPause() override;
+    aaudio_result_t requestPause_l() override;
 
-    aaudio_result_t requestFlush() override;
+    aaudio_result_t requestFlush_l() override;
 
     bool isFlushSupported() const override {
         // Only implement FLUSH for OUTPUT streams.
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index 310ffbe..ba86170 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -42,16 +42,20 @@
         : mPlayerBase(new MyPlayerBase())
         , mStreamId(AAudio_getNextStreamId())
         {
-    // mThread is a pthread_t of unknown size so we need memset.
-    memset(&mThread, 0, sizeof(mThread));
     setPeriodNanoseconds(0);
 }
 
 AudioStream::~AudioStream() {
-    // Please preserve this log because there have been several bugs related to
+    // Please preserve these logs because there have been several bugs related to
     // AudioStream deletion and late callbacks.
     ALOGD("%s(s#%u) mPlayerBase strongCount = %d",
             __func__, getId(), mPlayerBase->getStrongCount());
+
+    ALOGE_IF(pthread_equal(pthread_self(), mThread),
+            "%s() destructor running in callback", __func__);
+
+    ALOGE_IF(mHasThread, "%s() callback thread never join()ed", __func__);
+
     // If the stream is deleted when OPEN or in use then audio resources will leak.
     // This would indicate an internal error. So we want to find this ASAP.
     LOG_ALWAYS_FATAL_IF(!(getState() == AAUDIO_STREAM_STATE_CLOSED
@@ -164,7 +168,7 @@
             return AAUDIO_ERROR_INVALID_STATE;
     }
 
-    aaudio_result_t result = requestStart();
+    aaudio_result_t result = requestStart_l();
     if (result == AAUDIO_OK) {
         // We only call this for logging in "dumpsys audio". So ignore return code.
         (void) mPlayerBase->start();
@@ -214,7 +218,7 @@
             return AAUDIO_ERROR_INVALID_STATE;
     }
 
-    aaudio_result_t result = requestPause();
+    aaudio_result_t result = requestPause_l();
     if (result == AAUDIO_OK) {
         // We only call this for logging in "dumpsys audio". So ignore return code.
         (void) mPlayerBase->pause();
@@ -239,7 +243,7 @@
         return result;
     }
 
-    return requestFlush();
+    return requestFlush_l();
 }
 
 aaudio_result_t AudioStream::systemStopFromCallback() {
@@ -299,11 +303,11 @@
             return AAUDIO_ERROR_INVALID_STATE;
     }
 
-    return requestStop();
+    return requestStop_l();
 }
 
 aaudio_result_t AudioStream::safeRelease() {
-    // This get temporarily unlocked in the MMAP release() when joining callback threads.
+    // This may get temporarily unlocked in the MMAP release() when joining callback threads.
     std::lock_guard<std::mutex> lock(mStreamLock);
     if (collidesWithCallback()) {
         ALOGE("%s cannot be called from a callback!", __func__);
@@ -322,7 +326,14 @@
         ALOGE("%s cannot be called from a callback!", __func__);
         return AAUDIO_ERROR_INVALID_STATE;
     }
-    releaseCloseFinal();
+    releaseCloseFinal_l();
+    return AAUDIO_OK;
+}
+
+aaudio_result_t AudioStream::safeReleaseCloseFromCallback() {
+    // This get temporarily unlocked in the MMAP release() when joining callback threads.
+    std::lock_guard<std::mutex> lock(mStreamLock);
+    releaseCloseFinal_l();
     return AAUDIO_OK;
 }
 
@@ -403,23 +414,28 @@
     return procResult;
 }
 
-// This is the entry point for the new thread created by createThread().
+
+// This is the entry point for the new thread created by createThread_l().
 // It converts the 'C' function call to a C++ method call.
 static void* AudioStream_internalThreadProc(void* threadArg) {
     AudioStream *audioStream = (AudioStream *) threadArg;
-    // Use an sp<> to prevent the stream from being deleted while running.
+    // Prevent the stream from being deleted while being used.
+    // This is just for extra safety. It is probably not needed because
+    // this callback should be joined before the stream is closed.
     android::sp<AudioStream> protectedStream(audioStream);
+    // Balance the incStrong() in createThread_l().
+    protectedStream->decStrong(nullptr);
     return protectedStream->wrapUserThread();
 }
 
 // This is not exposed in the API.
 // But it is still used internally to implement callbacks for MMAP mode.
-aaudio_result_t AudioStream::createThread(int64_t periodNanoseconds,
-                                     aaudio_audio_thread_proc_t threadProc,
-                                     void* threadArg)
+aaudio_result_t AudioStream::createThread_l(int64_t periodNanoseconds,
+                                            aaudio_audio_thread_proc_t threadProc,
+                                            void* threadArg)
 {
     if (mHasThread) {
-        ALOGE("createThread() - mHasThread already true");
+        ALOGE("%s() - mHasThread already true", __func__);
         return AAUDIO_ERROR_INVALID_STATE;
     }
     if (threadProc == nullptr) {
@@ -429,10 +445,14 @@
     mThreadProc = threadProc;
     mThreadArg = threadArg;
     setPeriodNanoseconds(periodNanoseconds);
+    // Prevent this object from getting deleted before the thread has a chance to create
+    // its strong pointer. Assume the thread will call decStrong().
+    this->incStrong(nullptr);
     int err = pthread_create(&mThread, nullptr, AudioStream_internalThreadProc, this);
     if (err != 0) {
         android::status_t status = -errno;
-        ALOGE("createThread() - pthread_create() failed, %d", status);
+        ALOGE("%s() - pthread_create() failed, %d", __func__, status);
+        this->decStrong(nullptr); // Because the thread won't do it.
         return AAudioConvert_androidToAAudioResult(status);
     } else {
         // TODO Use AAudioThread or maybe AndroidThread
@@ -452,17 +472,23 @@
     }
 }
 
+aaudio_result_t AudioStream::joinThread(void** returnArg, int64_t timeoutNanoseconds) {
+    // This may get temporarily unlocked in the MMAP release() when joining callback threads.
+    std::lock_guard<std::mutex> lock(mStreamLock);
+    return joinThread_l(returnArg, timeoutNanoseconds);
+}
+
 // This must be called under mStreamLock.
-aaudio_result_t AudioStream::joinThread(void** returnArg, int64_t timeoutNanoseconds __unused)
+aaudio_result_t AudioStream::joinThread_l(void** returnArg, int64_t /* timeoutNanoseconds */)
 {
     if (!mHasThread) {
-        ALOGE("joinThread() - but has no thread");
+        ALOGD("joinThread() - but has no thread");
         return AAUDIO_ERROR_INVALID_STATE;
     }
     aaudio_result_t result = AAUDIO_OK;
     // If the callback is stopping the stream because the app passed back STOP
     // then we don't need to join(). The thread is already about to exit.
-    if (pthread_self() != mThread) {
+    if (!pthread_equal(pthread_self(), mThread)) {
         // Called from an app thread. Not the callback.
         // Unlock because the callback may be trying to stop the stream but is blocked.
         mStreamLock.unlock();
@@ -477,11 +503,15 @@
         if (err) {
             ALOGE("%s() pthread_join() returns err = %d", __func__, err);
             result = AAudioConvert_androidToAAudioResult(-err);
+        } else {
+            ALOGD("%s() pthread_join succeeded", __func__);
+            // This must be set false so that the callback thread can be created
+            // when the stream is restarted.
+            mHasThread = false;
         }
+    } else {
+        ALOGD("%s() pthread_join() called on itself!", __func__);
     }
-    // This must be set false so that the callback thread can be created
-    // when the stream is restarted.
-    mHasThread = false;
     return (result != AAUDIO_OK) ? result : mThreadRegistrationResult;
 }
 
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index e438477..d9a9d8e 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -60,7 +60,7 @@
     /* Asynchronous requests.
      * Use waitForStateChange() to wait for completion.
      */
-    virtual aaudio_result_t requestStart() = 0;
+    virtual aaudio_result_t requestStart_l() = 0;
 
     /**
      * Check the state to see if Pause is currently legal.
@@ -80,18 +80,17 @@
         return false;
     }
 
-    virtual aaudio_result_t requestPause()
-    {
+    virtual aaudio_result_t requestPause_l() {
         // Only implement this for OUTPUT streams.
         return AAUDIO_ERROR_UNIMPLEMENTED;
     }
 
-    virtual aaudio_result_t requestFlush() {
+    virtual aaudio_result_t requestFlush_l() {
         // Only implement this for OUTPUT streams.
         return AAUDIO_ERROR_UNIMPLEMENTED;
     }
 
-    virtual aaudio_result_t requestStop() = 0;
+    virtual aaudio_result_t requestStop_l() = 0;
 
 public:
     virtual aaudio_result_t getTimestamp(clockid_t clockId,
@@ -152,17 +151,6 @@
         setState(AAUDIO_STREAM_STATE_CLOSED);
     }
 
-    /**
-     * Release then close the stream.
-     */
-    void releaseCloseFinal() {
-        if (getState() != AAUDIO_STREAM_STATE_CLOSING) { // not already released?
-            // Ignore result and keep closing.
-            (void) release_l();
-        }
-        close_l();
-    }
-
     // This is only used to identify a stream in the logs without
     // revealing any pointers.
     aaudio_stream_id_t getId() {
@@ -171,9 +159,9 @@
 
     virtual aaudio_result_t setBufferSize(int32_t requestedFrames) = 0;
 
-    virtual aaudio_result_t createThread(int64_t periodNanoseconds,
-                                       aaudio_audio_thread_proc_t threadProc,
-                                       void *threadArg);
+    virtual aaudio_result_t createThread_l(int64_t periodNanoseconds,
+                                           aaudio_audio_thread_proc_t threadProc,
+                                           void *threadArg);
 
     aaudio_result_t joinThread(void **returnArg, int64_t timeoutNanoseconds);
 
@@ -432,6 +420,8 @@
      */
     aaudio_result_t safeReleaseClose();
 
+    aaudio_result_t safeReleaseCloseFromCallback();
+
 protected:
 
     // PlayerBase allows the system to control the stream volume.
@@ -543,6 +533,8 @@
         mSessionId = sessionId;
     }
 
+    aaudio_result_t joinThread_l(void **returnArg, int64_t timeoutNanoseconds);
+
     std::atomic<bool>    mCallbackEnabled{false};
 
     float                mDuckAndMuteVolume = 1.0f;
@@ -613,6 +605,17 @@
 
     aaudio_result_t safeStop();
 
+    /**
+     * Release then close the stream.
+     */
+    void releaseCloseFinal_l() {
+        if (getState() != AAUDIO_STREAM_STATE_CLOSING) { // not already released?
+            // Ignore result and keep closing.
+            (void) release_l();
+        }
+        close_l();
+    }
+
     std::mutex                 mStreamLock;
 
     const android::sp<MyPlayerBase>   mPlayerBase;
@@ -654,7 +657,7 @@
 
     // background thread ----------------------------------
     bool                        mHasThread = false;
-    pthread_t                   mThread; // initialized in constructor
+    pthread_t                   mThread = {};
 
     // These are set by the application thread and then read by the audio pthread.
     std::atomic<int64_t>        mPeriodNanoseconds; // for tuning SCHED_FIFO threads
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
index 33c1bf5..fdaa2ab 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
@@ -226,7 +226,7 @@
             ALOGD("%s() request DISCONNECT in data callback, device %d => %d",
                   __func__, (int) getDeviceId(), (int) deviceId);
             // If the stream is stopped before the data callback has a chance to handle the
-            // request then the requestStop() and requestPause() methods will handle it after
+            // request then the requestStop_l() and requestPause() methods will handle it after
             // the callback has stopped.
             mRequestDisconnect.request();
         } else {
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index d46ef56..45b2258 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
@@ -185,7 +185,7 @@
         // Did we get a valid track?
         status_t status = mAudioRecord->initCheck();
         if (status != OK) {
-            releaseCloseFinal();
+            safeReleaseClose();
             ALOGE("open(), initCheck() returned %d", status);
             return AAudioConvert_androidToAAudioResult(status);
         }
@@ -341,7 +341,7 @@
     return;
 }
 
-aaudio_result_t AudioStreamRecord::requestStart()
+aaudio_result_t AudioStreamRecord::requestStart_l()
 {
     if (mAudioRecord.get() == nullptr) {
         return AAUDIO_ERROR_INVALID_STATE;
@@ -365,7 +365,7 @@
     return AAUDIO_OK;
 }
 
-aaudio_result_t AudioStreamRecord::requestStop() {
+aaudio_result_t AudioStreamRecord::requestStop_l() {
     if (mAudioRecord.get() == nullptr) {
         return AAUDIO_ERROR_INVALID_STATE;
     }
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.h b/media/libaaudio/src/legacy/AudioStreamRecord.h
index ad8dfe4..fe9689f 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.h
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.h
@@ -41,8 +41,8 @@
     aaudio_result_t release_l() override;
     void close_l() override;
 
-    aaudio_result_t requestStart() override;
-    aaudio_result_t requestStop() override;
+    aaudio_result_t requestStart_l() override;
+    aaudio_result_t requestStop_l() override;
 
     virtual aaudio_result_t getTimestamp(clockid_t clockId,
                                          int64_t *framePosition,
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 307904e..1d036d0 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -178,7 +178,7 @@
     // Did we get a valid track?
     status_t status = mAudioTrack->initCheck();
     if (status != NO_ERROR) {
-        releaseCloseFinal();
+        safeReleaseClose();
         ALOGE("open(), initCheck() returned %d", status);
         return AAudioConvert_androidToAAudioResult(status);
     }
@@ -293,7 +293,7 @@
     return;
 }
 
-aaudio_result_t AudioStreamTrack::requestStart() {
+aaudio_result_t AudioStreamTrack::requestStart_l() {
     if (mAudioTrack.get() == nullptr) {
         ALOGE("requestStart() no AudioTrack");
         return AAUDIO_ERROR_INVALID_STATE;
@@ -320,7 +320,7 @@
     return AAUDIO_OK;
 }
 
-aaudio_result_t AudioStreamTrack::requestPause() {
+aaudio_result_t AudioStreamTrack::requestPause_l() {
     if (mAudioTrack.get() == nullptr) {
         ALOGE("%s() no AudioTrack", __func__);
         return AAUDIO_ERROR_INVALID_STATE;
@@ -336,7 +336,7 @@
     return checkForDisconnectRequest(false);
 }
 
-aaudio_result_t AudioStreamTrack::requestFlush() {
+aaudio_result_t AudioStreamTrack::requestFlush_l() {
     if (mAudioTrack.get() == nullptr) {
         ALOGE("%s() no AudioTrack", __func__);
         return AAUDIO_ERROR_INVALID_STATE;
@@ -350,7 +350,7 @@
     return AAUDIO_OK;
 }
 
-aaudio_result_t AudioStreamTrack::requestStop() {
+aaudio_result_t AudioStreamTrack::requestStop_l() {
     if (mAudioTrack.get() == nullptr) {
         ALOGE("%s() no AudioTrack", __func__);
         return AAUDIO_ERROR_INVALID_STATE;
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.h b/media/libaaudio/src/legacy/AudioStreamTrack.h
index 5a8fb39..654ea9b 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.h
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.h
@@ -44,10 +44,10 @@
     aaudio_result_t release_l() override;
     void close_l() override;
 
