diff --git a/drm/libmediadrm/DrmHal.cpp b/drm/libmediadrm/DrmHal.cpp
index f4c0341..b72348f 100644
--- a/drm/libmediadrm/DrmHal.cpp
+++ b/drm/libmediadrm/DrmHal.cpp
@@ -63,6 +63,7 @@
 
 typedef drm::V1_1::KeyRequestType KeyRequestType_V1_1;
 typedef drm::V1_2::Status Status_V1_2;
+typedef drm::V1_2::HdcpLevel HdcpLevel_V1_2;
 
 namespace {
 
@@ -156,26 +157,26 @@
     }
 }
 
-static DrmPlugin::HdcpLevel toHdcpLevel(HdcpLevel level) {
+static DrmPlugin::HdcpLevel toHdcpLevel(HdcpLevel_V1_2 level) {
     switch(level) {
-    case HdcpLevel::HDCP_NONE:
+    case HdcpLevel_V1_2::HDCP_NONE:
         return DrmPlugin::kHdcpNone;
-    case HdcpLevel::HDCP_V1:
+    case HdcpLevel_V1_2::HDCP_V1:
         return DrmPlugin::kHdcpV1;
-    case HdcpLevel::HDCP_V2:
+    case HdcpLevel_V1_2::HDCP_V2:
         return DrmPlugin::kHdcpV2;
-    case HdcpLevel::HDCP_V2_1:
+    case HdcpLevel_V1_2::HDCP_V2_1:
         return DrmPlugin::kHdcpV2_1;
-    case HdcpLevel::HDCP_V2_2:
+    case HdcpLevel_V1_2::HDCP_V2_2:
         return DrmPlugin::kHdcpV2_2;
-    case HdcpLevel::HDCP_NO_OUTPUT:
+    case HdcpLevel_V1_2::HDCP_V2_3:
+        return DrmPlugin::kHdcpV2_3;
+    case HdcpLevel_V1_2::HDCP_NO_OUTPUT:
         return DrmPlugin::kHdcpNoOutput;
     default:
         return DrmPlugin::kHdcpLevelUnknown;
     }
 }
-
-
 static ::KeyedVector toHidlKeyedVector(const KeyedVector<String8, String8>&
         keyedVector) {
     std::vector<KeyValue> stdKeyedVector;
@@ -1093,22 +1094,31 @@
     }
     status_t err = UNKNOWN_ERROR;
 
-    if (mPluginV1_1 == NULL) {
-        return ERROR_DRM_CANNOT_HANDLE;
-    }
-
     *connected = DrmPlugin::kHdcpLevelUnknown;
     *max = DrmPlugin::kHdcpLevelUnknown;
 
-    Return<void> hResult = mPluginV1_1->getHdcpLevels(
-            [&](Status status, const HdcpLevel& hConnected, const HdcpLevel& hMax) {
-                if (status == Status::OK) {
-                    *connected = toHdcpLevel(hConnected);
-                    *max = toHdcpLevel(hMax);
-                }
-                err = toStatusT(status);
-            }
-    );
+    Return<void> hResult;
+    if (mPluginV1_2 != NULL) {
+        hResult = mPluginV1_2->getHdcpLevels_1_2(
+                [&](Status_V1_2 status, const HdcpLevel_V1_2& hConnected, const HdcpLevel_V1_2& hMax) {
+                    if (status == Status_V1_2::OK) {
+                        *connected = toHdcpLevel(hConnected);
+                        *max = toHdcpLevel(hMax);
+                    }
+                    err = toStatusT_1_2(status);
+                });
+    } else if (mPluginV1_1 != NULL) {
+        hResult = mPluginV1_1->getHdcpLevels(
+                [&](Status status, const HdcpLevel& hConnected, const HdcpLevel& hMax) {
+                    if (status == Status::OK) {
+                        *connected = toHdcpLevel(static_cast<HdcpLevel_V1_2>(hConnected));
+                        *max = toHdcpLevel(static_cast<HdcpLevel_V1_2>(hMax));
+                    }
+                    err = toStatusT(status);
+                });
+    } else {
+        return ERROR_DRM_CANNOT_HANDLE;
+    }
 
     return hResult.isOk() ? err : DEAD_OBJECT;
 }
diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h b/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h
index a9b897b..ba5fa65 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h
@@ -63,6 +63,7 @@
 typedef drm::V1_1::KeyRequestType KeyRequestType_V1_1;
 typedef drm::V1_2::IDrmPluginListener IDrmPluginListener_V1_2;
 typedef drm::V1_2::Status Status_V1_2;
+typedef drm::V1_2::HdcpLevel HdcpLevel_V1_2;
 
 struct DrmPlugin : public IDrmPlugin {
     explicit DrmPlugin(SessionLibrary* sessionLibrary);
@@ -162,6 +163,13 @@
         return Void();
     }
 
+    Return<void> getHdcpLevels_1_2(getHdcpLevels_1_2_cb _hidl_cb) {
+        HdcpLevel_V1_2 connectedLevel = HdcpLevel_V1_2::HDCP_NONE;
+        HdcpLevel_V1_2 maxLevel = HdcpLevel_V1_2::HDCP_NO_OUTPUT;
+        _hidl_cb(Status_V1_2::OK, connectedLevel, maxLevel);
+        return Void();
+    }
+
     Return<void> getNumberOfSessions(getNumberOfSessions_cb _hidl_cb) override;
 
     Return<void> getSecurityLevel(const hidl_vec<uint8_t>& sessionId,
diff --git a/include/media/MediaExtractorPluginHelper.h b/include/media/MediaExtractorPluginHelper.h
index f4d4da6..b86f177 100644
--- a/include/media/MediaExtractorPluginHelper.h
+++ b/include/media/MediaExtractorPluginHelper.h
@@ -171,6 +171,9 @@
 };
 
 inline CMediaTrack *wrap(MediaTrackHelper *track) {
+    if (track == nullptr) {
+        return nullptr;
+    }
     CMediaTrack *wrapper = (CMediaTrack*) malloc(sizeof(CMediaTrack));
     wrapper->data = track;
     wrapper->free = [](void *data) -> void {
diff --git a/include/media/MediaTrack.h b/include/media/MediaTrack.h
index e828a7f..493eba3 100644
--- a/include/media/MediaTrack.h
+++ b/include/media/MediaTrack.h
@@ -142,7 +142,7 @@
 
 class MediaTrackCUnwrapper : public MediaTrack {
 public:
-    explicit MediaTrackCUnwrapper(CMediaTrack *wrapper);
+    static MediaTrackCUnwrapper *create(CMediaTrack *wrapper);
 
     virtual status_t start();
     virtual status_t stop();
@@ -155,6 +155,7 @@
     virtual ~MediaTrackCUnwrapper();
 
 private:
+    explicit MediaTrackCUnwrapper(CMediaTrack *wrapper);
     CMediaTrack *wrapper;
     MediaBufferGroup *bufferGroup;
 };
diff --git a/media/extractors/mp4/MPEG4Extractor.cpp b/media/extractors/mp4/MPEG4Extractor.cpp
index 0441359..cc1534a 100644
--- a/media/extractors/mp4/MPEG4Extractor.cpp
+++ b/media/extractors/mp4/MPEG4Extractor.cpp
@@ -985,6 +985,22 @@
             off64_t stop_offset = *offset + chunk_size;
             *offset = data_offset;
             while (*offset < stop_offset) {
+
+                // pass udata terminate
+                if (mIsQT && stop_offset - *offset == 4 && chunk_type == FOURCC("udta")) {
+                    // handle the case that udta terminates with terminate code x00000000
+                    // note that 0 terminator is optional and we just handle this case.
+                    uint32_t terminate_code = 1;
+                    mDataSource->readAt(*offset, &terminate_code, 4);
+                    if (0 == terminate_code) {
+                        *offset += 4;
+                        ALOGD("Terminal code for udta");
+                        continue;
+                    } else {
+                        ALOGW("invalid udta Terminal code");
+                    }
+                }
+
                 status_t err = parseChunk(offset, depth + 1);
                 if (err != OK) {
                     if (isTrack) {
@@ -5142,7 +5158,7 @@
         sampleCtsOffset = 0;
     }
 
-    if (size < (off64_t)(sampleCount * bytesPerSample)) {
+    if (size < (off64_t)sampleCount * bytesPerSample) {
         return -EINVAL;
     }
 
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index fffcda0..3b03601 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -62,7 +62,7 @@
         , mServiceStreamHandle(AAUDIO_HANDLE_INVALID)
         , mInService(inService)
         , mServiceInterface(serviceInterface)
-        , mAtomicTimestamp()
+        , mAtomicInternalTimestamp()
         , mWakeupDelayNanos(AAudioProperty_getWakeupDelayMicros() * AAUDIO_NANOS_PER_MICROSECOND)
         , mMinimumSleepNanos(AAudioProperty_getMinimumSleepMicros() * AAUDIO_NANOS_PER_MICROSECOND)
         {
@@ -349,8 +349,7 @@
     }
 }
 
-aaudio_result_t AudioStreamInternal::requestStop()
-{
+aaudio_result_t AudioStreamInternal::requestStop() {
     aaudio_result_t result = stopCallback();
     if (result != AAUDIO_OK) {
         return result;
@@ -364,7 +363,7 @@
 
     mClockModel.stop(AudioClock::getNanoseconds());
     setState(AAUDIO_STREAM_STATE_STOPPING);
-    mAtomicTimestamp.clear();
+    mAtomicInternalTimestamp.clear();
 
     return mServiceInterface.stopStream(mServiceStreamHandle);
 }
@@ -413,8 +412,8 @@
                            int64_t *framePosition,
                            int64_t *timeNanoseconds) {
     // Generated in server and passed to client. Return latest.
-    if (mAtomicTimestamp.isValid()) {
-        Timestamp timestamp = mAtomicTimestamp.read();
+    if (mAtomicInternalTimestamp.isValid()) {
+        Timestamp timestamp = mAtomicInternalTimestamp.read();
         int64_t position = timestamp.getPosition() + mFramesOffsetFromService;
         if (position >= 0) {
             *framePosition = position;
@@ -461,7 +460,7 @@
 
 aaudio_result_t AudioStreamInternal::onTimestampHardware(AAudioServiceMessage *message) {
     Timestamp timestamp(message->timestamp.position, message->timestamp.timestamp);
-    mAtomicTimestamp.write(timestamp);
+    mAtomicInternalTimestamp.write(timestamp);
     return AAUDIO_OK;
 }
 
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 3bb9e1e..1c88f52 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -163,7 +163,7 @@
 
     AAudioServiceInterface  &mServiceInterface;   // abstract interface to the service
 
-    SimpleDoubleBuffer<Timestamp>  mAtomicTimestamp;
+    SimpleDoubleBuffer<Timestamp>  mAtomicInternalTimestamp;
 
     AtomicRequestor          mNeedCatchUp;   // Ask read() or write() to sync on first timestamp.
 
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index 58ef7b1..7dcb620 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -259,6 +259,7 @@
 
         if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
             ALOGD("%s(): callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);
+            result = systemStopFromCallback();
             break;
         }
     }
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index 9af47b2..6af8e7d 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -71,7 +71,7 @@
 
     mClockModel.stop(AudioClock::getNanoseconds());
     setState(AAUDIO_STREAM_STATE_PAUSING);
-    mAtomicTimestamp.clear();
+    mAtomicInternalTimestamp.clear();
     return mServiceInterface.pauseStream(mServiceStreamHandle);
 }
 
@@ -294,6 +294,7 @@
             }
         } else if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
             ALOGD("%s(): callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);
+            result = systemStopFromCallback();
             break;
         }
     }
diff --git a/media/libaaudio/src/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index 2fb3986..0d71efc 100644
--- a/media/libaaudio/src/core/AAudioAudio.cpp
+++ b/media/libaaudio/src/core/AAudioAudio.cpp
@@ -316,7 +316,7 @@
 {
     AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
     ALOGD("%s(%p) called", __func__, stream);
-    return audioStream->systemStop();
+    return audioStream->systemStopFromApp();
 }
 
 AAUDIO_API aaudio_result_t AAudioStream_waitForStateChange(AAudioStream* stream,
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index 391af29..e39a075 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -119,21 +119,29 @@
     return AAUDIO_OK;
 }
 
-aaudio_result_t AudioStream::safeStart() {
+aaudio_result_t AudioStream::systemStart() {
     std::lock_guard<std::mutex> lock(mStreamLock);
+
     if (collidesWithCallback()) {
         ALOGE("%s cannot be called from a callback!", __func__);
         return AAUDIO_ERROR_INVALID_STATE;
     }
-    return requestStart();
+
+    aaudio_result_t result = requestStart();
+    if (result == AAUDIO_OK) {
+        // We only call this for logging in "dumpsys audio". So ignore return code.
+        (void) mPlayerBase->start();
+    }
+    return result;
 }
 
-aaudio_result_t AudioStream::safePause() {
+aaudio_result_t AudioStream::systemPause() {
+    std::lock_guard<std::mutex> lock(mStreamLock);
+
     if (!isPauseSupported()) {
         return AAUDIO_ERROR_UNIMPLEMENTED;
     }
 
-    std::lock_guard<std::mutex> lock(mStreamLock);
     if (collidesWithCallback()) {
         ALOGE("%s cannot be called from a callback!", __func__);
         return AAUDIO_ERROR_INVALID_STATE;
@@ -169,7 +177,12 @@
             return AAUDIO_ERROR_INVALID_STATE;
     }
 
-    return requestPause();
+    aaudio_result_t result = requestPause();
+    if (result == AAUDIO_OK) {
+        // We only call this for logging in "dumpsys audio". So ignore return code.
+        (void) mPlayerBase->pause();
+    }
+    return result;
 }
 
 aaudio_result_t AudioStream::safeFlush() {
@@ -192,12 +205,31 @@
     return requestFlush();
 }
 
-aaudio_result_t AudioStream::safeStop() {
+aaudio_result_t AudioStream::systemStopFromCallback() {
+    std::lock_guard<std::mutex> lock(mStreamLock);
+    aaudio_result_t result = safeStop();
+    if (result == AAUDIO_OK) {
+        // We only call this for logging in "dumpsys audio". So ignore return code.
+        (void) mPlayerBase->stop();
+    }
+    return result;
+}
+
+aaudio_result_t AudioStream::systemStopFromApp() {
     std::lock_guard<std::mutex> lock(mStreamLock);
     if (collidesWithCallback()) {
-        ALOGE("stream cannot be stopped from a callback!");
+        ALOGE("stream cannot be stopped by calling from a callback!");
         return AAUDIO_ERROR_INVALID_STATE;
     }
+    aaudio_result_t result = safeStop();
+    if (result == AAUDIO_OK) {
+        // We only call this for logging in "dumpsys audio". So ignore return code.
+        (void) mPlayerBase->stop();
+    }
+    return result;
+}
+
+aaudio_result_t AudioStream::safeStop() {
 
     switch (getState()) {
         // Proceed with stopping.
@@ -224,7 +256,7 @@
         case AAUDIO_STREAM_STATE_CLOSING:
         case AAUDIO_STREAM_STATE_CLOSED:
         default:
-            ALOGW("requestStop() stream not running, state = %s",
+            ALOGW("%s() stream not running, state = %s", __func__,
                   AAudio_convertStreamStateToText(getState()));
             return AAUDIO_ERROR_INVALID_STATE;
     }
@@ -349,21 +381,33 @@
     }
 }
 