-    aaudio_result_t requestStart() override;
-    aaudio_result_t requestPause() override;
-    aaudio_result_t requestFlush() override;
-    aaudio_result_t requestStop() override;
+    aaudio_result_t requestStart_l() override;
+    aaudio_result_t requestPause_l() override;
+    aaudio_result_t requestFlush_l() override;
+    aaudio_result_t requestStop_l() override;
 
     bool isFlushSupported() const override {
         // Only implement FLUSH for OUTPUT streams.
diff --git a/media/libaaudio/tests/test_stop_hang.cpp b/media/libaaudio/tests/test_stop_hang.cpp
index 2397b6c..982ff4a 100644
--- a/media/libaaudio/tests/test_stop_hang.cpp
+++ b/media/libaaudio/tests/test_stop_hang.cpp
@@ -45,7 +45,7 @@
                 {
                     // Will block if the thread is running.
                     // This mutex is used to close() immediately after the callback returns
-                    // and before the requestStop() is called.
+                    // and before the requestStop_l() is called.
                     std::lock_guard<std::mutex> lock(doneLock);
                     if (done) break;
                 }
diff --git a/media/libaudioclient/AidlConversion.cpp b/media/libaudioclient/AidlConversion.cpp
new file mode 100644
index 0000000..95a6a4a
--- /dev/null
+++ b/media/libaudioclient/AidlConversion.cpp
@@ -0,0 +1,1223 @@
+/*
+ * Copyright (C) 2020 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 <limits>
+
+#define LOG_TAG "AidlConversion"
+//#define LOG_NDEBUG 0
+#include <system/audio.h>
+#include <utils/Log.h>
+
+#include "media/AidlConversion.h"
+
+#define VALUE_OR_RETURN(result)                          \
+    ({                                                   \
+        auto _tmp = (result);                            \
+        if (!_tmp.ok()) return unexpected(_tmp.error()); \
+        _tmp.value();                                    \
+    })
+
+////////////////////////////////////////////////////////////////////////////////////////////////////
+// Utilities
+
+namespace android {
+
+using base::unexpected;
+
+namespace {
+
+////////////////////////////////////////////////////////////////////////////////////////////////////
+// The code below establishes:
+// IntegralTypeOf<T>, which works for either integral types (in which case it evaluates to T), or
+// enum types (in which case it evaluates to std::underlying_type_T<T>).
+
+template<typename T, typename = std::enable_if_t<std::is_integral_v<T> || std::is_enum_v<T>>>
+struct IntegralTypeOfStruct {
+    using Type = T;
+};
+
+template<typename T>
+struct IntegralTypeOfStruct<T, std::enable_if_t<std::is_enum_v<T>>> {
+    using Type = std::underlying_type_t<T>;
+};
+
+template<typename T>
+using IntegralTypeOf = typename IntegralTypeOfStruct<T>::Type;
+
+////////////////////////////////////////////////////////////////////////////////////////////////////
+// Utilities for handling bitmasks.
+
+template<typename Enum>
+Enum index2enum_index(int index) {
+    static_assert(std::is_enum_v<Enum> || std::is_integral_v<Enum>);
+    return static_cast<Enum>(index);
+}
+
+template<typename Enum>
+Enum index2enum_bitmask(int index) {
+    static_assert(std::is_enum_v<Enum> || std::is_integral_v<Enum>);
+    return static_cast<Enum>(1 << index);
+}
+
+template<typename Mask, typename Enum>
+Mask enumToMask_bitmask(Enum e) {
+    static_assert(std::is_enum_v<Enum> || std::is_integral_v<Enum>);
+    static_assert(std::is_enum_v<Mask> || std::is_integral_v<Mask>);
+    return static_cast<Mask>(e);
+}
+
+template<typename Mask, typename Enum>
+Mask enumToMask_index(Enum e) {
+    static_assert(std::is_enum_v<Enum> || std::is_integral_v<Enum>);
+    static_assert(std::is_enum_v<Mask> || std::is_integral_v<Mask>);
+    return static_cast<Mask>(static_cast<std::make_unsigned_t<IntegralTypeOf<Mask>>>(1)
+            << static_cast<int>(e));
+}
+
+template<typename DestMask, typename SrcMask, typename DestEnum, typename SrcEnum>
+ConversionResult<DestMask> convertBitmask(
+        SrcMask src, const std::function<ConversionResult<DestEnum>(SrcEnum)>& enumConversion,
+        const std::function<SrcEnum(int)>& srcIndexToEnum,
+        const std::function<DestMask(DestEnum)>& destEnumToMask) {
+    using UnsignedDestMask = std::make_unsigned_t<IntegralTypeOf<DestMask>>;
+    using UnsignedSrcMask = std::make_unsigned_t<IntegralTypeOf<SrcMask>>;
+
+    UnsignedDestMask dest = static_cast<UnsignedDestMask>(0);
+    UnsignedSrcMask usrc = static_cast<UnsignedSrcMask>(src);
+
+    int srcBitIndex = 0;
+    while (usrc != 0) {
+        if (usrc & 1) {
+            SrcEnum srcEnum = srcIndexToEnum(srcBitIndex);
+            DestEnum destEnum = VALUE_OR_RETURN(enumConversion(srcEnum));
+            DestMask destMask = destEnumToMask(destEnum);
+            dest |= destMask;
+        }
+        ++srcBitIndex;
+        usrc >>= 1;
+    }
+    return static_cast<DestMask>(dest);
+}
+
+template<typename Mask, typename Enum>
+bool bitmaskIsSet(Mask mask, Enum index) {
+    return (mask & enumToMask_index<Mask, Enum>(index)) != 0;
+}
+
+////////////////////////////////////////////////////////////////////////////////////////////////////
+
+template<typename To, typename From>
+ConversionResult<To> convertIntegral(From from) {
+    // Special handling is required for signed / vs. unsigned comparisons, since otherwise we may
+    // have the signed converted to unsigned and produce wrong results.
+    if (std::is_signed_v<From> && !std::is_signed_v<To>) {
+        if (from < 0 || from > std::numeric_limits<To>::max()) {
+            return unexpected(BAD_VALUE);
+        }
+    } else if (std::is_signed_v<To> && !std::is_signed_v<From>) {
+        if (from > std::numeric_limits<To>::max()) {
+            return unexpected(BAD_VALUE);
+        }
+    } else {
+        if (from < std::numeric_limits<To>::min() || from > std::numeric_limits<To>::max()) {
+            return unexpected(BAD_VALUE);
+        }
+    }
+    return static_cast<To>(from);
+}
+
+template<typename To, typename From>
+ConversionResult<To> convertReinterpret(From from) {
+    static_assert(sizeof(From) == sizeof(To));
+    return static_cast<To>(from);
+}
+
+enum class Direction {
+    INPUT, OUTPUT
+};
+
+ConversionResult<Direction> direction(media::AudioPortRole role, media::AudioPortType type) {
+    switch (type) {
+        case media::AudioPortType::DEVICE:
+            switch (role) {
+                case media::AudioPortRole::SOURCE:
+                    return Direction::INPUT;
+                case media::AudioPortRole::SINK:
+                    return Direction::OUTPUT;
+                default:
+                    break;
+            }
+            break;
+        case media::AudioPortType::MIX:
+            switch (role) {
+                case media::AudioPortRole::SOURCE:
+                    return Direction::OUTPUT;
+                case media::AudioPortRole::SINK:
+                    return Direction::INPUT;
+                default:
+                    break;
+            }
+            break;
+        default:
+            break;
+    }
+    return unexpected(BAD_VALUE);
+}
+
+ConversionResult<Direction> direction(audio_port_role_t role, audio_port_type_t type) {
+    switch (type) {
+        case AUDIO_PORT_TYPE_DEVICE:
+            switch (role) {
+                case AUDIO_PORT_ROLE_SOURCE:
+                    return Direction::INPUT;
+                case AUDIO_PORT_ROLE_SINK:
+                    return Direction::OUTPUT;
+                default:
+                    break;
+            }
+            break;
+        case AUDIO_PORT_TYPE_MIX:
+            switch (role) {
+                case AUDIO_PORT_ROLE_SOURCE:
+                    return Direction::OUTPUT;
+                case AUDIO_PORT_ROLE_SINK:
+                    return Direction::INPUT;
+                default:
+                    break;
+            }
+            break;
+        default:
+            break;
+    }
+    return unexpected(BAD_VALUE);
+}
+
+}  // namespace
+
+////////////////////////////////////////////////////////////////////////////////////////////////////
+// Converters
+
+// The legacy enum is unnamed. Thus, we use int.
+ConversionResult<int> aidl2legacy_AudioPortConfigType(media::AudioPortConfigType aidl) {
+    switch (aidl) {
+        case media::AudioPortConfigType::SAMPLE_RATE:
+            return AUDIO_PORT_CONFIG_SAMPLE_RATE;
+        case media::AudioPortConfigType::CHANNEL_MASK:
+            return AUDIO_PORT_CONFIG_CHANNEL_MASK;
+        case media::AudioPortConfigType::FORMAT:
+            return AUDIO_PORT_CONFIG_FORMAT;
+        case media::AudioPortConfigType::FLAGS:
+            return AUDIO_PORT_CONFIG_FLAGS;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+// The legacy enum is unnamed. Thus, we use int.
+ConversionResult<media::AudioPortConfigType> legacy2aidl_AudioPortConfigType(int legacy) {
+    switch (legacy) {
+        case AUDIO_PORT_CONFIG_SAMPLE_RATE:
+            return media::AudioPortConfigType::SAMPLE_RATE;
+        case AUDIO_PORT_CONFIG_CHANNEL_MASK:
+            return media::AudioPortConfigType::CHANNEL_MASK;
+        case AUDIO_PORT_CONFIG_FORMAT:
+            return media::AudioPortConfigType::FORMAT;
+        case AUDIO_PORT_CONFIG_FLAGS:
+            return media::AudioPortConfigType::FLAGS;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<unsigned int> aidl2legacy_int32_t_config_mask(int32_t aidl) {
+    return convertBitmask<unsigned int, int32_t, int, media::AudioPortConfigType>(
+            aidl, aidl2legacy_AudioPortConfigType,
+            // AudioPortConfigType enum is index-based.
+            index2enum_index<media::AudioPortConfigType>,
+            // AUDIO_PORT_CONFIG_* flags are mask-based.
+            enumToMask_bitmask<unsigned int, int>);
+}
+
+ConversionResult<int32_t> legacy2aidl_config_mask_int32_t(unsigned int legacy) {
+    return convertBitmask<int32_t, unsigned int, media::AudioPortConfigType, int>(
+            legacy, legacy2aidl_AudioPortConfigType,
+            // AUDIO_PORT_CONFIG_* flags are mask-based.
+            index2enum_bitmask<unsigned>,
+            // AudioPortConfigType enum is index-based.
+            enumToMask_index<int32_t, media::AudioPortConfigType>);
+}
+
+ConversionResult<audio_channel_mask_t> aidl2legacy_int32_t_audio_channel_mask_t(int32_t aidl) {
+    // TODO(ytai): should we convert bit-by-bit?
+    // One problem here is that the representation is both opaque and is different based on the
+    // context (input vs. output). Can determine based on type and role, as per useInChannelMask().
+    return convertReinterpret<audio_channel_mask_t>(aidl);
+}
+
+ConversionResult<int32_t> legacy2aidl_audio_channel_mask_t_int32_t(audio_channel_mask_t legacy) {
+    // TODO(ytai): should we convert bit-by-bit?
+    // One problem here is that the representation is both opaque and is different based on the
+    // context (input vs. output). Can determine based on type and role, as per useInChannelMask().
+    return convertReinterpret<int32_t>(legacy);
+}
+
+ConversionResult<audio_io_config_event> aidl2legacy_AudioIoConfigEvent_audio_io_config_event(
+        media::AudioIoConfigEvent aidl) {
+    switch (aidl) {
+        case media::AudioIoConfigEvent::OUTPUT_REGISTERED:
+            return AUDIO_OUTPUT_REGISTERED;
+        case media::AudioIoConfigEvent::OUTPUT_OPENED:
+            return AUDIO_OUTPUT_OPENED;
+        case media::AudioIoConfigEvent::OUTPUT_CLOSED:
+            return AUDIO_OUTPUT_CLOSED;
+        case media::AudioIoConfigEvent::OUTPUT_CONFIG_CHANGED:
+            return AUDIO_OUTPUT_CONFIG_CHANGED;
+        case media::AudioIoConfigEvent::INPUT_REGISTERED:
+            return AUDIO_INPUT_REGISTERED;
+        case media::AudioIoConfigEvent::INPUT_OPENED:
+            return AUDIO_INPUT_OPENED;
+        case media::AudioIoConfigEvent::INPUT_CLOSED:
+            return AUDIO_INPUT_CLOSED;
+        case media::AudioIoConfigEvent::INPUT_CONFIG_CHANGED:
+            return AUDIO_INPUT_CONFIG_CHANGED;
+        case media::AudioIoConfigEvent::CLIENT_STARTED:
+            return AUDIO_CLIENT_STARTED;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<media::AudioIoConfigEvent> legacy2aidl_audio_io_config_event_AudioIoConfigEvent(
+        audio_io_config_event legacy) {
+    switch (legacy) {
+        case AUDIO_OUTPUT_REGISTERED:
+            return media::AudioIoConfigEvent::OUTPUT_REGISTERED;
+        case AUDIO_OUTPUT_OPENED:
+            return media::AudioIoConfigEvent::OUTPUT_OPENED;
+        case AUDIO_OUTPUT_CLOSED:
+            return media::AudioIoConfigEvent::OUTPUT_CLOSED;
+        case AUDIO_OUTPUT_CONFIG_CHANGED:
+            return media::AudioIoConfigEvent::OUTPUT_CONFIG_CHANGED;
+        case AUDIO_INPUT_REGISTERED:
+            return media::AudioIoConfigEvent::INPUT_REGISTERED;
+        case AUDIO_INPUT_OPENED:
+            return media::AudioIoConfigEvent::INPUT_OPENED;
+        case AUDIO_INPUT_CLOSED:
+            return media::AudioIoConfigEvent::INPUT_CLOSED;
+        case AUDIO_INPUT_CONFIG_CHANGED:
+            return media::AudioIoConfigEvent::INPUT_CONFIG_CHANGED;
+        case AUDIO_CLIENT_STARTED:
+            return media::AudioIoConfigEvent::CLIENT_STARTED;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<audio_port_role_t> aidl2legacy_AudioPortRole_audio_port_role_t(
+        media::AudioPortRole aidl) {
+    switch (aidl) {
+        case media::AudioPortRole::NONE:
+            return AUDIO_PORT_ROLE_NONE;
+        case media::AudioPortRole::SOURCE:
+            return AUDIO_PORT_ROLE_SOURCE;
+        case media::AudioPortRole::SINK:
+            return AUDIO_PORT_ROLE_SINK;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<media::AudioPortRole> legacy2aidl_audio_port_role_t_AudioPortRole(
+        audio_port_role_t legacy) {
+    switch (legacy) {
+        case AUDIO_PORT_ROLE_NONE:
+            return media::AudioPortRole::NONE;
+        case AUDIO_PORT_ROLE_SOURCE:
+            return media::AudioPortRole::SOURCE;
+        case AUDIO_PORT_ROLE_SINK:
+            return media::AudioPortRole::SINK;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<audio_port_type_t> aidl2legacy_AudioPortType_audio_port_type_t(
+        media::AudioPortType aidl) {
+    switch (aidl) {
+        case media::AudioPortType::NONE:
+            return AUDIO_PORT_TYPE_NONE;
+        case media::AudioPortType::DEVICE:
+            return AUDIO_PORT_TYPE_DEVICE;
+        case media::AudioPortType::MIX:
+            return AUDIO_PORT_TYPE_MIX;
+        case media::AudioPortType::SESSION:
+            return AUDIO_PORT_TYPE_SESSION;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<media::AudioPortType> legacy2aidl_audio_port_type_t_AudioPortType(
+        audio_port_type_t legacy) {
+    switch (legacy) {
+        case AUDIO_PORT_TYPE_NONE:
+            return media::AudioPortType::NONE;
+        case AUDIO_PORT_TYPE_DEVICE:
+            return media::AudioPortType::DEVICE;
+        case AUDIO_PORT_TYPE_MIX:
+            return media::AudioPortType::MIX;
+        case AUDIO_PORT_TYPE_SESSION:
+            return media::AudioPortType::SESSION;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<audio_format_t> aidl2legacy_AudioFormat_audio_format_t(
+        media::audio::common::AudioFormat aidl) {
+    // This relies on AudioFormat being kept in sync with audio_format_t.
+    static_assert(sizeof(media::audio::common::AudioFormat) == sizeof(audio_format_t));
+    return static_cast<audio_format_t>(aidl);
+}
+
+ConversionResult<media::audio::common::AudioFormat> legacy2aidl_audio_format_t_AudioFormat(
+        audio_format_t legacy) {
+    // This relies on AudioFormat being kept in sync with audio_format_t.
+    static_assert(sizeof(media::audio::common::AudioFormat) == sizeof(audio_format_t));
+    return static_cast<media::audio::common::AudioFormat>(legacy);
+}
+
+ConversionResult<int> aidl2legacy_AudioGainMode_int(media::AudioGainMode aidl) {
+    switch (aidl) {
+        case media::AudioGainMode::JOINT:
+            return AUDIO_GAIN_MODE_JOINT;
+        case media::AudioGainMode::CHANNELS:
+            return AUDIO_GAIN_MODE_CHANNELS;
+        case media::AudioGainMode::RAMP:
+            return AUDIO_GAIN_MODE_RAMP;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<media::AudioGainMode> legacy2aidl_int_AudioGainMode(int legacy) {
+    switch (legacy) {
+        case AUDIO_GAIN_MODE_JOINT:
+            return media::AudioGainMode::JOINT;
+        case AUDIO_GAIN_MODE_CHANNELS:
+            return media::AudioGainMode::CHANNELS;
+        case AUDIO_GAIN_MODE_RAMP:
+            return media::AudioGainMode::RAMP;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<audio_gain_mode_t> aidl2legacy_int32_t_audio_gain_mode_t(int32_t aidl) {
+    return convertBitmask<audio_gain_mode_t, int32_t, int, media::AudioGainMode>(
+            aidl, aidl2legacy_AudioGainMode_int,
+            // AudioGainMode is index-based.
+            index2enum_index<media::AudioGainMode>,
+            // AUDIO_GAIN_MODE_* constants are mask-based.
+            enumToMask_bitmask<audio_gain_mode_t, int>);
+}
+
+ConversionResult<int32_t> legacy2aidl_audio_gain_mode_t_int32_t(audio_gain_mode_t legacy) {
+    return convertBitmask<int32_t, audio_gain_mode_t, media::AudioGainMode, int>(
+            legacy, legacy2aidl_int_AudioGainMode,
+            // AUDIO_GAIN_MODE_* constants are mask-based.
+            index2enum_bitmask<int>,
+            // AudioGainMode is index-based.
+            enumToMask_index<int32_t, media::AudioGainMode>);
+}
+
+ConversionResult<audio_devices_t> aidl2legacy_int32_t_audio_devices_t(int32_t aidl) {
+    // TODO(ytai): bitfield?
+    return convertReinterpret<audio_devices_t>(aidl);
+}
+
+ConversionResult<int32_t> legacy2aidl_audio_devices_t_int32_t(audio_devices_t legacy) {
+    // TODO(ytai): bitfield?
+    return convertReinterpret<int32_t>(legacy);
+}
+
+ConversionResult<audio_gain_config> aidl2legacy_AudioGainConfig_audio_gain_config(
+        const media::AudioGainConfig& aidl, media::AudioPortRole role, media::AudioPortType type) {
+    audio_gain_config legacy;
+    legacy.index = VALUE_OR_RETURN(convertIntegral<int>(aidl.index));
+    legacy.mode = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_gain_mode_t(aidl.mode));
+    legacy.channel_mask =
+            VALUE_OR_RETURN(aidl2legacy_int32_t_audio_channel_mask_t(aidl.channelMask));
+    const bool isInput = VALUE_OR_RETURN(direction(role, type)) == Direction::INPUT;
+    const bool isJoint = bitmaskIsSet(aidl.mode, media::AudioGainMode::JOINT);
+    size_t numValues = isJoint ? 1
+                               : isInput ? audio_channel_count_from_in_mask(legacy.channel_mask)
+                                         : audio_channel_count_from_out_mask(legacy.channel_mask);
+    if (aidl.values.size() != numValues || aidl.values.size() > std::size(legacy.values)) {
+        return unexpected(BAD_VALUE);
+    }
+    for (size_t i = 0; i < numValues; ++i) {
+        legacy.values[i] = VALUE_OR_RETURN(convertIntegral<int>(aidl.values[i]));
+    }
+    legacy.ramp_duration_ms = VALUE_OR_RETURN(convertIntegral<unsigned int>(aidl.rampDurationMs));
+    return legacy;
+}
+
+ConversionResult<media::AudioGainConfig> legacy2aidl_audio_gain_config_AudioGainConfig(
+        const audio_gain_config& legacy, audio_port_role_t role, audio_port_type_t type) {
+    media::AudioGainConfig aidl;
+    aidl.index = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.index));
+    aidl.mode = VALUE_OR_RETURN(legacy2aidl_audio_gain_mode_t_int32_t(legacy.mode));
+    aidl.channelMask =
+            VALUE_OR_RETURN(legacy2aidl_audio_channel_mask_t_int32_t(legacy.channel_mask));
+    const bool isInput = VALUE_OR_RETURN(direction(role, type)) == Direction::INPUT;
+    const bool isJoint = (legacy.mode & AUDIO_GAIN_MODE_JOINT) != 0;
+    size_t numValues = isJoint ? 1
+                               : isInput ? audio_channel_count_from_in_mask(legacy.channel_mask)
+                                         : audio_channel_count_from_out_mask(legacy.channel_mask);
+    aidl.values.resize(numValues);
+    for (size_t i = 0; i < numValues; ++i) {
+        aidl.values[i] = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.values[i]));
+    }
+    aidl.rampDurationMs = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.ramp_duration_ms));
+    return aidl;
+}
+
+ConversionResult<audio_input_flags_t> aidl2legacy_AudioInputFlags_audio_input_flags_t(
+        media::AudioInputFlags aidl) {
+    switch (aidl) {
+        case media::AudioInputFlags::FAST:
+            return AUDIO_INPUT_FLAG_FAST;
+        case media::AudioInputFlags::HW_HOTWORD:
+            return AUDIO_INPUT_FLAG_HW_HOTWORD;
+        case media::AudioInputFlags::RAW:
+            return AUDIO_INPUT_FLAG_RAW;
+        case media::AudioInputFlags::SYNC:
+            return AUDIO_INPUT_FLAG_SYNC;
+        case media::AudioInputFlags::MMAP_NOIRQ:
+            return AUDIO_INPUT_FLAG_MMAP_NOIRQ;
+        case media::AudioInputFlags::VOIP_TX:
+            return AUDIO_INPUT_FLAG_VOIP_TX;
+        case media::AudioInputFlags::HW_AV_SYNC:
+            return AUDIO_INPUT_FLAG_HW_AV_SYNC;
+        case media::AudioInputFlags::DIRECT:
+            return AUDIO_INPUT_FLAG_DIRECT;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<media::AudioInputFlags> legacy2aidl_audio_input_flags_t_AudioInputFlags(
+        audio_input_flags_t legacy) {
+    switch (legacy) {
+        case AUDIO_INPUT_FLAG_FAST:
+            return media::AudioInputFlags::FAST;
+        case AUDIO_INPUT_FLAG_HW_HOTWORD:
+            return media::AudioInputFlags::HW_HOTWORD;
+        case AUDIO_INPUT_FLAG_RAW:
+            return media::AudioInputFlags::RAW;
+        case AUDIO_INPUT_FLAG_SYNC:
+            return media::AudioInputFlags::SYNC;
+        case AUDIO_INPUT_FLAG_MMAP_NOIRQ:
+            return media::AudioInputFlags::MMAP_NOIRQ;
+        case AUDIO_INPUT_FLAG_VOIP_TX:
+            return media::AudioInputFlags::VOIP_TX;
+        case AUDIO_INPUT_FLAG_HW_AV_SYNC:
+            return media::AudioInputFlags::HW_AV_SYNC;
+        case AUDIO_INPUT_FLAG_DIRECT:
+            return media::AudioInputFlags::DIRECT;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<audio_output_flags_t> aidl2legacy_AudioOutputFlags_audio_output_flags_t(
+        media::AudioOutputFlags aidl) {
+    switch (aidl) {
+        case media::AudioOutputFlags::DIRECT:
+            return AUDIO_OUTPUT_FLAG_DIRECT;
+        case media::AudioOutputFlags::PRIMARY:
+            return AUDIO_OUTPUT_FLAG_PRIMARY;
+        case media::AudioOutputFlags::FAST:
+            return AUDIO_OUTPUT_FLAG_FAST;
+        case media::AudioOutputFlags::DEEP_BUFFER:
+            return AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
+        case media::AudioOutputFlags::COMPRESS_OFFLOAD:
+            return AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD;
+        case media::AudioOutputFlags::NON_BLOCKING:
+            return AUDIO_OUTPUT_FLAG_NON_BLOCKING;
+        case media::AudioOutputFlags::HW_AV_SYNC:
+            return AUDIO_OUTPUT_FLAG_HW_AV_SYNC;
+        case media::AudioOutputFlags::TTS:
+            return AUDIO_OUTPUT_FLAG_TTS;
+        case media::AudioOutputFlags::RAW:
+            return AUDIO_OUTPUT_FLAG_RAW;
+        case media::AudioOutputFlags::SYNC:
+            return AUDIO_OUTPUT_FLAG_SYNC;
+        case media::AudioOutputFlags::IEC958_NONAUDIO:
+            return AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO;
+        case media::AudioOutputFlags::DIRECT_PCM:
+            return AUDIO_OUTPUT_FLAG_DIRECT_PCM;
+        case media::AudioOutputFlags::MMAP_NOIRQ:
+            return AUDIO_OUTPUT_FLAG_MMAP_NOIRQ;
+        case media::AudioOutputFlags::VOIP_RX:
+            return AUDIO_OUTPUT_FLAG_VOIP_RX;
+        case media::AudioOutputFlags::INCALL_MUSIC:
+            return AUDIO_OUTPUT_FLAG_INCALL_MUSIC;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<media::AudioOutputFlags> legacy2aidl_audio_output_flags_t_AudioOutputFlags(
+        audio_output_flags_t legacy) {
+    switch (legacy) {
+        case AUDIO_OUTPUT_FLAG_DIRECT:
+            return media::AudioOutputFlags::DIRECT;
+        case AUDIO_OUTPUT_FLAG_PRIMARY:
+            return media::AudioOutputFlags::PRIMARY;
+        case AUDIO_OUTPUT_FLAG_FAST:
+            return media::AudioOutputFlags::FAST;
+        case AUDIO_OUTPUT_FLAG_DEEP_BUFFER:
+            return media::AudioOutputFlags::DEEP_BUFFER;
+        case AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD:
+            return media::AudioOutputFlags::COMPRESS_OFFLOAD;
+        case AUDIO_OUTPUT_FLAG_NON_BLOCKING:
+            return media::AudioOutputFlags::NON_BLOCKING;
+        case AUDIO_OUTPUT_FLAG_HW_AV_SYNC:
+            return media::AudioOutputFlags::HW_AV_SYNC;
+        case AUDIO_OUTPUT_FLAG_TTS:
+            return media::AudioOutputFlags::TTS;
+        case AUDIO_OUTPUT_FLAG_RAW:
+            return media::AudioOutputFlags::RAW;
+        case AUDIO_OUTPUT_FLAG_SYNC:
+            return media::AudioOutputFlags::SYNC;
+        case AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO:
+            return media::AudioOutputFlags::IEC958_NONAUDIO;
+        case AUDIO_OUTPUT_FLAG_DIRECT_PCM:
+            return media::AudioOutputFlags::DIRECT_PCM;
+        case AUDIO_OUTPUT_FLAG_MMAP_NOIRQ:
+            return media::AudioOutputFlags::MMAP_NOIRQ;
+        case AUDIO_OUTPUT_FLAG_VOIP_RX:
+            return media::AudioOutputFlags::VOIP_RX;
+        case AUDIO_OUTPUT_FLAG_INCALL_MUSIC:
+            return media::AudioOutputFlags::INCALL_MUSIC;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<audio_input_flags_t> aidl2legacy_audio_input_flags_mask(int32_t aidl) {
+    using LegacyMask = std::underlying_type_t<audio_input_flags_t>;
+
+    LegacyMask converted = VALUE_OR_RETURN(
+            (convertBitmask<LegacyMask, int32_t, audio_input_flags_t, media::AudioInputFlags>(
+                    aidl, aidl2legacy_AudioInputFlags_audio_input_flags_t,
+                    index2enum_index<media::AudioInputFlags>,
+                    enumToMask_bitmask<LegacyMask, audio_input_flags_t>)));
+    return static_cast<audio_input_flags_t>(converted);
+}
+
+ConversionResult<int32_t> legacy2aidl_audio_input_flags_mask(audio_input_flags_t legacy) {
+    using LegacyMask = std::underlying_type_t<audio_input_flags_t>;