-aaudio_result_t AudioStream::joinThread(void** returnArg, int64_t timeoutNanoseconds)
+aaudio_result_t AudioStream::joinThread(void** returnArg, int64_t timeoutNanoseconds __unused)
 {
     if (!mHasThread) {
         ALOGE("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) {
+        // Called from an app thread. Not the callback.
 #if 0
-    // TODO implement equivalent of pthread_timedjoin_np()
-    struct timespec abstime;
-    int err = pthread_timedjoin_np(mThread, returnArg, &abstime);
+        // TODO implement equivalent of pthread_timedjoin_np()
+        struct timespec abstime;
+        int err = pthread_timedjoin_np(mThread, returnArg, &abstime);
 #else
-    int err = pthread_join(mThread, returnArg);
+        int err = pthread_join(mThread, returnArg);
 #endif
+        if (err) {
+            ALOGE("%s() pthread_join() returns err = %d", __func__, err);
+            result = AAudioConvert_androidToAAudioResult(-err);
+        }
+    }
+    // This must be set false so that the callback thread can be created
+    // when the stream is restarted.
     mHasThread = false;
-    return err ? AAudioConvert_androidToAAudioResult(-errno) : mThreadRegistrationResult;
+    return (result != AAUDIO_OK) ? result : mThreadRegistrationResult;
 }
 
 aaudio_data_callback_result_t AudioStream::maybeCallDataCallback(void *audioData,
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index 60200b2..46951f5 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -51,21 +51,6 @@
 
     virtual ~AudioStream();
 
-    /**
-     * Lock a mutex and make sure we are not calling from a callback function.
-     * @return result of requestStart();
-     */
-    aaudio_result_t safeStart();
-
-    aaudio_result_t safePause();
-
-    aaudio_result_t safeFlush();
-
-    aaudio_result_t safeStop();
-
-    aaudio_result_t safeClose();
-
-    // =========== Begin ABSTRACT methods ===========================
 protected:
 
     /* Asynchronous requests.
@@ -74,7 +59,7 @@
     virtual aaudio_result_t requestStart() = 0;
 
     /**
-     * Check the state to see if Pause if currently legal.
+     * Check the state to see if Pause is currently legal.
      *
      * @param result pointer to return code
      * @return true if OK to continue, if false then return result
@@ -356,33 +341,28 @@
         mPlayerBase->unregisterWithAudioManager();
     }
 
-    // Pass start request through PlayerBase for tracking.
-    aaudio_result_t systemStart() {
-        mPlayerBase->start();
-        // Pass aaudio_result_t around the PlayerBase interface, which uses status__t.
-        return mPlayerBase->getResult();
-    }
+    aaudio_result_t systemStart();
 
-    // Pass pause request through PlayerBase for tracking.
-    aaudio_result_t systemPause() {
-        mPlayerBase->pause();
-        return mPlayerBase->getResult();
-    }
+    aaudio_result_t systemPause();
 
-    // Pass stop request through PlayerBase for tracking.
-    aaudio_result_t systemStop() {
-        mPlayerBase->stop();
-        return mPlayerBase->getResult();
-    }
+    aaudio_result_t safeFlush();
+
+    /**
+     * This is called when an app calls AAudioStream_requestStop();
+     * It prevents calls from a callback.
+     */
+    aaudio_result_t systemStopFromApp();
+
+    /**
+     * This is called internally when an app callback returns AAUDIO_CALLBACK_RESULT_STOP.
+     */
+    aaudio_result_t systemStopFromCallback();
+
+    aaudio_result_t safeClose();
 
 protected:
 
-    // PlayerBase allows the system to control the stream.
-    // Calling through PlayerBase->start() notifies the AudioManager of the player state.
-    // The AudioManager also can start/stop a stream by calling mPlayerBase->playerStart().
-    // systemStart() ==> mPlayerBase->start()   mPlayerBase->playerStart() ==> requestStart()
-    //                        \                           /
-    //                         ------ AudioManager -------
+    // PlayerBase allows the system to control the stream volume.
     class MyPlayerBase : public android::PlayerBase {
     public:
         explicit MyPlayerBase(AudioStream *parent);
@@ -406,20 +386,19 @@
 
         void clearParentReference() { mParent = nullptr; }
 
+        // Just a stub. The ability to start audio through PlayerBase is being deprecated.
         android::status_t playerStart() override {
-            // mParent should NOT be null. So go ahead and crash if it is.
-            mResult = mParent->safeStart();
-            return AAudioConvert_aaudioToAndroidStatus(mResult);
+            return android::NO_ERROR;
         }
 
+        // Just a stub. The ability to pause audio through PlayerBase is being deprecated.
         android::status_t playerPause() override {
-            mResult = mParent->safePause();
-            return AAudioConvert_aaudioToAndroidStatus(mResult);
+            return android::NO_ERROR;
         }
 
+        // Just a stub. The ability to stop audio through PlayerBase is being deprecated.
         android::status_t playerStop() override {
-            mResult = mParent->safeStop();
-            return AAudioConvert_aaudioToAndroidStatus(mResult);
+            return android::NO_ERROR;
         }
 
         android::status_t playerSetVolume() override {
@@ -548,6 +527,8 @@
 
 private:
 
+    aaudio_result_t safeStop();
+
     std::mutex                 mStreamLock;
 
     const android::sp<MyPlayerBase>   mPlayerBase;
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
index a6b9f5d..2edab58 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
@@ -78,8 +78,9 @@
 
 void AudioStreamLegacy::processCallbackCommon(aaudio_callback_operation_t opcode, void *info) {
     aaudio_data_callback_result_t callbackResult;
-    // This illegal size can be used to tell AudioFlinger to stop calling us.
-    // This takes advantage of AudioFlinger killing the stream.
+    // This illegal size can be used to tell AudioRecord or AudioTrack to stop calling us.
+    // This takes advantage of them killing the stream when they see a size out of range.
+    // That is an undocumented behavior.
     // TODO add to API in AudioRecord and AudioTrack
     const size_t SIZE_STOP_CALLBACKS = SIZE_MAX;
 
@@ -95,7 +96,7 @@
                 ALOGW("processCallbackCommon() data, stream disconnected");
                 audioBuffer->size = SIZE_STOP_CALLBACKS;
             } else if (!mCallbackEnabled.load()) {
-                ALOGW("processCallbackCommon() stopping because callback disabled");
+                ALOGW("processCallbackCommon() no data because callback disabled");
                 audioBuffer->size = SIZE_STOP_CALLBACKS;
             } else {
                 if (audioBuffer->frameCount == 0) {
@@ -115,10 +116,16 @@
                 }
                 if (callbackResult == AAUDIO_CALLBACK_RESULT_CONTINUE) {
                     audioBuffer->size = audioBuffer->frameCount * getBytesPerDeviceFrame();
-                } else { // STOP or invalid result
-                    ALOGW("%s() callback requested stop, fake an error", __func__);
-                    audioBuffer->size = SIZE_STOP_CALLBACKS;
-                    // Disable the callback just in case AudioFlinger keeps trying to call us.
+                } else {
+                    if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
+                        ALOGD("%s() callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);
+                    } else {
+                        ALOGW("%s() callback returned invalid result = %d",
+                              __func__, callbackResult);
+                    }
+                    audioBuffer->size = 0;
+                    systemStopFromCallback();
+                    // Disable the callback just in case the system keeps trying to call us.
                     mCallbackEnabled.store(false);
                 }
 
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index 40e22ac..f550089 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
@@ -486,6 +486,9 @@
                                                int64_t *framePosition,
                                                int64_t *timeNanoseconds) {
     ExtendedTimestamp extendedTimestamp;
+    if (getState() != AAUDIO_STREAM_STATE_STARTED) {
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
     status_t status = mAudioRecord->getTimestamp(&extendedTimestamp);
     if (status == WOULD_BLOCK) {
         return AAUDIO_ERROR_INVALID_STATE;
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 1ac2558..c995e99 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -288,7 +288,7 @@
 
 aaudio_result_t AudioStreamTrack::requestPause() {
     if (mAudioTrack.get() == nullptr) {
-        ALOGE("requestPause() no AudioTrack");
+        ALOGE("%s() no AudioTrack", __func__);
         return AAUDIO_ERROR_INVALID_STATE;
     }
 
@@ -304,7 +304,7 @@
 
 aaudio_result_t AudioStreamTrack::requestFlush() {
     if (mAudioTrack.get() == nullptr) {
-        ALOGE("requestFlush() no AudioTrack");
+        ALOGE("%s() no AudioTrack", __func__);
         return AAUDIO_ERROR_INVALID_STATE;
     }
 
@@ -318,7 +318,7 @@
 
 aaudio_result_t AudioStreamTrack::requestStop() {
     if (mAudioTrack.get() == nullptr) {
-        ALOGE("requestStop() no AudioTrack");
+        ALOGE("%s() no AudioTrack", __func__);
         return AAUDIO_ERROR_INVALID_STATE;
     }
 
diff --git a/media/libaaudio/tests/test_timestamps.cpp b/media/libaaudio/tests/test_timestamps.cpp
index dfa7815..7b1dfd3 100644
--- a/media/libaaudio/tests/test_timestamps.cpp
+++ b/media/libaaudio/tests/test_timestamps.cpp
@@ -35,6 +35,7 @@
 
 #define NUM_SECONDS             1
 #define NUM_LOOPS               4
+#define MAX_TESTS               20
 
 typedef struct TimestampInfo {
     int64_t         framesTotal;
@@ -53,6 +54,49 @@
     bool           forceUnderruns = false;
 } TimestampCallbackData_t;
 
+struct TimeStampTestLog {
+    aaudio_policy_t           isMmap;
+    aaudio_sharing_mode_t     sharingMode;
+    aaudio_performance_mode_t performanceMode;
+    aaudio_direction_t        direction;
+    aaudio_result_t           result;
+};
+
+static int s_numTests = 0;
+// Use a plain old array because we reference this from the callback and do not want any
+// automatic memory allocation.
+static TimeStampTestLog s_testLogs[MAX_TESTS]{};
+
+static void logTestResult(bool isMmap,
+                          aaudio_sharing_mode_t sharingMode,
+                          aaudio_performance_mode_t performanceMode,
+                          aaudio_direction_t direction,
+                          aaudio_result_t result) {
+    if(s_numTests >= MAX_TESTS) {
+        printf("ERROR - MAX_TESTS too small = %d\n", MAX_TESTS);
+        return;
+    }
+    s_testLogs[s_numTests].isMmap = isMmap;
+    s_testLogs[s_numTests].sharingMode = sharingMode;
+    s_testLogs[s_numTests].performanceMode = performanceMode;
+    s_testLogs[s_numTests].direction = direction;
+    s_testLogs[s_numTests].result = result;
+    s_numTests++;
+}
+
+static void printTestResults() {
+    for (int i = 0; i < s_numTests; i++) {
+        TimeStampTestLog *log = &s_testLogs[i];
+        printf("%2d: mmap = %3s, sharing = %9s, perf = %11s, dir = %6s ---- %4s\n",
+               i,
+               log->isMmap ? "yes" : "no",
+               getSharingModeText(log->sharingMode),
+               getPerformanceModeText(log->performanceMode),
+               getDirectionText(log->direction),
+               log->result ? "FAIL" : "pass");
+    }
+}
+
 // Callback function that fills the audio output buffer.
 aaudio_data_callback_result_t timestampDataCallbackProc(
         AAudioStream *stream,
@@ -115,6 +159,7 @@
     int32_t originalBufferSize = 0;
     int32_t requestedBufferSize = 0;
     int32_t finalBufferSize = 0;
+    bool    isMmap = false;
     aaudio_format_t actualDataFormat = AAUDIO_FORMAT_PCM_FLOAT;
     aaudio_sharing_mode_t actualSharingMode = AAUDIO_SHARING_MODE_SHARED;
     aaudio_sharing_mode_t actualPerformanceMode = AAUDIO_PERFORMANCE_MODE_NONE;
@@ -124,7 +169,8 @@
 
     memset(&sTimestampData, 0, sizeof(sTimestampData));
 
-    printf("------------ testTimeStamps(policy = %d, sharing = %s, perf = %s, dir = %s) -----------\n",
+    printf("\n=================================================================================\n");
+    printf("--------- testTimeStamps(policy = %d, sharing = %s, perf = %s, dir = %s) --------\n",
            mmapPolicy,
            getSharingModeText(sharingMode),
            getPerformanceModeText(performanceMode),
@@ -177,8 +223,8 @@
 
     printf("    chans = %3d, rate = %6d format = %d\n",
            actualChannelCount, actualSampleRate, actualDataFormat);
-    printf("    Is MMAP used? %s\n", AAudioStream_isMMapUsed(aaudioStream)
-                                     ? "yes" : "no");
+    isMmap = AAudioStream_isMMapUsed(aaudioStream);
+    printf("    Is MMAP used? %s\n", isMmap ? "yes" : "no");
 
     // This is the number of frames that are read in one chunk by a DMA controller
     // or a DSP or a mixer.
@@ -218,7 +264,7 @@
 
         for (int second = 0; second < NUM_SECONDS; second++) {
             // Give AAudio callback time to run in the background.
-            sleep(1);
+            usleep(200 * 1000);
 
             // Periodically print the progress so we know it hasn't died.
             printf("framesWritten = %d, XRuns = %d\n",
@@ -234,18 +280,25 @@
         }
 
         printf("timestampCount = %d\n", sTimestampData.timestampCount);
-        int printed = 0;
-        for (int i = 0; i < sTimestampData.timestampCount; i++) {
+        int printedGood = 0;
+        int printedBad = 0;
+        for (int i = 1; i < sTimestampData.timestampCount; i++) {
             TimestampInfo *timestamp = &sTimestampData.timestamps[i];
-            bool posChanged = (timestamp->timestampPosition != (timestamp - 1)->timestampPosition);
-            bool timeChanged = (timestamp->timestampNanos != (timestamp - 1)->timestampNanos);
-            if ((printed < 20) && ((i < 10) || posChanged || timeChanged)) {
-                printf("  %3d : frames %8lld, xferd %8lld", i,
-                       (long long) timestamp->framesTotal,
-                       (long long) timestamp->appPosition);
-                if (timestamp->result != AAUDIO_OK) {
-                    printf(", result = %s\n", AAudio_convertResultToText(timestamp->result));
-                } else {
+            if (timestamp->result != AAUDIO_OK) {
+                if (printedBad < 5) {
+                    printf("  %3d : frames %8lld, xferd %8lld, result = %s\n",
+                           i,
+                           (long long) timestamp->framesTotal,
+                           (long long) timestamp->appPosition,
+                           AAudio_convertResultToText(timestamp->result));
+                    printedBad++;
+                }
+            } else {
+                const bool posChanged = (timestamp->timestampPosition !=
+                                   (timestamp - 1)->timestampPosition);
+                const bool timeChanged = (timestamp->timestampNanos
+                        != (timestamp - 1)->timestampNanos);
+                if ((printedGood < 20) && (posChanged || timeChanged)) {
                     bool negative = timestamp->timestampPosition < 0;
                     bool retro = (i > 0 && (timestamp->timestampPosition <
                                             (timestamp - 1)->timestampPosition));
@@ -253,17 +306,39 @@
                                                    : (retro ? "  <= RETROGRADE!" : "");
 
                     double latency = calculateLatencyMillis(timestamp->timestampPosition,
-                                             timestamp->timestampNanos,
-                                             timestamp->appPosition,
-                                             timestamp->appNanoseconds,
-                                             actualSampleRate);
-                    printf(", STAMP: pos = %8lld, nanos = %8lld, lat = %7.1f msec %s\n",
+                                                            timestamp->timestampNanos,
+                                                            timestamp->appPosition,
+                                                            timestamp->appNanoseconds,
+                                                            actualSampleRate);
+                    printf("  %3d : frames %8lld, xferd %8lld",
+                           i,
+                           (long long) timestamp->framesTotal,
+                           (long long) timestamp->appPosition);
+                    printf(" STAMP: pos = %8lld, nanos = %8lld, lat = %7.1f msec %s\n",
                            (long long) timestamp->timestampPosition,
                            (long long) timestamp->timestampNanos,
                            latency,
                            message);
+                    printedGood++;
                 }
-                printed++;
+            }
+        }
+
+        if (printedGood == 0) {
+            printf("ERROR - AAudioStream_getTimestamp() never gave us a valid timestamp\n");
+            result = AAUDIO_ERROR_INTERNAL;
+        } else {
+            // Make sure we do not get timestamps when stopped.
+            int64_t position;
+            int64_t time;
+            aaudio_result_t tempResult = AAudioStream_getTimestamp(aaudioStream,
+                                                                   CLOCK_MONOTONIC,
+                                                                   &position, &time);
+            if (tempResult != AAUDIO_ERROR_INVALID_STATE) {
+                printf("ERROR - AAudioStream_getTimestamp() should return"
+                       " INVALID_STATE when stopped! %s\n",
+                       AAudio_convertResultToText(tempResult));
+                result = AAUDIO_ERROR_INTERNAL;
             }
         }
 