+
+    LegacyMask legacyMask = static_cast<LegacyMask>(legacy);
+    return convertBitmask<int32_t, LegacyMask, media::AudioInputFlags, audio_input_flags_t>(
+            legacyMask, legacy2aidl_audio_input_flags_t_AudioInputFlags,
+            index2enum_bitmask<audio_input_flags_t>,
+            enumToMask_index<int32_t, media::AudioInputFlags>);
+}
+
+ConversionResult<audio_output_flags_t> aidl2legacy_audio_output_flags_mask(int32_t aidl) {
+    using LegacyMask = std::underlying_type_t<audio_output_flags_t>;
+
+    LegacyMask converted = VALUE_OR_RETURN(
+            (convertBitmask<LegacyMask, int32_t, audio_output_flags_t, media::AudioOutputFlags>(
+                    aidl, aidl2legacy_AudioOutputFlags_audio_output_flags_t,
+                    index2enum_index<media::AudioOutputFlags>,
+                    enumToMask_bitmask<LegacyMask, audio_output_flags_t>)));
+    return convertReinterpret<audio_output_flags_t>(converted);
+}
+
+ConversionResult<int32_t> legacy2aidl_audio_output_flags_mask(audio_output_flags_t legacy) {
+    using LegacyMask = std::underlying_type_t<audio_output_flags_t>;
+
+    LegacyMask legacyMask = static_cast<LegacyMask>(legacy);
+    return convertBitmask<int32_t, LegacyMask, media::AudioOutputFlags, audio_output_flags_t>(
+            legacyMask, legacy2aidl_audio_output_flags_t_AudioOutputFlags,
+            index2enum_bitmask<audio_output_flags_t>,
+            enumToMask_index<int32_t, media::AudioOutputFlags>);
+}
+
+ConversionResult<audio_io_flags> aidl2legacy_AudioIoFlags_audio_io_flags(
+        const media::AudioIoFlags& aidl, media::AudioPortRole role, media::AudioPortType type) {
+    audio_io_flags legacy;
+    // Our way of representing a union in AIDL is to have multiple vectors and require that at most
+    // one of the them has size 1 and the rest are empty.
+    size_t totalSize = aidl.input.size() + aidl.output.size();
+    if (totalSize > 1) {
+        return unexpected(BAD_VALUE);
+    }
+
+    Direction dir = VALUE_OR_RETURN(direction(role, type));
+    switch (dir) {
+        case Direction::INPUT:
+            if (aidl.input.empty()) {
+                return unexpected(BAD_VALUE);
+            }
+            legacy.input = VALUE_OR_RETURN(aidl2legacy_audio_input_flags_mask(aidl.input[0]));
+            break;
+
+        case Direction::OUTPUT:
+            if (aidl.output.empty()) {
+                return unexpected(BAD_VALUE);
+            }
+            legacy.output = VALUE_OR_RETURN(aidl2legacy_audio_output_flags_mask(aidl.output[0]));
+            break;
+    }
+
+    return legacy;
+}
+
+ConversionResult<media::AudioIoFlags> legacy2aidl_audio_io_flags_AudioIoFlags(
+        const audio_io_flags& legacy, audio_port_role_t role, audio_port_type_t type) {
+    media::AudioIoFlags aidl;
+
+    Direction dir = VALUE_OR_RETURN(direction(role, type));
+    switch (dir) {
+        case Direction::INPUT:
+            aidl.input.push_back(VALUE_OR_RETURN(legacy2aidl_audio_input_flags_mask(legacy.input)));
+            break;
+        case Direction::OUTPUT:
+            aidl.output.push_back(
+                    VALUE_OR_RETURN(legacy2aidl_audio_output_flags_mask(legacy.output)));
+            break;
+    }
+    return aidl;
+}
+
+ConversionResult<audio_port_config_device_ext> aidl2legacy_AudioPortConfigDeviceExt(
+        const media::AudioPortConfigDeviceExt& aidl) {
+    audio_port_config_device_ext legacy;
+    legacy.hw_module = VALUE_OR_RETURN(convertReinterpret<audio_module_handle_t>(aidl.hwModule));
+    legacy.type = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_devices_t(aidl.type));
+    if (aidl.address.size() > AUDIO_DEVICE_MAX_ADDRESS_LEN - 1) {
+        return unexpected(BAD_VALUE);
+    }
+    std::strcpy(legacy.address, aidl.address.c_str());
+    return legacy;
+}
+
+ConversionResult<media::AudioPortConfigDeviceExt> legacy2aidl_AudioPortConfigDeviceExt(
+        const audio_port_config_device_ext& legacy) {
+    media::AudioPortConfigDeviceExt aidl;
+    aidl.hwModule = VALUE_OR_RETURN(convertReinterpret<int32_t>(legacy.hw_module));
+    aidl.type = VALUE_OR_RETURN(legacy2aidl_audio_devices_t_int32_t(legacy.type));
+
+    if (strnlen(legacy.address, AUDIO_DEVICE_MAX_ADDRESS_LEN) == AUDIO_DEVICE_MAX_ADDRESS_LEN) {
+        // No null-terminator.
+        return unexpected(BAD_VALUE);
+    }
+    aidl.address = legacy.address;
+    return aidl;
+}
+
+ConversionResult<audio_stream_type_t> aidl2legacy_AudioStreamType_audio_stream_type_t(
+        media::AudioStreamType aidl) {
+    switch (aidl) {
+        case media::AudioStreamType::DEFAULT:
+            return AUDIO_STREAM_DEFAULT;
+        case media::AudioStreamType::VOICE_CALL:
+            return AUDIO_STREAM_VOICE_CALL;
+        case media::AudioStreamType::SYSTEM:
+            return AUDIO_STREAM_SYSTEM;
+        case media::AudioStreamType::RING:
+            return AUDIO_STREAM_RING;
+        case media::AudioStreamType::MUSIC:
+            return AUDIO_STREAM_MUSIC;
+        case media::AudioStreamType::ALARM:
+            return AUDIO_STREAM_ALARM;
+        case media::AudioStreamType::NOTIFICATION:
+            return AUDIO_STREAM_NOTIFICATION;
+        case media::AudioStreamType::BLUETOOTH_SCO:
+            return AUDIO_STREAM_BLUETOOTH_SCO;
+        case media::AudioStreamType::ENFORCED_AUDIBLE:
+            return AUDIO_STREAM_ENFORCED_AUDIBLE;
+        case media::AudioStreamType::DTMF:
+            return AUDIO_STREAM_DTMF;
+        case media::AudioStreamType::TTS:
+            return AUDIO_STREAM_TTS;
+        case media::AudioStreamType::ACCESSIBILITY:
+            return AUDIO_STREAM_ACCESSIBILITY;
+        case media::AudioStreamType::ASSISTANT:
+            return AUDIO_STREAM_ASSISTANT;
+        case media::AudioStreamType::REROUTING:
+            return AUDIO_STREAM_REROUTING;
+        case media::AudioStreamType::PATCH:
+            return AUDIO_STREAM_PATCH;
+        case media::AudioStreamType::CALL_ASSISTANT:
+            return AUDIO_STREAM_CALL_ASSISTANT;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<media::AudioStreamType> legacy2aidl_audio_stream_type_t_AudioStreamType(
+        audio_stream_type_t legacy) {
+    switch (legacy) {
+        case AUDIO_STREAM_DEFAULT:
+            return media::AudioStreamType::DEFAULT;
+        case AUDIO_STREAM_VOICE_CALL:
+            return media::AudioStreamType::VOICE_CALL;
+        case AUDIO_STREAM_SYSTEM:
+            return media::AudioStreamType::SYSTEM;
+        case AUDIO_STREAM_RING:
+            return media::AudioStreamType::RING;
+        case AUDIO_STREAM_MUSIC:
+            return media::AudioStreamType::MUSIC;
+        case AUDIO_STREAM_ALARM:
+            return media::AudioStreamType::ALARM;
+        case AUDIO_STREAM_NOTIFICATION:
+            return media::AudioStreamType::NOTIFICATION;
+        case AUDIO_STREAM_BLUETOOTH_SCO:
+            return media::AudioStreamType::BLUETOOTH_SCO;
+        case AUDIO_STREAM_ENFORCED_AUDIBLE:
+            return media::AudioStreamType::ENFORCED_AUDIBLE;
+        case AUDIO_STREAM_DTMF:
+            return media::AudioStreamType::DTMF;
+        case AUDIO_STREAM_TTS:
+            return media::AudioStreamType::TTS;
+        case AUDIO_STREAM_ACCESSIBILITY:
+            return media::AudioStreamType::ACCESSIBILITY;
+        case AUDIO_STREAM_ASSISTANT:
+            return media::AudioStreamType::ASSISTANT;
+        case AUDIO_STREAM_REROUTING:
+            return media::AudioStreamType::REROUTING;
+        case AUDIO_STREAM_PATCH:
+            return media::AudioStreamType::PATCH;
+        case AUDIO_STREAM_CALL_ASSISTANT:
+            return media::AudioStreamType::CALL_ASSISTANT;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<audio_source_t> aidl2legacy_AudioSourceType_audio_source_t(
+        media::AudioSourceType aidl) {
+    switch (aidl) {
+        case media::AudioSourceType::DEFAULT:
+            return AUDIO_SOURCE_DEFAULT;
+        case media::AudioSourceType::MIC:
+            return AUDIO_SOURCE_MIC;
+        case media::AudioSourceType::VOICE_UPLINK:
+            return AUDIO_SOURCE_VOICE_UPLINK;
+        case media::AudioSourceType::VOICE_DOWNLINK:
+            return AUDIO_SOURCE_VOICE_DOWNLINK;
+        case media::AudioSourceType::VOICE_CALL:
+            return AUDIO_SOURCE_VOICE_CALL;
+        case media::AudioSourceType::CAMCORDER:
+            return AUDIO_SOURCE_CAMCORDER;
+        case media::AudioSourceType::VOICE_RECOGNITION:
+            return AUDIO_SOURCE_VOICE_RECOGNITION;
+        case media::AudioSourceType::VOICE_COMMUNICATION:
+            return AUDIO_SOURCE_VOICE_COMMUNICATION;
+        case media::AudioSourceType::REMOTE_SUBMIX:
+            return AUDIO_SOURCE_REMOTE_SUBMIX;
+        case media::AudioSourceType::UNPROCESSED:
+            return AUDIO_SOURCE_UNPROCESSED;
+        case media::AudioSourceType::VOICE_PERFORMANCE:
+            return AUDIO_SOURCE_VOICE_PERFORMANCE;
+        case media::AudioSourceType::ECHO_REFERENCE:
+            return AUDIO_SOURCE_ECHO_REFERENCE;
+        case media::AudioSourceType::FM_TUNER:
+            return AUDIO_SOURCE_FM_TUNER;
+        case media::AudioSourceType::HOTWORD:
+            return AUDIO_SOURCE_HOTWORD;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<media::AudioSourceType> legacy2aidl_audio_source_t_AudioSourceType(
+        audio_source_t legacy) {
+    switch (legacy) {
+        case AUDIO_SOURCE_DEFAULT:
+            return media::AudioSourceType::DEFAULT;
+        case AUDIO_SOURCE_MIC:
+            return media::AudioSourceType::MIC;
+        case AUDIO_SOURCE_VOICE_UPLINK:
+            return media::AudioSourceType::VOICE_UPLINK;
+        case AUDIO_SOURCE_VOICE_DOWNLINK:
+            return media::AudioSourceType::VOICE_DOWNLINK;
+        case AUDIO_SOURCE_VOICE_CALL:
+            return media::AudioSourceType::VOICE_CALL;
+        case AUDIO_SOURCE_CAMCORDER:
+            return media::AudioSourceType::CAMCORDER;
+        case AUDIO_SOURCE_VOICE_RECOGNITION:
+            return media::AudioSourceType::VOICE_RECOGNITION;
+        case AUDIO_SOURCE_VOICE_COMMUNICATION:
+            return media::AudioSourceType::VOICE_COMMUNICATION;
+        case AUDIO_SOURCE_REMOTE_SUBMIX:
+            return media::AudioSourceType::REMOTE_SUBMIX;
+        case AUDIO_SOURCE_UNPROCESSED:
+            return media::AudioSourceType::UNPROCESSED;
+        case AUDIO_SOURCE_VOICE_PERFORMANCE:
+            return media::AudioSourceType::VOICE_PERFORMANCE;
+        case AUDIO_SOURCE_ECHO_REFERENCE:
+            return media::AudioSourceType::ECHO_REFERENCE;
+        case AUDIO_SOURCE_FM_TUNER:
+            return media::AudioSourceType::FM_TUNER;
+        case AUDIO_SOURCE_HOTWORD:
+            return media::AudioSourceType::HOTWORD;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<audio_session_t> aidl2legacy_AudioSessionType_audio_session_t(
+        media::AudioSessionType aidl) {
+    switch (aidl) {
+        case media::AudioSessionType::DEVICE:
+            return AUDIO_SESSION_DEVICE;
+        case media::AudioSessionType::OUTPUT_STAGE:
+            return AUDIO_SESSION_OUTPUT_STAGE;
+        case media::AudioSessionType::OUTPUT_MIX:
+            return AUDIO_SESSION_OUTPUT_MIX;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<media::AudioSessionType> legacy2aidl_audio_session_t_AudioSessionType(
+        audio_session_t legacy) {
+    switch (legacy) {
+        case AUDIO_SESSION_DEVICE:
+            return media::AudioSessionType::DEVICE;
+        case AUDIO_SESSION_OUTPUT_STAGE:
+            return media::AudioSessionType::OUTPUT_STAGE;
+        case AUDIO_SESSION_OUTPUT_MIX:
+            return media::AudioSessionType::OUTPUT_MIX;
+        default:
+            return unexpected(BAD_VALUE);
+    }
+}
+
+// This type is unnamed in the original definition, thus we name it here.
+using audio_port_config_mix_ext_usecase = decltype(audio_port_config_mix_ext::usecase);
+
+ConversionResult<audio_port_config_mix_ext_usecase> aidl2legacy_AudioPortConfigMixExtUseCase(
+        const media::AudioPortConfigMixExtUseCase& aidl, media::AudioPortRole role) {
+    audio_port_config_mix_ext_usecase legacy;
+
+    // Our way of representing a union in AIDL is to have multiple vectors and require that exactly
+    // one of the them has size 1 and the rest are empty.
+    size_t totalSize = aidl.stream.size() + aidl.source.size();
+    if (totalSize > 1) {
+        return unexpected(BAD_VALUE);
+    }
+
+    switch (role) {
+        case media::AudioPortRole::NONE:
+            if (totalSize != 0) {
+                return unexpected(BAD_VALUE);
+            }
+            break;
+
+        case media::AudioPortRole::SOURCE:
+            // This is not a bug. A SOURCE role corresponds to the stream field.
+            if (aidl.stream.empty()) {
+                return unexpected(BAD_VALUE);
+            }
+            legacy.stream = VALUE_OR_RETURN(
+                    aidl2legacy_AudioStreamType_audio_stream_type_t(aidl.stream[0]));
+            break;
+
+        case media::AudioPortRole::SINK:
+            // This is not a bug. A SINK role corresponds to the source field.
+            if (aidl.source.empty()) {
+                return unexpected(BAD_VALUE);
+            }
+            legacy.source =
+                    VALUE_OR_RETURN(aidl2legacy_AudioSourceType_audio_source_t(aidl.source[0]));
+            break;
+
+        default:
+            LOG_ALWAYS_FATAL("Shouldn't get here");
+    }
+    return legacy;
+}
+
+ConversionResult<media::AudioPortConfigMixExtUseCase> legacy2aidl_AudioPortConfigMixExtUseCase(
+        const audio_port_config_mix_ext_usecase& legacy, audio_port_role_t role) {
+    media::AudioPortConfigMixExtUseCase aidl;
+
+    switch (role) {
+        case AUDIO_PORT_ROLE_NONE:
+            break;
+        case AUDIO_PORT_ROLE_SOURCE:
+            // This is not a bug. A SOURCE role corresponds to the stream field.
+            aidl.stream.push_back(VALUE_OR_RETURN(
+                                          legacy2aidl_audio_stream_type_t_AudioStreamType(
+                                                  legacy.stream)));
+            break;
+        case AUDIO_PORT_ROLE_SINK:
+            // This is not a bug. A SINK role corresponds to the source field.
+            aidl.source.push_back(
+                    VALUE_OR_RETURN(legacy2aidl_audio_source_t_AudioSourceType(legacy.source)));
+            break;
+        default:
+            LOG_ALWAYS_FATAL("Shouldn't get here");
+    }
+    return aidl;
+}
+
+ConversionResult<audio_port_config_mix_ext> aidl2legacy_AudioPortConfigMixExt(
+        const media::AudioPortConfigMixExt& aidl, media::AudioPortRole role) {
+    audio_port_config_mix_ext legacy;
+    legacy.hw_module = VALUE_OR_RETURN(convertReinterpret<audio_module_handle_t>(aidl.hwModule));
+    legacy.handle = VALUE_OR_RETURN(convertReinterpret<audio_io_handle_t>(aidl.handle));
+    legacy.usecase = VALUE_OR_RETURN(aidl2legacy_AudioPortConfigMixExtUseCase(aidl.usecase, role));
+    return legacy;
+}
+
+ConversionResult<media::AudioPortConfigMixExt> legacy2aidl_AudioPortConfigMixExt(
+        const audio_port_config_mix_ext& legacy, audio_port_role_t role) {
+    media::AudioPortConfigMixExt aidl;
+    aidl.hwModule = VALUE_OR_RETURN(convertReinterpret<int32_t>(legacy.hw_module));
+    aidl.handle = VALUE_OR_RETURN(convertReinterpret<int32_t>(legacy.handle));
+    aidl.usecase = VALUE_OR_RETURN(legacy2aidl_AudioPortConfigMixExtUseCase(legacy.usecase, role));
+    return aidl;
+}
+
+ConversionResult<audio_port_config_session_ext> aidl2legacy_AudioPortConfigSessionExt(
+        const media::AudioPortConfigSessionExt& aidl) {
+    audio_port_config_session_ext legacy;
+    legacy.session = VALUE_OR_RETURN(aidl2legacy_AudioSessionType_audio_session_t(aidl.session));
+    return legacy;
+}
+
+ConversionResult<media::AudioPortConfigSessionExt> legacy2aidl_AudioPortConfigSessionExt(
+        const audio_port_config_session_ext& legacy) {
+    media::AudioPortConfigSessionExt aidl;
+    aidl.session = VALUE_OR_RETURN(legacy2aidl_audio_session_t_AudioSessionType(legacy.session));
+    return aidl;
+}
+
+// This type is unnamed in the original definition, thus we name it here.
+using audio_port_config_ext = decltype(audio_port_config::ext);
+
+ConversionResult<audio_port_config_ext> aidl2legacy_AudioPortConfigExt(
+        const media::AudioPortConfigExt& aidl, media::AudioPortType type,
+        media::AudioPortRole role) {
+    audio_port_config_ext legacy;
+    // Our way of representing a union in AIDL is to have multiple vectors and require that at most
+    // one of the them has size 1 and the rest are empty.
+    size_t totalSize = aidl.device.size() + aidl.mix.size() + aidl.session.size();
+    if (totalSize > 1) {
+        return unexpected(BAD_VALUE);
+    }
+    switch (type) {
+        case media::AudioPortType::NONE:
+            if (totalSize != 0) {
+                return unexpected(BAD_VALUE);
+            }
+            break;
+        case media::AudioPortType::DEVICE:
+            if (aidl.device.empty()) {
+                return unexpected(BAD_VALUE);
+            }
+            legacy.device = VALUE_OR_RETURN(aidl2legacy_AudioPortConfigDeviceExt(aidl.device[0]));
+            break;
+        case media::AudioPortType::MIX:
+            if (aidl.mix.empty()) {
+                return unexpected(BAD_VALUE);
+            }
+            legacy.mix = VALUE_OR_RETURN(aidl2legacy_AudioPortConfigMixExt(aidl.mix[0], role));
+            break;
+        case media::AudioPortType::SESSION:
+            if (aidl.session.empty()) {
+                return unexpected(BAD_VALUE);
+            }
+            legacy.session =
+                    VALUE_OR_RETURN(aidl2legacy_AudioPortConfigSessionExt(aidl.session[0]));
+            break;
+        default:
+            LOG_ALWAYS_FATAL("Shouldn't get here");
+    }
+    return legacy;
+}
+
+ConversionResult<media::AudioPortConfigExt> legacy2aidl_AudioPortConfigExt(
+        const audio_port_config_ext& legacy, audio_port_type_t type, audio_port_role_t role) {
+    media::AudioPortConfigExt aidl;
+
+    switch (type) {
+        case AUDIO_PORT_TYPE_NONE:
+            break;
+        case AUDIO_PORT_TYPE_DEVICE:
+            aidl.device.push_back(
+                    VALUE_OR_RETURN(legacy2aidl_AudioPortConfigDeviceExt(legacy.device)));
+            break;
+        case AUDIO_PORT_TYPE_MIX:
+            aidl.mix.push_back(
+                    VALUE_OR_RETURN(legacy2aidl_AudioPortConfigMixExt(legacy.mix, role)));
+            break;
+        case AUDIO_PORT_TYPE_SESSION:
+            aidl.session.push_back(
+                    VALUE_OR_RETURN(legacy2aidl_AudioPortConfigSessionExt(legacy.session)));
+            break;
+        default:
+            LOG_ALWAYS_FATAL("Shouldn't get here");
+    }
+    return aidl;
+}
+
+ConversionResult<audio_port_config> aidl2legacy_AudioPortConfig_audio_port_config(
+        const media::AudioPortConfig& aidl) {
+    audio_port_config legacy;
+    legacy.id = VALUE_OR_RETURN(convertReinterpret<audio_port_handle_t>(aidl.id));
+    legacy.role = VALUE_OR_RETURN(aidl2legacy_AudioPortRole_audio_port_role_t(aidl.role));
+    legacy.type = VALUE_OR_RETURN(aidl2legacy_AudioPortType_audio_port_type_t(aidl.type));
+    legacy.config_mask = VALUE_OR_RETURN(aidl2legacy_int32_t_config_mask(aidl.configMask));
+    if (bitmaskIsSet(aidl.configMask, media::AudioPortConfigType::SAMPLE_RATE)) {
+        legacy.sample_rate = VALUE_OR_RETURN(convertIntegral<unsigned int>(aidl.sampleRate));
+    }
+    if (bitmaskIsSet(aidl.configMask, media::AudioPortConfigType::CHANNEL_MASK)) {
+        legacy.channel_mask =
+                VALUE_OR_RETURN(aidl2legacy_int32_t_audio_channel_mask_t(aidl.channelMask));
+    }
+    if (bitmaskIsSet(aidl.configMask, media::AudioPortConfigType::FORMAT)) {
+        legacy.format = VALUE_OR_RETURN(aidl2legacy_AudioFormat_audio_format_t(aidl.format));
+    }
+    if (bitmaskIsSet(aidl.configMask, media::AudioPortConfigType::GAIN)) {
+        legacy.gain = VALUE_OR_RETURN(
+                aidl2legacy_AudioGainConfig_audio_gain_config(aidl.gain, aidl.role, aidl.type));
+    }
+    if (bitmaskIsSet(aidl.configMask, media::AudioPortConfigType::FLAGS)) {
+        legacy.flags = VALUE_OR_RETURN(
+                aidl2legacy_AudioIoFlags_audio_io_flags(aidl.flags, aidl.role, aidl.type));
+    }
+    legacy.ext = VALUE_OR_RETURN(aidl2legacy_AudioPortConfigExt(aidl.ext, aidl.type, aidl.role));
+    return legacy;
+}
+
+ConversionResult<media::AudioPortConfig> legacy2aidl_audio_port_config_AudioPortConfig(
+        const audio_port_config& legacy) {
+    media::AudioPortConfig aidl;
+    aidl.id = VALUE_OR_RETURN(convertReinterpret<audio_port_handle_t>(legacy.id));
+    aidl.role = VALUE_OR_RETURN(legacy2aidl_audio_port_role_t_AudioPortRole(legacy.role));
+    aidl.type = VALUE_OR_RETURN(legacy2aidl_audio_port_type_t_AudioPortType(legacy.type));
+    aidl.configMask = VALUE_OR_RETURN(legacy2aidl_config_mask_int32_t(legacy.config_mask));
+    if (legacy.config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+        aidl.sampleRate = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.sample_rate));
+    }
+    if (legacy.config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+        aidl.channelMask =
+                VALUE_OR_RETURN(legacy2aidl_audio_channel_mask_t_int32_t(legacy.channel_mask));
+    }
+    if (legacy.config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+        aidl.format = VALUE_OR_RETURN(legacy2aidl_audio_format_t_AudioFormat(legacy.format));
+    }
+    if (legacy.config_mask & AUDIO_PORT_CONFIG_GAIN) {
+        aidl.gain = VALUE_OR_RETURN(legacy2aidl_audio_gain_config_AudioGainConfig(
+                legacy.gain, legacy.role, legacy.type));
+    }
+    if (legacy.config_mask & AUDIO_PORT_CONFIG_FLAGS) {
+        aidl.flags = VALUE_OR_RETURN(
+                legacy2aidl_audio_io_flags_AudioIoFlags(legacy.flags, legacy.role, legacy.type));
+    }
+    aidl.ext =
+            VALUE_OR_RETURN(legacy2aidl_AudioPortConfigExt(legacy.ext, legacy.type, legacy.role));
+    return aidl;
+}
+
+ConversionResult<struct audio_patch> aidl2legacy_AudioPatch_audio_patch(
+        const media::AudioPatch& aidl) {
+    struct audio_patch legacy;
+    legacy.id = VALUE_OR_RETURN(convertReinterpret<audio_patch_handle_t>(aidl.id));
+    legacy.num_sinks = VALUE_OR_RETURN(convertIntegral<unsigned int>(aidl.sinks.size()));
+    if (legacy.num_sinks > AUDIO_PATCH_PORTS_MAX) {
+        return unexpected(BAD_VALUE);
+    }
+    for (size_t i = 0; i < legacy.num_sinks; ++i) {
+        legacy.sinks[i] =
+                VALUE_OR_RETURN(aidl2legacy_AudioPortConfig_audio_port_config(aidl.sinks[i]));
+    }
+    legacy.num_sources = VALUE_OR_RETURN(convertIntegral<unsigned int>(aidl.sources.size()));
+    if (legacy.num_sources > AUDIO_PATCH_PORTS_MAX) {
+        return unexpected(BAD_VALUE);
+    }
+    for (size_t i = 0; i < legacy.num_sources; ++i) {
+        legacy.sources[i] =
+                VALUE_OR_RETURN(aidl2legacy_AudioPortConfig_audio_port_config(aidl.sources[i]));
+    }
+    return legacy;
+}
+
+ConversionResult<media::AudioPatch> legacy2aidl_audio_patch_AudioPatch(
+        const struct audio_patch& legacy) {
+    media::AudioPatch aidl;
+    aidl.id = VALUE_OR_RETURN(convertReinterpret<int32_t>(legacy.id));
+
+    if (legacy.num_sinks > AUDIO_PATCH_PORTS_MAX) {
+        return unexpected(BAD_VALUE);
+    }
+    for (unsigned int i = 0; i < legacy.num_sinks; ++i) {
+        aidl.sinks.push_back(
+                VALUE_OR_RETURN(legacy2aidl_audio_port_config_AudioPortConfig(legacy.sinks[i])));
+    }
+    if (legacy.num_sources > AUDIO_PATCH_PORTS_MAX) {
+        return unexpected(BAD_VALUE);
+    }
+    for (unsigned int i = 0; i < legacy.num_sources; ++i) {
+        aidl.sources.push_back(
+                VALUE_OR_RETURN(legacy2aidl_audio_port_config_AudioPortConfig(legacy.sources[i])));
+    }
+    return aidl;
+}
+
+ConversionResult<sp<AudioIoDescriptor>> aidl2legacy_AudioIoDescriptor_AudioIoDescriptor(
+        const media::AudioIoDescriptor& aidl) {
+    sp<AudioIoDescriptor> legacy(new AudioIoDescriptor());
+    legacy->mIoHandle = VALUE_OR_RETURN(convertReinterpret<audio_io_handle_t>(aidl.ioHandle));
+    legacy->mPatch = VALUE_OR_RETURN(aidl2legacy_AudioPatch_audio_patch(aidl.patch));
+    legacy->mSamplingRate = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.samplingRate));
+    legacy->mFormat = VALUE_OR_RETURN(aidl2legacy_AudioFormat_audio_format_t(aidl.format));
+    legacy->mChannelMask =
+            VALUE_OR_RETURN(aidl2legacy_int32_t_audio_channel_mask_t(aidl.channelMask));
+    legacy->mFrameCount = VALUE_OR_RETURN(convertIntegral<size_t>(aidl.frameCount));
+    legacy->mFrameCountHAL = VALUE_OR_RETURN(convertIntegral<size_t>(aidl.frameCountHAL));
+    legacy->mLatency = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.latency));
+    legacy->mPortId = VALUE_OR_RETURN(convertReinterpret<audio_port_handle_t>(aidl.portId));
+    return legacy;
+}
+
+ConversionResult<media::AudioIoDescriptor> legacy2aidl_AudioIoDescriptor_AudioIoDescriptor(
+        const sp<AudioIoDescriptor>& legacy) {
+    media::AudioIoDescriptor aidl;
+    aidl.ioHandle = VALUE_OR_RETURN(convertReinterpret<int32_t>(legacy->mIoHandle));
+    aidl.patch = VALUE_OR_RETURN(legacy2aidl_audio_patch_AudioPatch(legacy->mPatch));
+    aidl.samplingRate = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy->mSamplingRate));
+    aidl.format = VALUE_OR_RETURN(legacy2aidl_audio_format_t_AudioFormat(legacy->mFormat));
+    aidl.channelMask = VALUE_OR_RETURN(convertReinterpret<int32_t>(legacy->mChannelMask));
+    aidl.frameCount = VALUE_OR_RETURN(convertIntegral<int64_t>(legacy->mFrameCount));
+    aidl.frameCountHAL = VALUE_OR_RETURN(convertIntegral<int64_t>(legacy->mFrameCountHAL));
+    aidl.latency = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy->mLatency));
+    aidl.portId = VALUE_OR_RETURN(convertReinterpret<int32_t>(legacy->mPortId));
+    return aidl;
+}
+
+}  // namespace android
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index d7e9461..fef0ca9 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -2,6 +2,7 @@
     name: "libaudioclient_headers",
     vendor_available: true,
     min_sdk_version: "29",
+    host_supported: true,
 