@@ -273,12 +348,14 @@
     }
 
 finish:
+
+    logTestResult(isMmap, sharingMode, performanceMode, direction, result);
+
     if (aaudioStream != nullptr) {
         AAudioStream_close(aaudioStream);
     }
     AAudioStreamBuilder_delete(aaudioBuilder);
     printf("result = %d = %s\n", result, AAudio_convertResultToText(result));
-
     return result;
 }
 
@@ -292,7 +369,7 @@
     // in a buffer if we hang or crash.
     setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
 
-    printf("Test Timestamps V0.1.3\n");
+    printf("Test Timestamps V0.1.4\n");
 
     // Legacy
     aaudio_policy_t policy = AAUDIO_POLICY_NEVER;
@@ -332,5 +409,7 @@
                             AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
                             AAUDIO_DIRECTION_OUTPUT);
 
+    printTestResults();
+
     return (result == AAUDIO_OK) ? EXIT_SUCCESS : EXIT_FAILURE;
 }
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index 827df6a..1417aaf 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -50,6 +50,7 @@
         "libmediametrics",
         "libmediautils",
         "libnblog",
+        "libprocessgroup",
         "libutils",
     ],
     export_shared_lib_headers: ["libbinder"],
diff --git a/media/libaudioclient/AudioRecord.cpp b/media/libaudioclient/AudioRecord.cpp
index 3223647..72a23e3 100644
--- a/media/libaudioclient/AudioRecord.cpp
+++ b/media/libaudioclient/AudioRecord.cpp
@@ -26,6 +26,7 @@
 #include <media/AudioRecord.h>
 #include <utils/Log.h>
 #include <private/media/AudioTrackShared.h>
+#include <processgroup/sched_policy.h>
 #include <media/IAudioFlinger.h>
 #include <media/MediaAnalyticsItem.h>
 #include <media/TypeConverter.h>
@@ -1398,6 +1399,17 @@
     return mAudioRecord->getActiveMicrophones(activeMicrophones).transactionError();
 }
 
+status_t AudioRecord::setMicrophoneDirection(audio_microphone_direction_t direction)
+{
+    AutoMutex lock(mLock);
+    return mAudioRecord->setMicrophoneDirection(direction).transactionError();
+}
+
+status_t AudioRecord::setMicrophoneFieldDimension(float zoom) {
+    AutoMutex lock(mLock);
+    return mAudioRecord->setMicrophoneFieldDimension(zoom).transactionError();
+}
+
 // =========================================================================
 
 void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index b444d2d..e9a0e22 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -29,6 +29,7 @@
 #include <media/AudioTrack.h>
 #include <utils/Log.h>
 #include <private/media/AudioTrackShared.h>
+#include <processgroup/sched_policy.h>
 #include <media/IAudioFlinger.h>
 #include <media/IAudioPolicyService.h>
 #include <media/AudioParameter.h>
diff --git a/media/libaudioclient/aidl/android/media/IAudioRecord.aidl b/media/libaudioclient/aidl/android/media/IAudioRecord.aidl
index 01e0a71..cf9c7f4 100644
--- a/media/libaudioclient/aidl/android/media/IAudioRecord.aidl
+++ b/media/libaudioclient/aidl/android/media/IAudioRecord.aidl
@@ -36,4 +36,12 @@
   /* Get a list of current active microphones.
    */
   void getActiveMicrophones(out MicrophoneInfo[] activeMicrophones);
+
+  /* Set the microphone direction (for processing).
+   */
+  void setMicrophoneDirection(int /*audio_microphone_direction_t*/ direction);
+
+  /* Set the microphone zoom (for processing).
+   */
+  void setMicrophoneFieldDimension(float zoom);
 }
diff --git a/media/libaudioclient/include/media/AudioRecord.h b/media/libaudioclient/include/media/AudioRecord.h
index 35a7e05..ebee124 100644
--- a/media/libaudioclient/include/media/AudioRecord.h
+++ b/media/libaudioclient/include/media/AudioRecord.h
@@ -534,6 +534,14 @@
      */
             status_t    getActiveMicrophones(std::vector<media::MicrophoneInfo>* activeMicrophones);
 
+    /* Set the Microphone direction (for processing purposes).
+     */
+            status_t    setMicrophoneDirection(audio_microphone_direction_t direction);
+
+    /* Set the Microphone zoom factor (for processing purposes).
+     */
+            status_t    setMicrophoneFieldDimension(float zoom);
+
      /* Get the unique port ID assigned to this AudioRecord instance by audio policy manager.
       * The ID is unique across all audioserver clients and can change during the life cycle
       * of a given AudioRecord instance if the connection to audioserver is restored.
diff --git a/media/libaudiohal/impl/DeviceHalHidl.cpp b/media/libaudiohal/impl/DeviceHalHidl.cpp
index 7a9e843..a1e869f 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.cpp
+++ b/media/libaudiohal/impl/DeviceHalHidl.cpp
@@ -268,6 +268,8 @@
         audio_input_flags_t flags,
         const char *address,
         audio_source_t source,
+        audio_devices_t outputDevice,
+        const char *outputDeviceAddress,
         sp<StreamInHalInterface> *inStream) {
     if (mDevice == 0) return NO_INIT;
     DeviceAddress hidlDevice;
@@ -283,6 +285,17 @@
     //       for now, only send the main source at 1dbfs
     SinkMetadata sinkMetadata = {{{ .source = AudioSource(source), .gain = 1 }}};
 #endif
+#if MAJOR_VERSION < 5
+    (void)outputDevice;
+    (void)outputDeviceAddress;
+#else
+    if (outputDevice != AUDIO_DEVICE_NONE) {
+        DeviceAddress hidlOutputDevice;
+        status = deviceAddressFromHal(outputDevice, outputDeviceAddress, &hidlOutputDevice);
+        if (status != OK) return status;
+        sinkMetadata.tracks[0].destination.device(std::move(hidlOutputDevice));
+    }
+#endif
     Return<void> ret = mDevice->openInputStream(
             handle,
             hidlDevice,
diff --git a/media/libaudiohal/impl/DeviceHalHidl.h b/media/libaudiohal/impl/DeviceHalHidl.h
index 291c88f..f7d465f 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.h
+++ b/media/libaudiohal/impl/DeviceHalHidl.h
@@ -86,6 +86,8 @@
             audio_input_flags_t flags,
             const char *address,
             audio_source_t source,
+            audio_devices_t outputDevice,
+            const char *outputDeviceAddress,
             sp<StreamInHalInterface> *inStream);
 