     header_libs: [
         "libaudiofoundation_headers",
@@ -12,7 +13,12 @@
     export_header_lib_headers: [
         "libaudiofoundation_headers",
     ],
-    host_supported: true,
+    static_libs: [
+        "audioflinger-aidl-unstable-cpp",
+    ],
+    export_static_lib_headers: [
+        "audioflinger-aidl-unstable-cpp",
+    ],
     target: {
         darwin: {
             enabled: false,
@@ -29,6 +35,7 @@
         "AudioVolumeGroup.cpp",
     ],
     shared_libs: [
+        "audioflinger-aidl-unstable-cpp",
         "capture_state_listener-aidl-cpp",
         "libaudiofoundation",
         "libaudioutils",
@@ -44,6 +51,7 @@
     include_dirs: ["system/media/audio_utils/include"],
     export_include_dirs: ["include"],
     export_shared_lib_headers: [
+        "audioflinger-aidl-unstable-cpp",
         "capture_state_listener-aidl-cpp",
     ],
 }
@@ -73,7 +81,6 @@
         "AudioTrack.cpp",
         "AudioTrackShared.cpp",
         "IAudioFlinger.cpp",
-        "IAudioFlingerClient.cpp",
         "IAudioPolicyService.cpp",
         "IAudioPolicyServiceClient.cpp",
         "IAudioTrack.cpp",
@@ -83,7 +90,9 @@
         "TrackPlayerBase.cpp",
     ],
     shared_libs: [
+        "audioflinger-aidl-unstable-cpp",
         "capture_state_listener-aidl-cpp",
+        "libaudioclient_aidl_conversion",
         "libaudiofoundation",
         "libaudioutils",
         "libaudiopolicy",
@@ -101,7 +110,10 @@
         "libutils",
         "libvibrator",
     ],
-    export_shared_lib_headers: ["libbinder"],
+    export_shared_lib_headers: [
+        "audioflinger-aidl-unstable-cpp",
+        "libbinder",
+    ],
 