     // Returns whether createAudioPatch and releaseAudioPatch operations are supported.
diff --git a/media/libaudiohal/impl/DeviceHalLocal.cpp b/media/libaudiohal/impl/DeviceHalLocal.cpp
index dffe9da..ee68252 100644
--- a/media/libaudiohal/impl/DeviceHalLocal.cpp
+++ b/media/libaudiohal/impl/DeviceHalLocal.cpp
@@ -131,6 +131,8 @@
         audio_input_flags_t flags,
         const char *address,
         audio_source_t source,
+        audio_devices_t /*outputDevice*/,
+        const char */*outputDeviceAddress*/,
         sp<StreamInHalInterface> *inStream) {
     audio_stream_in_t *halStream;
     ALOGV("open_input_stream handle: %d devices: %x flags: %#x "
diff --git a/media/libaudiohal/impl/DeviceHalLocal.h b/media/libaudiohal/impl/DeviceHalLocal.h
index 18bd879..36db72e 100644
--- a/media/libaudiohal/impl/DeviceHalLocal.h
+++ b/media/libaudiohal/impl/DeviceHalLocal.h
@@ -79,6 +79,8 @@
             audio_input_flags_t flags,
             const char *address,
             audio_source_t source,
+            audio_devices_t outputDevice,
+            const char *outputDeviceAddress,
             sp<StreamInHalInterface> *inStream);
 
     // Returns whether createAudioPatch and releaseAudioPatch operations are supported.
diff --git a/media/libaudiohal/impl/StreamHalHidl.cpp b/media/libaudiohal/impl/StreamHalHidl.cpp
index c12b362..2e35be6 100644
--- a/media/libaudiohal/impl/StreamHalHidl.cpp
+++ b/media/libaudiohal/impl/StreamHalHidl.cpp
@@ -854,5 +854,29 @@
 }
 #endif
 
+#if MAJOR_VERSION < 5
+status_t StreamInHalHidl::setMicrophoneDirection(audio_microphone_direction_t direction __unused) {
+    if (mStream == 0) return NO_INIT;
+    return INVALID_OPERATION;
+}
+
+status_t StreamInHalHidl::setMicrophoneFieldDimension(float zoom __unused) {
+    if (mStream == 0) return NO_INIT;
+    return INVALID_OPERATION;
+}
+#else
+status_t StreamInHalHidl::setMicrophoneDirection(audio_microphone_direction_t direction) {
+    if (!mStream) return NO_INIT;
+    return processReturn("setMicrophoneDirection",
+                mStream->setMicrophoneDirection(static_cast<MicrophoneDirection>(direction)));
+}
+
+status_t StreamInHalHidl::setMicrophoneFieldDimension(float zoom) {
+    if (!mStream) return NO_INIT;
+    return processReturn("setMicrophoneFieldDimension",
+                mStream->setMicrophoneFieldDimension(zoom));
+}
+#endif
+
 } // namespace CPP_VERSION
 } // namespace android
diff --git a/media/libaudiohal/impl/StreamHalHidl.h b/media/libaudiohal/impl/StreamHalHidl.h
index f7b507e..9ac1067 100644
--- a/media/libaudiohal/impl/StreamHalHidl.h
+++ b/media/libaudiohal/impl/StreamHalHidl.h
@@ -220,6 +220,12 @@
     // Get active microphones
     virtual status_t getActiveMicrophones(std::vector<media::MicrophoneInfo> *microphones);
 
+    // Set microphone direction (for processing)
+    virtual status_t setMicrophoneDirection(audio_microphone_direction_t direction) override;
+
+    // Set microphone zoom (for processing)
+    virtual status_t setMicrophoneFieldDimension(float zoom) override;
+
     // Called when the metadata of the stream's sink has been changed.
     status_t updateSinkMetadata(const SinkMetadata& sinkMetadata) override;
 
diff --git a/media/libaudiohal/impl/StreamHalLocal.cpp b/media/libaudiohal/impl/StreamHalLocal.cpp
index 26d30d4..fcb809b 100644
--- a/media/libaudiohal/impl/StreamHalLocal.cpp
+++ b/media/libaudiohal/impl/StreamHalLocal.cpp
@@ -368,5 +368,26 @@
 }
 #endif
 
+#if MAJOR_VERSION < 5
+status_t StreamInHalLocal::setMicrophoneDirection(audio_microphone_direction_t direction __unused) {
+    return INVALID_OPERATION;
+}
+
+status_t StreamInHalLocal::setMicrophoneFieldDimension(float zoom __unused) {
+    return INVALID_OPERATION;
+}
+#else
+status_t StreamInHalLocal::setMicrophoneDirection(audio_microphone_direction_t direction) {
+    if (mStream->set_microphone_direction == NULL) return INVALID_OPERATION;
+    return mStream->set_microphone_direction(mStream, direction);
+}
+
+status_t StreamInHalLocal::setMicrophoneFieldDimension(float zoom) {
+    if (mStream->set_microphone_field_dimension == NULL) return INVALID_OPERATION;
+    return mStream->set_microphone_field_dimension(mStream, zoom);
+
+}
+#endif
+
 } // namespace CPP_VERSION
 } // namespace android
diff --git a/media/libaudiohal/impl/StreamHalLocal.h b/media/libaudiohal/impl/StreamHalLocal.h
index 4fd1960..3d6c50e 100644
--- a/media/libaudiohal/impl/StreamHalLocal.h
+++ b/media/libaudiohal/impl/StreamHalLocal.h
@@ -204,6 +204,12 @@
     // Get active microphones
     virtual status_t getActiveMicrophones(std::vector<media::MicrophoneInfo> *microphones);
 
+    // Sets microphone direction (for processing)
+    virtual status_t setMicrophoneDirection(audio_microphone_direction_t direction);
+
+    // Sets microphone zoom (for processing)
+    virtual status_t setMicrophoneFieldDimension(float zoom);
+
     // Called when the metadata of the stream's sink has been changed.
     status_t updateSinkMetadata(const SinkMetadata& sinkMetadata) override;
 
diff --git a/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h b/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
index 7de8eb3..e565237 100644
--- a/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
@@ -84,6 +84,8 @@
             audio_input_flags_t flags,
             const char *address,
             audio_source_t source,
+            audio_devices_t outputDevice,
+            const char *outputDeviceAddress,
             sp<StreamInHalInterface> *inStream) = 0;
 
     // Returns whether createAudioPatch and releaseAudioPatch operations are supported.
diff --git a/media/libaudiohal/include/media/audiohal/StreamHalInterface.h b/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
index bd71dc0..ed8282f 100644
--- a/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
@@ -179,6 +179,12 @@
     // Get active microphones
     virtual status_t getActiveMicrophones(std::vector<media::MicrophoneInfo> *microphones) = 0;
 
+    // Set direction for capture processing
+    virtual status_t setMicrophoneDirection(audio_microphone_direction_t) = 0;
+
+    // Set zoom factor for capture stream
+    virtual status_t setMicrophoneFieldDimension(float zoom) = 0;
+
     struct SinkMetadata {
         std::vector<record_track_metadata_t> tracks;
     };
diff --git a/media/libmedia/Android.bp b/media/libmedia/Android.bp
index 3efb5de..68dae56 100644
--- a/media/libmedia/Android.bp
+++ b/media/libmedia/Android.bp
@@ -213,6 +213,7 @@
         "android.hidl.token@1.0-utils",
         "liblog",
         "libcutils",
+        "libprocessgroup",
         "libutils",
         "libbinder",
         "libsonivox",
diff --git a/media/libmedia/IMediaExtractor.cpp b/media/libmedia/IMediaExtractor.cpp
index e9a6230..fb6d3a2 100644
--- a/media/libmedia/IMediaExtractor.cpp
+++ b/media/libmedia/IMediaExtractor.cpp
@@ -19,6 +19,7 @@
 #include <utils/Log.h>
 
 #include <stdint.h>
+#include <time.h>
 #include <sys/types.h>
 
 #include <binder/IPCThreadState.h>
@@ -219,10 +220,16 @@
     Vector<wp<IMediaSource>> tracks;
     Vector<String8> trackDescriptions;
     String8 toString() const;
+    time_t when;
 } ExtractorInstance;
 
 String8 ExtractorInstance::toString() const {
-    String8 str = name;
+    String8 str;
+    char timeString[32];
+    strftime(timeString, sizeof(timeString), "%m-%d %T", localtime(&when));
+    str.append(timeString);
+    str.append(": ");
+    str.append(name);
     str.append(" for mime ");
     str.append(mime);
     str.append(", source ");
@@ -287,6 +294,7 @@
     ex.sourceDescription = source->toString();
     ex.owner = IPCThreadState::self()->getCallingPid();
     ex.extractor = extractor;
+    ex.when = time(NULL);
 