     include_dirs: [
         "frameworks/av/media/libnbaio/include_mono/",
@@ -140,6 +152,32 @@
     },
 }
 
+cc_library_shared {
+    name: "libaudioclient_aidl_conversion",
+    srcs: ["AidlConversion.cpp"],
+    local_include_dirs: ["include"],
+    shared_libs: [
+        "audioclient-types-aidl-unstable-cpp",
+        "libbase",
+        "liblog",
+        "libutils",
+    ],
+    export_shared_lib_headers: [
+        "audioclient-types-aidl-unstable-cpp",
+    ],
+    cflags: [
+        "-Wall",
+        "-Werror",
+        "-Wno-error=deprecated-declarations",
+    ],
+    sanitize: {
+        misc_undefined: [
+            "unsigned-integer-overflow",
+            "signed-integer-overflow",
+        ],
+    },
+}
+
 // AIDL interface between libaudioclient and framework.jar
 filegroup {
     name: "libaudioclient_aidl",
@@ -189,3 +227,61 @@
         "shared-file-region-aidl",
     ],
 }
+
+aidl_interface {
+    name: "audioclient-types-aidl",
+    unstable: true,
+    host_supported: true,
+    vendor_available: true,
+    double_loadable: true,
+    local_include_dir: "aidl",
+    srcs: [
+        "aidl/android/media/AudioGainConfig.aidl",
+        "aidl/android/media/AudioGainMode.aidl",
+        "aidl/android/media/AudioInputFlags.aidl",
+        "aidl/android/media/AudioIoConfigEvent.aidl",
+        "aidl/android/media/AudioIoDescriptor.aidl",
+        "aidl/android/media/AudioIoFlags.aidl",
+        "aidl/android/media/AudioOutputFlags.aidl",
+        "aidl/android/media/AudioPatch.aidl",
+        "aidl/android/media/AudioPortConfig.aidl",
+        "aidl/android/media/AudioPortConfigType.aidl",
+        "aidl/android/media/AudioPortConfigDeviceExt.aidl",
+        "aidl/android/media/AudioPortConfigExt.aidl",
+        "aidl/android/media/AudioPortConfigMixExt.aidl",
+        "aidl/android/media/AudioPortConfigMixExtUseCase.aidl",
+        "aidl/android/media/AudioPortConfigSessionExt.aidl",
+        "aidl/android/media/AudioPortRole.aidl",
+        "aidl/android/media/AudioPortType.aidl",
+        "aidl/android/media/AudioSessionType.aidl",
+        "aidl/android/media/AudioSourceType.aidl",
+        "aidl/android/media/AudioStreamType.aidl",
+    ],
+    imports: [
+        "audio_common-aidl",
+    ],
+}
+
+aidl_interface {
+    name: "audioflinger-aidl",
+    unstable: true,
+    local_include_dir: "aidl",
+    host_supported: true,
+    vendor_available: true,
+    srcs: [
+        "aidl/android/media/IAudioFlingerClient.aidl",
+    ],
+    imports: [
+        "audioclient-types-aidl",
+    ],
+    double_loadable: true,
+    backend: {
+        cpp: {
+            min_sdk_version: "29",
+            apex_available: [
+                "//apex_available:platform",
+                "com.android.media",
+            ],
+        },
+    },
+}
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index edb0889..0507879 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -23,6 +23,7 @@
 #include <binder/IServiceManager.h>
 #include <binder/ProcessState.h>
 #include <binder/IPCThreadState.h>
+#include <media/AidlConversion.h>
 #include <media/AudioResamplerPublic.h>
 #include <media/AudioSystem.h>
 #include <media/IAudioFlinger.h>
@@ -32,10 +33,17 @@
 
 #include <system/audio.h>
 
+#define VALUE_OR_RETURN(x) \
+    ({ auto _tmp = (x); \
+       if (!_tmp.ok()) return Status::fromStatusT(_tmp.error()); \
+       _tmp.value(); })
+
 // ----------------------------------------------------------------------------
 
 namespace android {
 
+using binder::Status;
+
 // client singleton for AudioFlinger binder interface
 Mutex AudioSystem::gLock;
 Mutex AudioSystem::gLockErrorCallbacks;
@@ -521,11 +529,17 @@
     ALOGW("AudioFlinger server died!");
 }
 
-void AudioSystem::AudioFlingerClient::ioConfigChanged(audio_io_config_event event,
-                                                      const sp<AudioIoDescriptor>& ioDesc) {
+Status AudioSystem::AudioFlingerClient::ioConfigChanged(
+        media::AudioIoConfigEvent _event,
+        const media::AudioIoDescriptor& _ioDesc) {
+    audio_io_config_event event = VALUE_OR_RETURN(
+            aidl2legacy_AudioIoConfigEvent_audio_io_config_event(_event));
+    sp<AudioIoDescriptor> ioDesc(
+            VALUE_OR_RETURN(aidl2legacy_AudioIoDescriptor_AudioIoDescriptor(_ioDesc)));
+
     ALOGV("ioConfigChanged() event %d", event);
 
-    if (ioDesc == 0 || ioDesc->mIoHandle == AUDIO_IO_HANDLE_NONE) return;
+    if (ioDesc->mIoHandle == AUDIO_IO_HANDLE_NONE) return Status::ok();
 
     audio_port_handle_t deviceId = AUDIO_PORT_HANDLE_NONE;
     std::vector<sp<AudioDeviceCallback>> callbacksToCall;
@@ -640,6 +654,8 @@
         // If callbacksToCall is not empty, it implies ioDesc->mIoHandle and deviceId are valid
         cb->onAudioDeviceUpdate(ioDesc->mIoHandle, deviceId);
     }
+
+    return Status::ok();
 }
 
 status_t AudioSystem::AudioFlingerClient::getInputBufferSize(
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index 7c304a1..d86182e 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -373,7 +373,7 @@
         return reply.readString8();
     }
 
-    virtual void registerClient(const sp<IAudioFlingerClient>& client)
+    virtual void registerClient(const sp<media::IAudioFlingerClient>& client)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -1213,7 +1213,7 @@
 
         case REGISTER_CLIENT: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient>(
+            sp<media::IAudioFlingerClient> client = interface_cast<media::IAudioFlingerClient>(
                     data.readStrongBinder());
             registerClient(client);
             return NO_ERROR;
diff --git a/media/libaudioclient/IAudioFlingerClient.cpp b/media/libaudioclient/IAudioFlingerClient.cpp
deleted file mode 100644
index 47eb7dc..0000000
--- a/media/libaudioclient/IAudioFlingerClient.cpp
+++ /dev/null
@@ -1,91 +0,0 @@
-/*
- * Copyright (C) 2009 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 "IAudioFlingerClient"
-#include <utils/Log.h>
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <binder/Parcel.h>
-
-#include <media/IAudioFlingerClient.h>
-#include <media/AudioSystem.h>
-
-namespace android {
-
-enum {
-    IO_CONFIG_CHANGED = IBinder::FIRST_CALL_TRANSACTION
-};
-
-class BpAudioFlingerClient : public BpInterface<IAudioFlingerClient>
-{
-public:
-    explicit BpAudioFlingerClient(const sp<IBinder>& impl)
-        : BpInterface<IAudioFlingerClient>(impl)
-    {
-    }
-
-    void ioConfigChanged(audio_io_config_event event, const sp<AudioIoDescriptor>& ioDesc)
-    {
-        Parcel data, reply;
-        data.writeInterfaceToken(IAudioFlingerClient::getInterfaceDescriptor());
-        data.writeInt32(event);
-        data.writeInt32((int32_t)ioDesc->mIoHandle);
-        data.write(&ioDesc->mPatch, sizeof(struct audio_patch));
-        data.writeInt32(ioDesc->mSamplingRate);
-        data.writeInt32(ioDesc->mFormat);
-        data.writeInt32(ioDesc->mChannelMask);
-        data.writeInt64(ioDesc->mFrameCount);
-        data.writeInt64(ioDesc->mFrameCountHAL);
-        data.writeInt32(ioDesc->mLatency);
-        data.writeInt32(ioDesc->mPortId);
-        remote()->transact(IO_CONFIG_CHANGED, data, &reply, IBinder::FLAG_ONEWAY);
-    }
-};
-
-IMPLEMENT_META_INTERFACE(AudioFlingerClient, "android.media.IAudioFlingerClient");
-
-// ----------------------------------------------------------------------
-
-status_t BnAudioFlingerClient::onTransact(
-    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
-{
-    switch (code) {
-    case IO_CONFIG_CHANGED: {
-            CHECK_INTERFACE(IAudioFlingerClient, data, reply);
-            audio_io_config_event event = (audio_io_config_event)data.readInt32();
-            sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor();
-            ioDesc->mIoHandle = (audio_io_handle_t) data.readInt32();
-            data.read(&ioDesc->mPatch, sizeof(struct audio_patch));
-            ioDesc->mSamplingRate = data.readInt32();
-            ioDesc->mFormat = (audio_format_t) data.readInt32();
-            ioDesc->mChannelMask = (audio_channel_mask_t) data.readInt32();
-            ioDesc->mFrameCount = data.readInt64();
-            ioDesc->mFrameCountHAL = data.readInt64();
-            ioDesc->mLatency = data.readInt32();
-            ioDesc->mPortId = data.readInt32();
-            ioConfigChanged(event, ioDesc);
-            return NO_ERROR;
-        } break;
-        default:
-            return BBinder::onTransact(code, data, reply, flags);
-    }
-}
-
-// ----------------------------------------------------------------------------
-
-} // namespace android
diff --git a/media/libaudioclient/aidl/android/media/AudioGainConfig.aidl b/media/libaudioclient/aidl/android/media/AudioGainConfig.aidl
new file mode 100644
index 0000000..b93c2dc
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioGainConfig.aidl
@@ -0,0 +1,44 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+/**
+ * {@hide}
+ */
+parcelable AudioGainConfig {
+    /** Index of the corresponding audio_gain in the audio_port gains[] table. */
+    int index;
+
+    /** Mode requested for this command. Bitfield indexed by AudioGainMode. */
+    int mode;
+
+    /**
+     * Channels which gain value follows. N/A in joint mode.
+     * Interpreted as audio_channel_mask_t.
+     */
+    int channelMask;
+
+    /**
+     * Gain values in millibels.
+     * For each channel ordered from LSb to MSb in channel mask. The number of values is 1 in joint
+     * mode, otherwise equals the number of bits implied by channelMask.
+     */
+    int[]  values;
+
+    /** Ramp duration in ms. */
+    int rampDurationMs;
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioGainMode.aidl b/media/libaudioclient/aidl/android/media/AudioGainMode.aidl
new file mode 100644
index 0000000..39395e5
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioGainMode.aidl
@@ -0,0 +1,23 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+@Backing(type="int")
+enum AudioGainMode {
+    JOINT    = 0,
+    CHANNELS = 1,
+    RAMP     = 2,
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioInputFlags.aidl b/media/libaudioclient/aidl/android/media/AudioInputFlags.aidl
new file mode 100644
index 0000000..8f517e7
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioInputFlags.aidl
@@ -0,0 +1,28 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+@Backing(type="int")
+enum AudioInputFlags {
+    FAST       = 0,
+    HW_HOTWORD = 1,
+    RAW        = 2,
+    SYNC       = 3,
+    MMAP_NOIRQ = 4,
+    VOIP_TX    = 5,
+    HW_AV_SYNC = 6,
+    DIRECT     = 7,
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioIoConfigEvent.aidl b/media/libaudioclient/aidl/android/media/AudioIoConfigEvent.aidl
new file mode 100644
index 0000000..d5f23a1
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioIoConfigEvent.aidl
@@ -0,0 +1,33 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+/**
+ * {@hide}
+ */
+@Backing(type="int")
+enum AudioIoConfigEvent {
+    OUTPUT_REGISTERED = 0,
+    OUTPUT_OPENED = 1,
+    OUTPUT_CLOSED = 2,
+    OUTPUT_CONFIG_CHANGED = 3,
+    INPUT_REGISTERED = 4,
+    INPUT_OPENED = 5,
+    INPUT_CLOSED = 6,
+    INPUT_CONFIG_CHANGED = 7,
+    CLIENT_STARTED = 8,
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioIoDescriptor.aidl b/media/libaudioclient/aidl/android/media/AudioIoDescriptor.aidl
new file mode 100644
index 0000000..876ef9b
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioIoDescriptor.aidl
@@ -0,0 +1,42 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+import android.media.AudioPatch;
+import android.media.audio.common.AudioFormat;
+
+/**
+ * {@hide}
+ */
+parcelable AudioIoDescriptor {
+    /** Interpreted as audio_io_handle_t. */
+    int ioHandle;
+    AudioPatch patch;
+    int samplingRate;
+    AudioFormat format;
+    /** Interpreted as audio_channel_mask_t. */
+    int channelMask;
+    long frameCount;
+    long frameCountHAL;
+    /** Only valid for output. */
+    int latency;
+    /**
+     * Interpreted as audio_port_handle_t.
+     * valid for event AUDIO_CLIENT_STARTED.
+     */
+    int portId;
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioIoFlags.aidl b/media/libaudioclient/aidl/android/media/AudioIoFlags.aidl
new file mode 100644
index 0000000..1fe2acc
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioIoFlags.aidl
@@ -0,0 +1,30 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+/**
+ * {@hide}
+ */
+// TODO(b/150948558): This should be a union. In the meantime, we require
+// that exactly one of the below arrays has a single element and the rest
+// are empty.
+parcelable AudioIoFlags {
+    /** Bitmask indexed by AudioInputFlags. */
+    int[] input;
+    /** Bitmask indexed by AudioOutputFlags. */
+    int[] output;
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioOutputFlags.aidl b/media/libaudioclient/aidl/android/media/AudioOutputFlags.aidl
new file mode 100644
index 0000000..aebf871
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioOutputFlags.aidl
@@ -0,0 +1,35 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+@Backing(type="int")
+enum AudioOutputFlags {
+    DIRECT           = 0,
+    PRIMARY          = 1,
+    FAST             = 2,
+    DEEP_BUFFER      = 3,
+    COMPRESS_OFFLOAD = 4,
+    NON_BLOCKING     = 5,
+    HW_AV_SYNC       = 6,
+    TTS              = 7,
+    RAW              = 8,
+    SYNC             = 9,
+    IEC958_NONAUDIO  = 10,
+    DIRECT_PCM       = 11,
+    MMAP_NOIRQ       = 12,
+    VOIP_RX          = 13,
+    INCALL_MUSIC     = 14,
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioPatch.aidl b/media/libaudioclient/aidl/android/media/AudioPatch.aidl
new file mode 100644
index 0000000..8519faf
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioPatch.aidl
@@ -0,0 +1,32 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+import android.media.AudioPortConfig;
+
+/**
+ * {@hide}
+ */
+parcelable AudioPatch {
+    /**
+     * Patch unique ID.
+     * Interpreted as audio_patch_handle_t.
+     */
+    int id;
+    AudioPortConfig[] sources;
+    AudioPortConfig[] sinks;
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfig.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfig.aidl
new file mode 100644
index 0000000..2dd30a4
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioPortConfig.aidl
@@ -0,0 +1,59 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+import android.media.AudioGainConfig;
+import android.media.AudioIoFlags;
+import android.media.AudioPortConfigExt;
+import android.media.AudioPortConfigType;
+import android.media.AudioPortRole;
+import android.media.AudioPortType;
+import android.media.audio.common.AudioFormat;
+
+/**
+ * {@hide}
+ */
+parcelable AudioPortConfig {
+    /**
+     * Port unique ID.
+     * Interpreted as audio_port_handle_t.
+     */
+    int id;
+    /** Sink or source. */
+    AudioPortRole role;
+    /** Device, mix ... */
+    AudioPortType type;
+    /** Bitmask, indexed by AudioPortConfigType. */
+    int configMask;
+    /** Sampling rate in Hz. */
+    int sampleRate;
+    /**
+     * Channel mask, if applicable.
+     * Interpreted as audio_channel_mask_t.
+     * TODO: bitmask?
+     */
+    int channelMask;
+    /**
+     * Format, if applicable.
+     */
+    AudioFormat format;
+    /** Gain to apply, if applicable. */
+    AudioGainConfig gain;
+    /** Framework only: HW_AV_SYNC, DIRECT, ... */
+    AudioIoFlags flags;
+    AudioPortConfigExt ext;
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfigDeviceExt.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfigDeviceExt.aidl
new file mode 100644
index 0000000..a99aa9b
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioPortConfigDeviceExt.aidl
@@ -0,0 +1,36 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+/**
+ * {@hide}
+ */
+parcelable AudioPortConfigDeviceExt {
+    /**
+     * Module the device is attached to.
+     * Interpreted as audio_module_handle_t.
+     */
+    int hwModule;
+    /**
+     * Device type (e.g AUDIO_DEVICE_OUT_SPEAKER).
+     * Interpreted as audio_devices_t.
+     * TODO: Convert to a standalone AIDL representation.
+     */
+    int type;
+    /** Device address. "" if N/A. */
+    @utf8InCpp String address;
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfigExt.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfigExt.aidl
new file mode 100644
index 0000000..83e985e
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioPortConfigExt.aidl
@@ -0,0 +1,37 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+import android.media.AudioPortConfigDeviceExt;
+import android.media.AudioPortConfigMixExt;
+import android.media.AudioPortConfigSessionExt;
+
+/**
+ * {@hide}
+ */
+parcelable AudioPortConfigExt {
+    // TODO(b/150948558): This should be a union. In the meantime, we require
+    // that exactly one of the below arrays has a single element and the rest
+    // are empty.
+
+    /** Device specific info. */
+    AudioPortConfigDeviceExt[] device;
+    /** Mix specific info. */
+    AudioPortConfigMixExt[] mix;
+    /** Session specific info. */
+    AudioPortConfigSessionExt[] session;
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfigMixExt.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfigMixExt.aidl
new file mode 100644
index 0000000..d3226f2
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioPortConfigMixExt.aidl
@@ -0,0 +1,36 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+import android.media.AudioPortConfigMixExtUseCase;
+
+/**
+ * {@hide}
+ */
+parcelable AudioPortConfigMixExt {
+    /**
+     * Module the stream is attached to.
+     * Interpreted as audio_module_handle_t.
+     */
+    int hwModule;
+    /**
+     * I/O handle of the input/output stream.
+     * Interpreted as audio_io_handle_t.
+     */
+    int handle;
+    AudioPortConfigMixExtUseCase usecase;
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfigMixExtUseCase.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfigMixExtUseCase.aidl
new file mode 100644
index 0000000..675daf8
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioPortConfigMixExtUseCase.aidl
@@ -0,0 +1,34 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+import android.media.AudioSourceType;
+import android.media.AudioStreamType;
+
+/**
+ * {@hide}
+ */
+parcelable AudioPortConfigMixExtUseCase {
+    // TODO(b/150948558): This should be a union. In the meantime, we require
+    // that exactly one of the below arrays has a single element and the rest
+    // are empty.
+
+    /** This to be set if the containing config has the AudioPortRole::SOURCE role. */
+    AudioStreamType[] stream;
+    /** This to be set if the containing config has the AudioPortRole::SINK role. */
+    AudioSourceType[] source;
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfigSessionExt.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfigSessionExt.aidl
new file mode 100644
index 0000000..d3261d9
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioPortConfigSessionExt.aidl
@@ -0,0 +1,26 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+import android.media.AudioSessionType;
+
+/**
+ * {@hide}
+ */
+parcelable AudioPortConfigSessionExt {
+    AudioSessionType session;
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfigType.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfigType.aidl
new file mode 100644
index 0000000..c7bb4d8
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioPortConfigType.aidl
@@ -0,0 +1,25 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+@Backing(type="int")
+enum AudioPortConfigType {
+    SAMPLE_RATE  = 0,
+    CHANNEL_MASK = 1,
+    FORMAT       = 2,
+    GAIN         = 3,
+    FLAGS        = 4,
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortRole.aidl b/media/libaudioclient/aidl/android/media/AudioPortRole.aidl
new file mode 100644
index 0000000..3212325
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioPortRole.aidl
@@ -0,0 +1,23 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+@Backing(type="int")
+enum AudioPortRole {
+    NONE = 0,
+    SOURCE = 1,
+    SINK = 2,
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortType.aidl b/media/libaudioclient/aidl/android/media/AudioPortType.aidl
new file mode 100644
index 0000000..90eea9a
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioPortType.aidl
@@ -0,0 +1,24 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+@Backing(type="int")
+enum AudioPortType {
+    NONE = 0,
+    DEVICE = 1,
+    MIX = 2,
+    SESSION = 3,
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioSessionType.aidl b/media/libaudioclient/aidl/android/media/AudioSessionType.aidl
new file mode 100644
index 0000000..d305c29
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioSessionType.aidl
@@ -0,0 +1,25 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+@Backing(type="int")
+enum AudioSessionType {
+    DEVICE = -2,
+    OUTPUT_STAGE = -1,
+    OUTPUT_MIX = 0,
+    ALLOCATE = 0,
+    NONE = 0,
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioSourceType.aidl b/media/libaudioclient/aidl/android/media/AudioSourceType.aidl
new file mode 100644
index 0000000..f6ecc46
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioSourceType.aidl
@@ -0,0 +1,39 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+@Backing(type="int")
+enum AudioSourceType {
+    DEFAULT = 0,
+    MIC = 1,
+    VOICE_UPLINK = 2,
+    VOICE_DOWNLINK = 3,
+    VOICE_CALL = 4,
+    CAMCORDER = 5,
+    VOICE_RECOGNITION = 6,
+    VOICE_COMMUNICATION = 7,
+    REMOTE_SUBMIX = 8,
+    UNPROCESSED = 9,
+    VOICE_PERFORMANCE = 10,
+    ECHO_REFERENCE = 1997,
+    FM_TUNER = 1998,
+    /**
+     * A low-priority, preemptible audio source for for background software
+     * hotword detection. Same tuning as VOICE_RECOGNITION.
+     * Used only internally by the framework.
+     */
+    HOTWORD = 1999,
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioStreamType.aidl b/media/libaudioclient/aidl/android/media/AudioStreamType.aidl
new file mode 100644
index 0000000..803b87b
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioStreamType.aidl
@@ -0,0 +1,39 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+@Backing(type="int")
+enum AudioStreamType {
+    DEFAULT = -1,
+    VOICE_CALL = 0,
+    SYSTEM = 1,
+    RING = 2,
+    MUSIC = 3,
+    ALARM = 4,
+    NOTIFICATION = 5,
+    BLUETOOTH_SCO = 6,
+    ENFORCED_AUDIBLE = 7,
+    DTMF = 8,
+    TTS = 9,
+    ACCESSIBILITY = 10,
+    ASSISTANT = 11,
+    /** For dynamic policy output mixes. Only used by the audio policy */
+    REROUTING = 12,
+    /** For audio flinger tracks volume. Only used by the audioflinger */
+    PATCH = 13,
+    /** stream for corresponding to AUDIO_USAGE_CALL_ASSISTANT */
+    CALL_ASSISTANT = 14,
+}
diff --git a/media/libaudioclient/aidl/android/media/IAudioFlingerClient.aidl b/media/libaudioclient/aidl/android/media/IAudioFlingerClient.aidl
new file mode 100644
index 0000000..421c31c
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/IAudioFlingerClient.aidl
@@ -0,0 +1,30 @@
+/*
+ * Copyright (C) 2020 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.media;
+
+import android.media.AudioIoConfigEvent;
+import android.media.AudioIoDescriptor;
+
+/**
+ * A callback interface for AudioFlinger.
+ *
+ * {@hide}
+ */
+interface IAudioFlingerClient {
+    oneway void ioConfigChanged(AudioIoConfigEvent event,
+                                in AudioIoDescriptor ioDesc);
+}
diff --git a/media/libaudioclient/include/media/AidlConversion.h b/media/libaudioclient/include/media/AidlConversion.h
new file mode 100644
index 0000000..a1b9b82
--- /dev/null
+++ b/media/libaudioclient/include/media/AidlConversion.h
@@ -0,0 +1,147 @@
+/*
+ * Copyright (C) 2020 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>
+
+#include <android-base/result.h>
+#include <android/media/AudioGainMode.h>
+#include <android/media/AudioInputFlags.h>
+#include <android/media/AudioIoConfigEvent.h>
+#include <android/media/AudioIoDescriptor.h>
+#include <android/media/AudioOutputFlags.h>
+#include <android/media/AudioPortConfigType.h>
+
+#include <media/AudioIoDescriptor.h>
+
+namespace android {
+
+template <typename T>
+using ConversionResult = base::expected<T, status_t>;
+
+// The legacy enum is unnamed. Thus, we use int.
+ConversionResult<int> aidl2legacy_AudioPortConfigType(media::AudioPortConfigType aidl);
+// The legacy enum is unnamed. Thus, we use int.
+ConversionResult<media::AudioPortConfigType> legacy2aidl_AudioPortConfigType(int legacy);
+
+ConversionResult<unsigned int> aidl2legacy_int32_t_config_mask(int32_t aidl);
+ConversionResult<int32_t> legacy2aidl_config_mask_int32_t(unsigned int legacy);
+
+ConversionResult<audio_channel_mask_t> aidl2legacy_int32_t_audio_channel_mask_t(int32_t aidl);
+ConversionResult<int32_t> legacy2aidl_audio_channel_mask_t_int32_t(audio_channel_mask_t legacy);
+
+ConversionResult<audio_io_config_event> aidl2legacy_AudioIoConfigEvent_audio_io_config_event(
+        media::AudioIoConfigEvent aidl);
+ConversionResult<media::AudioIoConfigEvent> legacy2aidl_audio_io_config_event_AudioIoConfigEvent(
+        audio_io_config_event legacy);
+
+ConversionResult<audio_port_role_t> aidl2legacy_AudioPortRole_audio_port_role_t(
+        media::AudioPortRole aidl);
+ConversionResult<media::AudioPortRole> legacy2aidl_audio_port_role_t_AudioPortRole(
+        audio_port_role_t legacy);
+
+ConversionResult<audio_port_type_t> aidl2legacy_AudioPortType_audio_port_type_t(
+        media::AudioPortType aidl);
+ConversionResult<media::AudioPortType> legacy2aidl_audio_port_type_t_AudioPortType(
+        audio_port_type_t legacy);
+
+ConversionResult<audio_format_t> aidl2legacy_AudioFormat_audio_format_t(
+        media::audio::common::AudioFormat aidl);
+ConversionResult<media::audio::common::AudioFormat> legacy2aidl_audio_format_t_AudioFormat(
+        audio_format_t legacy);
+
+ConversionResult<int> aidl2legacy_AudioGainMode_int(media::AudioGainMode aidl);
+ConversionResult<media::AudioGainMode> legacy2aidl_int_AudioGainMode(int legacy);
+
+ConversionResult<audio_gain_mode_t> aidl2legacy_int32_t_audio_gain_mode_t(int32_t aidl);
+ConversionResult<int32_t> legacy2aidl_audio_gain_mode_t_int32_t(audio_gain_mode_t legacy);
+
+ConversionResult<audio_devices_t> aidl2legacy_int32_t_audio_devices_t(int32_t aidl);
+ConversionResult<int32_t> legacy2aidl_audio_devices_t_int32_t(audio_devices_t legacy);
+
+ConversionResult<audio_gain_config> aidl2legacy_AudioGainConfig_audio_gain_config(
+        const media::AudioGainConfig& aidl, media::AudioPortRole role, media::AudioPortType type);
+ConversionResult<media::AudioGainConfig> legacy2aidl_audio_gain_config_AudioGainConfig(
+        const audio_gain_config& legacy, audio_port_role_t role, audio_port_type_t type);
+
+ConversionResult<audio_input_flags_t> aidl2legacy_AudioInputFlags_audio_input_flags_t(
+        media::AudioInputFlags aidl);
+ConversionResult<media::AudioInputFlags> legacy2aidl_audio_input_flags_t_AudioInputFlags(
+        audio_input_flags_t legacy);
+
+ConversionResult<audio_output_flags_t> aidl2legacy_AudioOutputFlags_audio_output_flags_t(
+        media::AudioOutputFlags aidl);
+ConversionResult<media::AudioOutputFlags> legacy2aidl_audio_output_flags_t_AudioOutputFlags(
+        audio_output_flags_t legacy);
+
+ConversionResult<audio_input_flags_t> aidl2legacy_audio_input_flags_mask(int32_t aidl);
+ConversionResult<int32_t> legacy2aidl_audio_input_flags_mask(audio_input_flags_t legacy);
+
+ConversionResult<audio_output_flags_t> aidl2legacy_audio_output_flags_mask(int32_t aidl);
+ConversionResult<int32_t> legacy2aidl_audio_output_flags_mask(audio_output_flags_t legacy);
+
+ConversionResult<audio_io_flags> aidl2legacy_AudioIoFlags_audio_io_flags(
+        const media::AudioIoFlags& aidl, media::AudioPortRole role, media::AudioPortType type);
+ConversionResult<media::AudioIoFlags> legacy2aidl_audio_io_flags_AudioIoFlags(
+        const audio_io_flags& legacy, audio_port_role_t role, audio_port_type_t type);
+
+ConversionResult<audio_port_config_device_ext> aidl2legacy_AudioPortConfigDeviceExt(
+        const media::AudioPortConfigDeviceExt& aidl);
+ConversionResult<media::AudioPortConfigDeviceExt> legacy2aidl_AudioPortConfigDeviceExt(
+        const audio_port_config_device_ext& legacy);
+
+ConversionResult<audio_stream_type_t> aidl2legacy_AudioStreamType_audio_stream_type_t(
+        media::AudioStreamType aidl);
+ConversionResult<media::AudioStreamType> legacy2aidl_audio_stream_type_t_AudioStreamType(
+        audio_stream_type_t legacy);
+
+ConversionResult<audio_source_t> aidl2legacy_AudioSourceType_audio_source_t(
+        media::AudioSourceType aidl);
+ConversionResult<media::AudioSourceType> legacy2aidl_audio_source_t_AudioSourceType(
+        audio_source_t legacy);
+
+ConversionResult<audio_session_t> aidl2legacy_AudioSessionType_audio_session_t(
+        media::AudioSessionType aidl);
+ConversionResult<media::AudioSessionType> legacy2aidl_audio_session_t_AudioSessionType(
+        audio_session_t legacy);
+
+ConversionResult<audio_port_config_mix_ext> aidl2legacy_AudioPortConfigMixExt(
+        const media::AudioPortConfigMixExt& aidl, media::AudioPortRole role);
+ConversionResult<media::AudioPortConfigMixExt> legacy2aidl_AudioPortConfigMixExt(
+        const audio_port_config_mix_ext& legacy, audio_port_role_t role);
+
+ConversionResult<audio_port_config_session_ext> aidl2legacy_AudioPortConfigSessionExt(
+        const media::AudioPortConfigSessionExt& aidl);
+ConversionResult<media::AudioPortConfigSessionExt> legacy2aidl_AudioPortConfigSessionExt(
+        const audio_port_config_session_ext& legacy);
+
+ConversionResult<audio_port_config> aidl2legacy_AudioPortConfig_audio_port_config(
+        const media::AudioPortConfig& aidl);
+ConversionResult<media::AudioPortConfig> legacy2aidl_audio_port_config_AudioPortConfig(
+        const audio_port_config& legacy);
+
+ConversionResult<struct audio_patch> aidl2legacy_AudioPatch_audio_patch(
+        const media::AudioPatch& aidl);
+ConversionResult<media::AudioPatch> legacy2aidl_audio_patch_AudioPatch(
+        const struct audio_patch& legacy);
+
+ConversionResult<sp<AudioIoDescriptor>> aidl2legacy_AudioIoDescriptor_AudioIoDescriptor(
+        const media::AudioIoDescriptor& aidl);
+ConversionResult<media::AudioIoDescriptor> legacy2aidl_AudioIoDescriptor_AudioIoDescriptor(
+        const sp<AudioIoDescriptor>& legacy);
+
+}  // namespace android
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index 848743a..dfc1982 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -19,12 +19,12 @@
 