     {
         Mutex::Autolock lock(sExtractorsLock);
diff --git a/media/libmedia/IMediaMetadataRetriever.cpp b/media/libmedia/IMediaMetadataRetriever.cpp
index 590ba1a..f9fa86e 100644
--- a/media/libmedia/IMediaMetadataRetriever.cpp
+++ b/media/libmedia/IMediaMetadataRetriever.cpp
@@ -23,6 +23,7 @@
 #include <media/IDataSource.h>
 #include <media/IMediaHTTPService.h>
 #include <media/IMediaMetadataRetriever.h>
+#include <processgroup/sched_policy.h>
 #include <utils/String8.h>
 #include <utils/KeyedVector.h>
 
diff --git a/media/libmedia/TypeConverter.cpp b/media/libmedia/TypeConverter.cpp
index 0ab0e9b..c24e046 100644
--- a/media/libmedia/TypeConverter.cpp
+++ b/media/libmedia/TypeConverter.cpp
@@ -74,6 +74,7 @@
     MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
     MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
     MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_HDMI),
+    MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_HDMI_ARC),
     MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_TELEPHONY_RX),
     MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
     MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index 2547888..c7da7c7 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -862,9 +862,9 @@
 //static
 sp<CodecBase> MediaCodec::GetCodecBase(const AString &name, const char *owner) {
     if (owner) {
-        if (strncmp(owner, "default", 8) == 0) {
+        if (strcmp(owner, "default") == 0) {
             return new ACodec;
-        } else if (strncmp(owner, "codec2", 7) == 0) {
+        } else if (strncmp(owner, "codec2", 6) == 0) {
             return CreateCCodec();
         }
     }
diff --git a/media/libstagefright/MediaExtractor.cpp b/media/libstagefright/MediaExtractor.cpp
index 9511931..4ed3382 100644
--- a/media/libstagefright/MediaExtractor.cpp
+++ b/media/libstagefright/MediaExtractor.cpp
@@ -57,7 +57,7 @@
 }
 
 MediaTrack *MediaExtractorCUnwrapper::getTrack(size_t index) {
-    return new MediaTrackCUnwrapper(plugin->getTrack(plugin->data, index));
+    return MediaTrackCUnwrapper::create(plugin->getTrack(plugin->data, index));
 }
 
 status_t MediaExtractorCUnwrapper::getTrackMetaData(
diff --git a/media/libstagefright/MediaTrack.cpp b/media/libstagefright/MediaTrack.cpp
index 036e79d..89c9b25 100644
--- a/media/libstagefright/MediaTrack.cpp
+++ b/media/libstagefright/MediaTrack.cpp
@@ -65,6 +65,13 @@
     bufferGroup = nullptr;
 }
 
+MediaTrackCUnwrapper *MediaTrackCUnwrapper::create(CMediaTrack *cmediatrack) {
+    if (cmediatrack == nullptr) {
+        return nullptr;
+    }
+    return new MediaTrackCUnwrapper(cmediatrack);
+}
+
 MediaTrackCUnwrapper::~MediaTrackCUnwrapper() {
     wrapper->free(wrapper->data);
     free(wrapper);
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 26f76c0..0d6ef46 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -2379,7 +2379,8 @@
         return BAD_VALUE;
     }
 
-    sp<ThreadBase> thread = openInput_l(module, input, config, *devices, address, source, flags);
+    sp<ThreadBase> thread = openInput_l(
+            module, input, config, *devices, address, source, flags, AUDIO_DEVICE_NONE, String8{});
 
     if (thread != 0) {
         // notify client processes of the new input creation
@@ -2395,7 +2396,9 @@
                                                          audio_devices_t devices,
                                                          const String8& address,
                                                          audio_source_t source,
-                                                         audio_input_flags_t flags)
+                                                         audio_input_flags_t flags,
+                                                         audio_devices_t outputDevice,
+                                                         const String8& outputDeviceAddress)
 {
     AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices);
     if (inHwDev == NULL) {
@@ -2424,7 +2427,8 @@
     sp<DeviceHalInterface> inHwHal = inHwDev->hwDevice();
     sp<StreamInHalInterface> inStream;
     status_t status = inHwHal->openInputStream(
-            *input, devices, &halconfig, flags, address.string(), source, &inStream);
+            *input, devices, &halconfig, flags, address.string(), source,
+            outputDevice, outputDeviceAddress, &inStream);
     ALOGV("openInput_l() openInputStream returned input %p, devices %#x, SamplingRate %d"
            ", Format %#x, Channels %#x, flags %#x, status %d addr %s",
             inStream.get(),
@@ -2447,7 +2451,8 @@
         ALOGV("openInput_l() reopening with proposed sampling rate and channel mask");
         inStream.clear();
         status = inHwHal->openInputStream(
-                *input, devices, &halconfig, flags, address.string(), source, &inStream);
+                *input, devices, &halconfig, flags, address.string(), source,
+                outputDevice, outputDeviceAddress, &inStream);
         // FIXME log this new status; HAL should not propose any further changes
     }
 
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 6c698f6..c1169d2 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -579,6 +579,10 @@
         virtual binder::Status   stop();
         virtual binder::Status   getActiveMicrophones(
                 std::vector<media::MicrophoneInfo>* activeMicrophones);
+        virtual binder::Status   setMicrophoneDirection(
+                int /*audio_microphone_direction_t*/ direction);
+        virtual binder::Status   setMicrophoneFieldDimension(float zoom);
+
     private:
         const sp<RecordThread::RecordTrack> mRecordTrack;
 
@@ -620,7 +624,9 @@
                                            audio_devices_t device,
                                            const String8& address,
                                            audio_source_t source,
-                                           audio_input_flags_t flags);
+                                           audio_input_flags_t flags,
+                                           audio_devices_t outputDevice,
+                                           const String8& outputDeviceAddress);
               sp<ThreadBase> openOutput_l(audio_module_handle_t module,
                                               audio_io_handle_t *output,
                                               audio_config_t *config,
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index 63a9ec4..3381e4d 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -211,6 +211,8 @@
                 ((patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) &&
                  ((patch->sinks[0].ext.device.hw_module != srcModule) ||
                   !audioHwDevice->supportsAudioPatches()))) {
+                audio_devices_t outputDevice = AUDIO_DEVICE_NONE;
+                String8 outputDeviceAddress;
                 if (patch->num_sources == 2) {
                     if (patch->sources[1].type != AUDIO_PORT_TYPE_MIX ||
                             (patch->num_sinks != 0 && patch->sinks[0].ext.device.hw_module !=
@@ -261,6 +263,8 @@
                         goto exit;
                     }
                     newPatch.mPlayback.setThread(reinterpret_cast<PlaybackThread*>(thread.get()));
+                    outputDevice = device;
+                    outputDeviceAddress = address;
                 }
                 audio_devices_t device = patch->sources[0].ext.device.type;
                 String8 address = String8(patch->sources[0].ext.device.address);
@@ -293,7 +297,9 @@
                                                                     device,
                                                                     address,
                                                                     AUDIO_SOURCE_MIC,
-                                                                    flags);
+                                                                    flags,
+                                                                    outputDevice,
+                                                                    outputDeviceAddress);
                 ALOGV("mAudioFlinger.openInput_l() returned %p inChannelMask %08x",
                       thread.get(), config.channel_mask);
                 if (thread == 0) {
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index 85f5456..32af7d5 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -71,6 +71,9 @@
 
             status_t    getActiveMicrophones(std::vector<media::MicrophoneInfo>* activeMicrophones);
 
+            status_t    setMicrophoneDirection(audio_microphone_direction_t direction);
+            status_t    setMicrophoneFieldDimension(float zoom);
+
     static  bool        checkServerLatencySupported(
                                 audio_format_t format, audio_input_flags_t flags) {
                             return audio_is_linear_pcm(format)
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 607d2d1..31a8c7d 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -7582,6 +7582,20 @@
     return status;
 }
 