 #include <sys/types.h>
 
+#include <android/media/BnAudioFlingerClient.h>
 #include <media/AudioDeviceTypeAddr.h>
 #include <media/AudioPolicy.h>
 #include <media/AudioProductStrategy.h>
 #include <media/AudioVolumeGroup.h>
 #include <media/AudioIoDescriptor.h>
-#include <media/IAudioFlingerClient.h>
 #include <media/IAudioPolicyServiceClient.h>
 #include <media/MicrophoneInfo.h>
 #include <set>
@@ -531,7 +531,7 @@
 
 private:
 
-    class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient
+    class AudioFlingerClient: public IBinder::DeathRecipient, public media::BnAudioFlingerClient
     {
     public:
         AudioFlingerClient() :
@@ -551,9 +551,9 @@
 
         // indicate a change in the configuration of an output or input: keeps the cached
         // values for output/input parameters up-to-date in client process
-        virtual void ioConfigChanged(audio_io_config_event event,
-                                     const sp<AudioIoDescriptor>& ioDesc);
-
+        binder::Status ioConfigChanged(
+                media::AudioIoConfigEvent event,
+                const media::AudioIoDescriptor& ioDesc) override;
 
         status_t addAudioDeviceCallback(const wp<AudioDeviceCallback>& callback,
                                                audio_io_handle_t audioIo,
diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index a01b681..413db71 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -29,7 +29,6 @@
 #include <media/AudioClient.h>
 #include <media/DeviceDescriptorBase.h>
 #include <media/IAudioTrack.h>
-#include <media/IAudioFlingerClient.h>
 #include <system/audio.h>
 #include <system/audio_effect.h>
 #include <system/audio_policy.h>
@@ -39,6 +38,7 @@
 #include <vector>
 
 #include "android/media/IAudioRecord.h"
+#include "android/media/IAudioFlingerClient.h"
 #include "android/media/IAudioTrackCallback.h"
 #include "android/media/IEffect.h"
 #include "android/media/IEffectClient.h"
@@ -420,7 +420,7 @@
     // Register an object to receive audio input/output change and track notifications.
     // For a given calling pid, AudioFlinger disregards any registrations after the first.
     // Thus the IAudioFlingerClient must be a singleton per process.
-    virtual void registerClient(const sp<IAudioFlingerClient>& client) = 0;
+    virtual void registerClient(const sp<media::IAudioFlingerClient>& client) = 0;
 
     // retrieve the audio recording buffer size in bytes
     // FIXME This API assumes a route, and so should be deprecated.
diff --git a/media/libaudioclient/include/media/IAudioFlingerClient.h b/media/libaudioclient/include/media/IAudioFlingerClient.h
deleted file mode 100644
index 0080bc9..0000000
--- a/media/libaudioclient/include/media/IAudioFlingerClient.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_IAUDIOFLINGERCLIENT_H
-#define ANDROID_IAUDIOFLINGERCLIENT_H
-
-
-#include <utils/RefBase.h>
-#include <binder/IInterface.h>
-#include <utils/KeyedVector.h>
-#include <system/audio.h>
-#include <media/AudioIoDescriptor.h>
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-class IAudioFlingerClient : public IInterface
-{
-public:
-    DECLARE_META_INTERFACE(AudioFlingerClient);
-
-    // Notifies a change of audio input/output configuration.
-    virtual void ioConfigChanged(audio_io_config_event event,
-                                 const sp<AudioIoDescriptor>& ioDesc) = 0;
-
-};
-
-
-// ----------------------------------------------------------------------------
-
-class BnAudioFlingerClient : public BnInterface<IAudioFlingerClient>
-{
-public:
-    virtual status_t    onTransact( uint32_t code,
-                                    const Parcel& data,
-                                    Parcel* reply,
-                                    uint32_t flags = 0);
-};
-
-// ----------------------------------------------------------------------------
-
-}; // namespace android
-
-#endif // ANDROID_IAUDIOFLINGERCLIENT_H
diff --git a/media/libmediatranscoding/TranscodingClientManager.cpp b/media/libmediatranscoding/TranscodingClientManager.cpp
index b57baa5..ae1f7a5 100644
--- a/media/libmediatranscoding/TranscodingClientManager.cpp
+++ b/media/libmediatranscoding/TranscodingClientManager.cpp
@@ -31,7 +31,10 @@
 
 static_assert(sizeof(ClientIdType) == sizeof(void*), "ClientIdType should be pointer-sized");
 
-static constexpr const char* MEDIA_PROVIDER_PKG_NAME = "com.google.android.providers.media.module";
+static constexpr const char* MEDIA_PROVIDER_PKG_NAMES[] = {
+        "com.android.providers.media.module",
+        "com.google.android.providers.media.module",
+};
 
 using ::aidl::android::media::BnTranscodingClient;
 using ::aidl::android::media::IMediaTranscodingService;  // For service error codes
@@ -261,16 +264,16 @@
 TranscodingClientManager::TranscodingClientManager(
         const std::shared_ptr<ControllerClientInterface>& controller)
       : mDeathRecipient(AIBinder_DeathRecipient_new(BinderDiedCallback)),
-        mSessionController(controller),
-        mMediaProviderUid(-1) {
+        mSessionController(controller) {
     ALOGD("TranscodingClientManager started");
     uid_t mpuid;
-    if (TranscodingUidPolicy::getUidForPackage(String16(MEDIA_PROVIDER_PKG_NAME), mpuid) ==
-        NO_ERROR) {
-        ALOGI("Found MediaProvider uid: %d", mpuid);
-        mMediaProviderUid = mpuid;
-    } else {
-        ALOGW("Couldn't get uid for MediaProvider.");
+    for (const char* pkgName : MEDIA_PROVIDER_PKG_NAMES) {
+        if (TranscodingUidPolicy::getUidForPackage(String16(pkgName), mpuid) == NO_ERROR) {
+            ALOGI("Found %s's uid: %d", pkgName, mpuid);
+            mMediaProviderUid.insert(mpuid);
+        } else {
+            ALOGW("Couldn't get uid for %s.", pkgName);
+        }
     }
 }
 
@@ -303,7 +306,7 @@
 }
 
 bool TranscodingClientManager::isTrustedCallingUid(uid_t uid) {
-    if (uid > 0 && uid == mMediaProviderUid) {
+    if (uid > 0 && mMediaProviderUid.count(uid) > 0) {
         return true;
     }
 
diff --git a/media/libmediatranscoding/include/media/TranscodingClientManager.h b/media/libmediatranscoding/include/media/TranscodingClientManager.h
index 5feeae9..451f993 100644
--- a/media/libmediatranscoding/include/media/TranscodingClientManager.h
+++ b/media/libmediatranscoding/include/media/TranscodingClientManager.h
@@ -109,7 +109,7 @@
     ::ndk::ScopedAIBinder_DeathRecipient mDeathRecipient;
 
     std::shared_ptr<ControllerClientInterface> mSessionController;
-    uid_t mMediaProviderUid;
+    std::unordered_set<uid_t> mMediaProviderUid;
 
     static std::atomic<ClientIdType> sCookieCounter;
     static std::mutex sCookie2ClientLock;
diff --git a/media/libshmem/Android.bp b/media/libshmem/Android.bp
index fae98ed..c8d2284 100644
--- a/media/libshmem/Android.bp
+++ b/media/libshmem/Android.bp
@@ -41,6 +41,7 @@
     srcs: ["ShmemTest.cpp"],
     shared_libs: [
         "libbinder",
+        "libcutils",
         "libshmemcompat",
         "libshmemutil",
         "libutils",
diff --git a/media/libshmem/ShmemCompat.cpp b/media/libshmem/ShmemCompat.cpp
index 5dd83f4..44fe39c 100644
--- a/media/libshmem/ShmemCompat.cpp
+++ b/media/libshmem/ShmemCompat.cpp
@@ -24,15 +24,12 @@
 
 bool convertSharedFileRegionToIMemory(const SharedFileRegion& shmem,
                                       sp<IMemory>* result) {
+    assert(result != nullptr);
+
     if (!validateSharedFileRegion(shmem)) {
         return false;
     }
 
-    if (shmem.fd.get() < 0) {
-        *result = nullptr;
-        return true;
-    }
-
     // Heap offset and size must be page aligned.
     const size_t pageSize = getpagesize();
     const size_t pageMask = ~(pageSize - 1);
@@ -62,16 +59,19 @@
 
 bool convertIMemoryToSharedFileRegion(const sp<IMemory>& mem,
                                       SharedFileRegion* result) {
+    assert(mem != nullptr);
+    assert(result != nullptr);
+
     *result = SharedFileRegion();
-    if (mem == nullptr) {
-        return true;
-    }
 
     ssize_t offset;
     size_t size;
 
     sp<IMemoryHeap> heap = mem->getMemory(&offset, &size);
-    if (heap != nullptr) {
+    if (size > 0) {
+        if (heap == nullptr) {
+            return false;
+        }
         // Make sure the offset and size do not overflow from int64 boundaries.
         if (size > std::numeric_limits<int64_t>::max() ||
                 offset > std::numeric_limits<int64_t>::max() ||
@@ -90,9 +90,33 @@
         result->size = size;
         result->offset = heap->getOffset() + offset;
     }
-
     return true;
 }
 
+bool convertNullableSharedFileRegionToIMemory(const std::optional<SharedFileRegion>& shmem,
+                                              sp<IMemory>* result) {
+    assert(result != nullptr);
+
+    if (!shmem.has_value()) {
+        result->clear();
+        return true;
+    }
+
+    return convertSharedFileRegionToIMemory(shmem.value(), result);
+}
+
+bool convertNullableIMemoryToSharedFileRegion(const sp<IMemory>& mem,
+                                              std::optional<SharedFileRegion>* result) {
+    assert(result != nullptr);
+
+    if (mem == nullptr) {
+        result->reset();
+        return true;
+    }
+
+    result->emplace();
+    return convertIMemoryToSharedFileRegion(mem, &result->value());
+}
+
 }  // namespace media
 }  // namespace android
diff --git a/media/libshmem/ShmemTest.cpp b/media/libshmem/ShmemTest.cpp
index 4f11b51..d076ad0 100644
--- a/media/libshmem/ShmemTest.cpp
+++ b/media/libshmem/ShmemTest.cpp
@@ -17,6 +17,7 @@
 
 #include "binder/MemoryBase.h"
 #include "binder/MemoryHeapBase.h"
+#include "cutils/ashmem.h"
 #include "media/ShmemCompat.h"
 #include "media/ShmemUtil.h"
 
@@ -24,11 +25,22 @@
 namespace media {
 namespace {
 
-// Creates a SharedFileRegion instance with a null FD.
+// Creates a SharedFileRegion instance.
 SharedFileRegion makeSharedFileRegion(int64_t offset, int64_t size) {
     SharedFileRegion shmem;
     shmem.offset = offset;
     shmem.size = size;
+    int fd = ashmem_create_region("", size + offset);
+    assert(fd >= 0);
+    shmem.fd = os::ParcelFileDescriptor(base::unique_fd(fd));
+    return shmem;
+}
+
+// Creates a SharedFileRegion instance with an invalid FD.
+SharedFileRegion makeInvalidSharedFileRegion(int64_t offset, int64_t size) {
+    SharedFileRegion shmem;
+    shmem.offset = offset;
+    shmem.size = size;
     return shmem;
 }
 
@@ -46,9 +58,7 @@
     EXPECT_TRUE(validateSharedFileRegion(makeSharedFileRegion(1, 2)));
     EXPECT_FALSE(validateSharedFileRegion(makeSharedFileRegion(-1, 2)));
     EXPECT_FALSE(validateSharedFileRegion(makeSharedFileRegion(2, -1)));
-    EXPECT_TRUE(validateSharedFileRegion(makeSharedFileRegion(
-            std::numeric_limits<int64_t>::max(),
-            std::numeric_limits<int64_t>::max())));
+    EXPECT_FALSE(validateSharedFileRegion(makeInvalidSharedFileRegion(1, 2)));
 }
 
 TEST(ShmemTest, Conversion) {
@@ -72,12 +82,11 @@
 TEST(ShmemTest, NullConversion) {
     sp<IMemory> reconstructed;
     {
-        SharedFileRegion shmem;
+        std::optional<SharedFileRegion> shmem;
         sp<IMemory> imem;
-        ASSERT_TRUE(convertIMemoryToSharedFileRegion(imem, &shmem));
-        ASSERT_EQ(0, shmem.size);
-        ASSERT_LT(shmem.fd.get(), 0);
-        ASSERT_TRUE(convertSharedFileRegionToIMemory(shmem, &reconstructed));
+        ASSERT_TRUE(convertNullableIMemoryToSharedFileRegion(imem, &shmem));
+        ASSERT_FALSE(shmem.has_value());
+        ASSERT_TRUE(convertNullableSharedFileRegionToIMemory(shmem, &reconstructed));
     }
     ASSERT_EQ(nullptr, reconstructed);
 }
diff --git a/media/libshmem/ShmemUtil.cpp b/media/libshmem/ShmemUtil.cpp
index a6d047f..e075346 100644
--- a/media/libshmem/ShmemUtil.cpp
+++ b/media/libshmem/ShmemUtil.cpp
@@ -19,6 +19,11 @@
 namespace media {
 
 bool validateSharedFileRegion(const SharedFileRegion& shmem) {
+    // FD must be valid.
+    if (shmem.fd.get() < 0) {
+        return false;
+    }
+
     // Size and offset must be non-negative.
     if (shmem.size < 0 || shmem.offset < 0) {
         return false;
diff --git a/media/libshmem/aidl/android/media/SharedFileRegion.aidl b/media/libshmem/aidl/android/media/SharedFileRegion.aidl
index c99ad95..a910e69 100644
--- a/media/libshmem/aidl/android/media/SharedFileRegion.aidl
+++ b/media/libshmem/aidl/android/media/SharedFileRegion.aidl
@@ -20,13 +20,15 @@
  * A shared file region.
  *
  * This type contains the required information to share a region of a file between processes over
- * AIDL. An invalid (null) region may be represented using a negative file descriptor.
+ * AIDL.
+ * An instance of this type represents a valid FD. For representing a null SharedFileRegion, use a
+ * @nullable SharedFileRegion.
  * Primarily, this is intended for shared memory blocks.
  *
  * @hide
  */
 parcelable SharedFileRegion {
-    /** File descriptor of the region. */
+    /** File descriptor of the region. Must be valid. */
     ParcelFileDescriptor fd;
     /** Offset, in bytes within the file of the start of the region. Must be non-negative. */
     long offset;
diff --git a/media/libshmem/include/media/ShmemCompat.h b/media/libshmem/include/media/ShmemCompat.h
index 3bf7f67..ba59f25 100644
--- a/media/libshmem/include/media/ShmemCompat.h
+++ b/media/libshmem/include/media/ShmemCompat.h
@@ -19,6 +19,8 @@
 // This module contains utilities for interfacing between legacy code that is using IMemory and new
 // code that is using android.os.SharedFileRegion.
 
+#include <optional>
+
 #include "android/media/SharedFileRegion.h"
 #include "binder/IMemory.h"
 #include "utils/StrongPointer.h"
@@ -29,8 +31,7 @@
 /**
  * Converts a SharedFileRegion parcelable to an IMemory instance.
  * @param shmem The SharedFileRegion instance.
- * @param result The resulting IMemory instance, or null of the SharedFileRegion is null (has a
- *               negative FD).
+ * @param result The resulting IMemory instance. May not be null.
  * @return true if the conversion is successful (should always succeed under normal circumstances,
  *         failure usually means corrupt data).
  */
@@ -38,8 +39,19 @@
                                       sp<IMemory>* result);
 
 /**
+ * Converts a nullable SharedFileRegion parcelable to an IMemory instance.
+ * @param shmem The SharedFileRegion instance.
+ * @param result The resulting IMemory instance. May not be null. Pointee assigned to null,
+ *               if the input is null.
+ * @return true if the conversion is successful (should always succeed under normal circumstances,
+ *         failure usually means corrupt data).
+ */
+bool convertNullableSharedFileRegionToIMemory(const std::optional<SharedFileRegion>& shmem,
+                                              sp<IMemory>* result);
+
+/**
  * Converts an IMemory instance to SharedFileRegion.
- * @param mem The IMemory instance. May be null.
+ * @param mem The IMemory instance. May not be null.
  * @param result The resulting SharedFileRegion instance.
  * @return true if the conversion is successful (should always succeed under normal circumstances,
  *         failure usually means corrupt data).
@@ -47,5 +59,16 @@
 bool convertIMemoryToSharedFileRegion(const sp<IMemory>& mem,
                                       SharedFileRegion* result);
 
+/**
+ * Converts a nullable IMemory instance to a nullable SharedFileRegion.
+ * @param mem The IMemory instance. May be null.
+ * @param result The resulting SharedFileRegion instance. May not be null. Assigned to empty,
+ *               if the input is null.
+ * @return true if the conversion is successful (should always succeed under normal circumstances,
+ *         failure usually means corrupt data).
+ */
+bool convertNullableIMemoryToSharedFileRegion(const sp<IMemory>& mem,
+                                              std::optional<SharedFileRegion>* result);
+
 }  // namespace media
 }  // namespace android
diff --git a/media/libshmem/include/media/ShmemUtil.h b/media/libshmem/include/media/ShmemUtil.h
index 563cb71..3a7a5a5 100644
--- a/media/libshmem/include/media/ShmemUtil.h
+++ b/media/libshmem/include/media/ShmemUtil.h
@@ -25,7 +25,6 @@
 
 /**
  * Checks whether a SharedFileRegion instance is valid (all the fields have sane values).
- * A null SharedFileRegion (having a negative FD) is considered valid.
  */
 bool validateSharedFileRegion(const SharedFileRegion& shmem);
 
diff --git a/services/audioflinger/Android.bp b/services/audioflinger/Android.bp
index 12f6eba..261af5a 100644
--- a/services/audioflinger/Android.bp
+++ b/services/audioflinger/Android.bp
@@ -35,6 +35,9 @@
     ],
 
     shared_libs: [
+        "audioflinger-aidl-unstable-cpp",
+        "audioclient-types-aidl-unstable-cpp",
+        "libaudioclient_aidl_conversion",
         "libaudiofoundation",
         "libaudiohal",
         "libaudioprocessing",
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index eae9437..e589eb9 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -22,6 +22,13 @@
 // Define AUDIO_ARRAYS_STATIC_CHECK to check all audio arrays are correct
 #define AUDIO_ARRAYS_STATIC_CHECK 1
 
+#define VALUE_OR_FATAL(result) \
+    ({ auto _tmp = (result); \
+       LOG_ALWAYS_FATAL_IF(!_tmp.ok(), \
+                           "Failed result (%d)", \
+                           _tmp.error()); \
+       _tmp.value(); })
+
 #include "Configuration.h"
 #include <dirent.h>
 #include <math.h>
@@ -68,6 +75,7 @@
 #include <powermanager/PowerManager.h>
 
 #include <media/IMediaLogService.h>
+#include <media/AidlConversion.h>
 #include <media/nbaio/Pipe.h>
 #include <media/nbaio/PipeReader.h>
 #include <mediautils/BatteryNotifier.h>
@@ -1774,7 +1782,7 @@
     return BAD_VALUE;
 }
 
-void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
+void AudioFlinger::registerClient(const sp<media::IAudioFlingerClient>& client)
 {
     Mutex::Autolock _l(mLock);
     if (client == 0) {
@@ -1849,13 +1857,18 @@
 
 void AudioFlinger::ioConfigChanged(audio_io_config_event event,
                                    const sp<AudioIoDescriptor>& ioDesc,
-                                   pid_t pid)
-{
+                                   pid_t pid) {
+    media::AudioIoDescriptor descAidl = VALUE_OR_FATAL(
+            legacy2aidl_AudioIoDescriptor_AudioIoDescriptor(ioDesc));
+    media::AudioIoConfigEvent eventAidl = VALUE_OR_FATAL(
+            legacy2aidl_audio_io_config_event_AudioIoConfigEvent(event));
+
     Mutex::Autolock _l(mClientLock);
     size_t size = mNotificationClients.size();
     for (size_t i = 0; i < size; i++) {
         if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) {
-            mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc);
+            mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(eventAidl,
+                                                                                   descAidl);
         }
     }
 }
@@ -1929,7 +1942,7 @@
 // ----------------------------------------------------------------------------
 
 AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
-                                                     const sp<IAudioFlingerClient>& client,
+                                                     const sp<media::IAudioFlingerClient>& client,
                                                      pid_t pid,
                                                      uid_t uid)
     : mAudioFlinger(audioFlinger), mPid(pid), mUid(uid), mAudioFlingerClient(client)
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 14a4df7..65d672a 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -33,16 +33,16 @@
 #include <sys/types.h>
 #include <limits.h>
 
+#include <android/media/IAudioFlingerClient.h>
 #include <android/media/IAudioTrackCallback.h>
 #include <android/os/BnExternalVibrationController.h>
-#include <android-base/macros.h>
 
+#include <android-base/macros.h>
 #include <cutils/atomic.h>
 #include <cutils/compiler.h>
-#include <cutils/properties.h>
 
+#include <cutils/properties.h>
 #include <media/IAudioFlinger.h>
-#include <media/IAudioFlingerClient.h>
 #include <media/IAudioTrack.h>
 #include <media/AudioSystem.h>
 #include <media/AudioTrack.h>
@@ -177,7 +177,7 @@
     virtual     status_t    setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
     virtual     String8     getParameters(audio_io_handle_t ioHandle, const String8& keys) const;
 
-    virtual     void        registerClient(const sp<IAudioFlingerClient>& client);
+    virtual     void        registerClient(const sp<media::IAudioFlingerClient>& client);
 
     virtual     size_t      getInputBufferSize(uint32_t sampleRate, audio_format_t format,
                                                audio_channel_mask_t channelMask) const;
@@ -490,12 +490,12 @@
     class NotificationClient : public IBinder::DeathRecipient {
     public:
                             NotificationClient(const sp<AudioFlinger>& audioFlinger,
-                                                const sp<IAudioFlingerClient>& client,
+                                                const sp<media::IAudioFlingerClient>& client,
                                                 pid_t pid,
                                                 uid_t uid);
         virtual             ~NotificationClient();
 
-                sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; }
+                sp<media::IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; }
                 pid_t getPid() const { return mPid; }
                 uid_t getUid() const { return mUid; }
 
@@ -508,7 +508,7 @@
         const sp<AudioFlinger>  mAudioFlinger;
         const pid_t             mPid;
         const uid_t             mUid;
-        const sp<IAudioFlingerClient> mAudioFlingerClient;
+        const sp<media::IAudioFlingerClient> mAudioFlingerClient;
     };
 
     // --- MediaLogNotifier ---
diff --git a/services/oboeservice/AAudioEndpointManager.cpp b/services/oboeservice/AAudioEndpointManager.cpp
index 9f34153..483a264 100644
--- a/services/oboeservice/AAudioEndpointManager.cpp
+++ b/services/oboeservice/AAudioEndpointManager.cpp
@@ -306,6 +306,7 @@
                 mSharedStreams.end());
 
         serviceEndpoint->close();
+
         mSharedCloseCount++;
         ALOGV("%s(%p) closed for device %d",
               __func__, serviceEndpoint.get(), serviceEndpoint->getDeviceId());
diff --git a/services/oboeservice/AAudioServiceEndpointShared.cpp b/services/oboeservice/AAudioServiceEndpointShared.cpp
index f5de59f..caf6139 100644
--- a/services/oboeservice/AAudioServiceEndpointShared.cpp
+++ b/services/oboeservice/AAudioServiceEndpointShared.cpp
@@ -88,23 +88,30 @@
 }
 
 void AAudioServiceEndpointShared::close() {
-    getStreamInternal()->releaseCloseFinal();
+    stopSharingThread();
+    getStreamInternal()->safeReleaseClose();
 }
 
 // Glue between C and C++ callbacks.
 static void *aaudio_endpoint_thread_proc(void *arg) {
     assert(arg != nullptr);
+    ALOGD("%s() called", __func__);
 
-    // The caller passed in a smart pointer to prevent the endpoint from getting deleted
-    // while the thread was launching.
-    sp<AAudioServiceEndpointShared> *endpointForThread =
-            static_cast<sp<AAudioServiceEndpointShared> *>(arg);
-    sp<AAudioServiceEndpointShared> endpoint = *endpointForThread;
-    delete endpointForThread; // Just use scoped smart pointer. Don't need this anymore.
+    // Prevent the stream from being deleted while being used.
+    // This is just for extra safety. It is probably not needed because
+    // this callback should be joined before the stream is closed.
+    AAudioServiceEndpointShared *endpointPtr =
+        static_cast<AAudioServiceEndpointShared *>(arg);
+    android::sp<AAudioServiceEndpointShared> endpoint(endpointPtr);
+    // Balance the incStrong() in startSharingThread_l().
+    endpoint->decStrong(nullptr);
+
     void *result = endpoint->callbackLoop();
     // Close now so that the HW resource is freed and we can open a new device.
     if (!endpoint->isConnected()) {
-        endpoint->close();
+        ALOGD("%s() call safeReleaseCloseFromCallback()", __func__);
+        // Release and close under a lock with no check for callback collisions.
+        endpoint->getStreamInternal()->safeReleaseCloseFromCallback();
     }
 
     return result;
@@ -116,14 +123,14 @@
                           * AAUDIO_NANOS_PER_SECOND
                           / getSampleRate();
     mCallbackEnabled.store(true);
-    // Pass a smart pointer so the thread can hold a reference.
-    sp<AAudioServiceEndpointShared> *endpointForThread = new sp<AAudioServiceEndpointShared>(this);
-    aaudio_result_t result = getStreamInternal()->createThread(periodNanos,
-                                                               aaudio_endpoint_thread_proc,
-                                                               endpointForThread);
+    // Prevent this object from getting deleted before the thread has a chance to create
+    // its strong pointer. Assume the thread will call decStrong().
+    this->incStrong(nullptr);
+    aaudio_result_t result = getStreamInternal()->createThread_l(periodNanos,
+                                                                 aaudio_endpoint_thread_proc,
+                                                                 this);
     if (result != AAUDIO_OK) {
-        // The thread can't delete it so we have to do it here.
-        delete endpointForThread;
+        this->decStrong(nullptr); // Because the thread won't do it.
     }
     return result;
 }
@@ -141,13 +148,13 @@
     {
         std::lock_guard<std::mutex> lock(mLockStreams);
         if (++mRunningStreamCount == 1) { // atomic
-            result = getStreamInternal()->requestStart();
+            result = getStreamInternal()->requestStart_l();
             if (result != AAUDIO_OK) {
                 --mRunningStreamCount;
             } else {
                 result = startSharingThread_l();
                 if (result != AAUDIO_OK) {
-                    getStreamInternal()->requestStop();
+                    getStreamInternal()->requestStop_l();
                     --mRunningStreamCount;
                 }
             }
@@ -161,7 +168,7 @@
         if (result != AAUDIO_OK) {
             if (--mRunningStreamCount == 0) { // atomic
                 stopSharingThread();
-                getStreamInternal()->requestStop();
+                getStreamInternal()->requestStop_l();
             }
         }
     }
@@ -176,7 +183,7 @@
 
     if (--mRunningStreamCount == 0) { // atomic
         stopSharingThread(); // the sharing thread locks mLockStreams
-        getStreamInternal()->requestStop();
+        getStreamInternal()->requestStop_l();
     }
     return AAUDIO_OK;
 }
diff --git a/services/oboeservice/AAudioServiceEndpointShared.h b/services/oboeservice/AAudioServiceEndpointShared.h
index 020b926..91a86c1 100644
--- a/services/oboeservice/AAudioServiceEndpointShared.h
+++ b/services/oboeservice/AAudioServiceEndpointShared.h
@@ -37,6 +37,8 @@
 public:
     explicit AAudioServiceEndpointShared(AudioStreamInternal *streamInternal);
 
+    virtual ~AAudioServiceEndpointShared() = default;
+
     std::string dump() const override;
 
     aaudio_result_t open(const aaudio::AAudioStreamRequest &request) override;
@@ -55,12 +57,12 @@
 
     virtual void   *callbackLoop() = 0;
 
-protected:
-
     AudioStreamInternal *getStreamInternal() const {
         return mStreamInternal.get();
     };
 
+protected:
+
     aaudio_result_t          startSharingThread_l();
 
     aaudio_result_t          stopSharingThread();
diff --git a/services/oboeservice/AAudioThread.cpp b/services/oboeservice/AAudioThread.cpp
index ed7895b..68496ac 100644
--- a/services/oboeservice/AAudioThread.cpp
+++ b/services/oboeservice/AAudioThread.cpp
@@ -37,10 +37,13 @@
     setup("AAudio");
 }
 
-void AAudioThread::setup(const char *prefix) {
-    // mThread is a pthread_t of unknown size so we need memset().
-    memset(&mThread, 0, sizeof(mThread));
+AAudioThread::~AAudioThread() {
+    ALOGE_IF(pthread_equal(pthread_self(), mThread),
+            "%s() destructor running in thread", __func__);
+    ALOGE_IF(mHasThread, "%s() thread never joined", __func__);
+}
 
+void AAudioThread::setup(const char *prefix) {
     // Name the thread with an increasing index, "prefix_#", for debugging.
     uint32_t index = mNextThreadIndex++;
     // Wrap the index so that we do not hit the 16 char limit
@@ -57,7 +60,7 @@
     }
 }
 
-// This is the entry point for the new thread created by createThread().
+// This is the entry point for the new thread created by createThread_l().
 // It converts the 'C' function call to a C++ method call.
 static void * AAudioThread_internalThreadProc(void *arg) {
     AAudioThread *aaudioThread = (AAudioThread *) arg;
@@ -90,13 +93,18 @@
         ALOGE("stop() but no thread running");
         return AAUDIO_ERROR_INVALID_STATE;
     }
+    // Check to see if the thread is trying to stop itself.
+    if (pthread_equal(pthread_self(), mThread)) {
+        ALOGE("%s() attempt to pthread_join() from launched thread!", __func__);
+        return AAUDIO_ERROR_INTERNAL;
+    }
+
     int err = pthread_join(mThread, nullptr);
-    mHasThread = false;
     if (err != 0) {
         ALOGE("stop() - pthread_join() returned %d %s", err, strerror(err));
         return AAudioConvert_androidToAAudioResult(-err);
     } else {
+        mHasThread = false;
         return AAUDIO_OK;
     }
 }
-
diff --git a/services/oboeservice/AAudioThread.h b/services/oboeservice/AAudioThread.h
index dcce68a..08a8a98 100644
--- a/services/oboeservice/AAudioThread.h
+++ b/services/oboeservice/AAudioThread.h
@@ -46,7 +46,7 @@
 
     explicit AAudioThread(const char *prefix);
 
-    virtual ~AAudioThread() = default;
+    virtual ~AAudioThread();
 
     /**
      * Start the thread running.
@@ -73,7 +73,7 @@
 
     Runnable    *mRunnable = nullptr;
     bool         mHasThread = false;
-    pthread_t    mThread; // initialized in constructor
+    pthread_t    mThread = {};
 
     static std::atomic<uint32_t> mNextThreadIndex;
     char         mName[16]; // max length for a pthread_name