+status_t AudioFlinger::RecordThread::setMicrophoneDirection(audio_microphone_direction_t direction)
+{
+    ALOGV("RecordThread::setMicrophoneDirection");
+    AutoMutex _l(mLock);
+    return mInput->stream->setMicrophoneDirection(direction);
+}
+
+status_t AudioFlinger::RecordThread::setMicrophoneFieldDimension(float zoom)
+{
+    ALOGV("RecordThread::setMicrophoneFieldDimension");
+    AutoMutex _l(mLock);
+    return mInput->stream->setMicrophoneFieldDimension(zoom);
+}
+
 void AudioFlinger::RecordThread::updateMetadata_l()
 {
     if (mInput == nullptr || mInput->stream == nullptr ||
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 5d06773..aab7601 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -1545,6 +1545,9 @@
 
             status_t    getActiveMicrophones(std::vector<media::MicrophoneInfo>* activeMicrophones);
 
+            status_t    setMicrophoneDirection(audio_microphone_direction_t direction);
+            status_t    setMicrophoneFieldDimension(float zoom);
+
             void        updateMetadata_l() override;
 
             bool        fastTrackAvailable() const { return mFastTrackAvail; }
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 9a7f1f1..d23d19d 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1710,6 +1710,18 @@
             mRecordTrack->getActiveMicrophones(activeMicrophones));
 }
 
+binder::Status AudioFlinger::RecordHandle::setMicrophoneDirection(
+        int /*audio_microphone_direction_t*/ direction) {
+    ALOGV("%s()", __func__);
+    return binder::Status::fromStatusT(mRecordTrack->setMicrophoneDirection(
+            static_cast<audio_microphone_direction_t>(direction)));
+}
+
+binder::Status AudioFlinger::RecordHandle::setMicrophoneFieldDimension(float zoom) {
+    ALOGV("%s()", __func__);
+    return binder::Status::fromStatusT(mRecordTrack->setMicrophoneFieldDimension(zoom));
+}
+
 // ----------------------------------------------------------------------------
 #undef LOG_TAG
 #define LOG_TAG "AF::RecordTrack"
@@ -2004,6 +2016,27 @@
     }
 }
 
+status_t AudioFlinger::RecordThread::RecordTrack::setMicrophoneDirection(
+        audio_microphone_direction_t direction) {
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        RecordThread *recordThread = (RecordThread *)thread.get();
+        return recordThread->setMicrophoneDirection(direction);
+    } else {
+        return BAD_VALUE;
+    }
+}
+
+status_t AudioFlinger::RecordThread::RecordTrack::setMicrophoneFieldDimension(float zoom) {
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        RecordThread *recordThread = (RecordThread *)thread.get();
+        return recordThread->setMicrophoneFieldDimension(zoom);
+    } else {
+        return BAD_VALUE;
+    }
+}
+
 // ----------------------------------------------------------------------------
 #undef LOG_TAG
 #define LOG_TAG "AF::PatchRecord"
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
index fa9ba0b..d4cfd1e 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
@@ -23,11 +23,12 @@
 #include "AudioIODescriptorInterface.h"
 #include "AudioPort.h"
 #include "ClientDescriptor.h"
+#include "DeviceDescriptor.h"
 #include "EffectDescriptor.h"
+#include "IOProfile.h"
 
 namespace android {
 
-class IOProfile;
 class AudioMix;
 class AudioPolicyClientInterface;
 
@@ -42,10 +43,16 @@
     audio_port_handle_t getId() const;
     audio_module_handle_t getModuleHandle() const;
 
+    audio_devices_t getDeviceType() const { return (mDevice != nullptr) ?
+                    mDevice->type() : AUDIO_DEVICE_NONE; }
+    sp<DeviceDescriptor> getDevice() const { return mDevice; }
+    void setDevice(const sp<DeviceDescriptor> &device) { mDevice = device; }
+    DeviceVector supportedDevices() const  {
+        return mProfile != nullptr ? mProfile->getSupportedDevices() :  DeviceVector(); }
+
     void dump(String8 *dst) const override;
 
     audio_io_handle_t   mIoHandle = AUDIO_IO_HANDLE_NONE; // input handle
-    audio_devices_t     mDevice = AUDIO_DEVICE_NONE;  // current device this input is routed to
     AudioMix            *mPolicyMix = nullptr;        // non NULL when used by a dynamic policy
     const sp<IOProfile> mProfile;                     // I/O profile this output derives from
 
@@ -61,6 +68,7 @@
     bool isSourceActive(audio_source_t source) const;
     audio_source_t source() const;
     bool isSoundTrigger() const;
+    audio_attributes_t getHighestPriorityAttributes() const;
     void setClientActive(const sp<RecordClientDescriptor>& client, bool active);
     int32_t activeCount() { return mGlobalActiveCount; }
     void trackEffectEnabled(const sp<EffectDescriptor> &effect, bool enabled);
@@ -71,8 +79,7 @@
     void setPatchHandle(audio_patch_handle_t handle) override;
 
     status_t open(const audio_config_t *config,
-                  audio_devices_t device,
-                  const String8& address,
+                  const sp<DeviceDescriptor> &device,
                   audio_source_t source,
                   audio_input_flags_t flags,
                   audio_io_handle_t *input);
@@ -99,6 +106,8 @@
 
     audio_patch_handle_t mPatchHandle = AUDIO_PATCH_HANDLE_NONE;
     audio_port_handle_t  mId = AUDIO_PORT_HANDLE_NONE;
+    sp<DeviceDescriptor> mDevice = nullptr; /**< current device this input is routed to */
+
     // Because a preemptible capture session can preempt another one, we end up in an endless loop
     // situation were each session is allowed to restart after being preempted,
     // thus preempting the other one which restarts and so on.
@@ -120,8 +129,8 @@
     sp<AudioInputDescriptor> getInputFromId(audio_port_handle_t id) const;
 
     // count active capture sessions using one of the specified devices.
-    // ignore devices if AUDIO_DEVICE_IN_DEFAULT is passed
-    uint32_t activeInputsCountOnDevices(audio_devices_t devices = AUDIO_DEVICE_IN_DEFAULT) const;
+    // ignore devices if empty vector is passed
+    uint32_t activeInputsCountOnDevices(const DeviceVector &devices) const;
 
     /**
      * return io handle of active input or 0 if no input is active
@@ -130,8 +139,6 @@
      */
     Vector<sp <AudioInputDescriptor> > getActiveInputs();
 
-    audio_devices_t getSupportedDevices(audio_io_handle_t handle) const;
-
     sp<AudioInputDescriptor> getInputForClient(audio_port_handle_t portId);
 
     void trackEffectEnabled(const sp<EffectDescriptor> &effect, bool enabled);
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
index ed995e0..14b995b 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -26,13 +26,14 @@
 #include "AudioIODescriptorInterface.h"
 #include "AudioPort.h"
 #include "ClientDescriptor.h"
+#include "DeviceDescriptor.h"
+#include <map>
 
 namespace android {
 
 class IOProfile;
 class AudioMix;
 class AudioPolicyClientInterface;
-class DeviceDescriptor;
 
 // descriptor for audio outputs. Used to maintain current configuration of each opened audio output
 // and keep track of the usage of this output by each audio stream type.
@@ -48,14 +49,12 @@
     void        log(const char* indent);
 
     audio_port_handle_t getId() const;
-    virtual audio_devices_t device() const;
-    virtual bool sharesHwModuleWith(const sp<AudioOutputDescriptor>& outputDesc);
-    virtual audio_devices_t supportedDevices();
+    virtual DeviceVector devices() const { return mDevices; }
+    bool sharesHwModuleWith(const sp<AudioOutputDescriptor>& outputDesc);
+    virtual DeviceVector supportedDevices() const  { return mDevices; }
     virtual bool isDuplicated() const { return false; }
     virtual uint32_t latency() { return 0; }
     virtual bool isFixedVolume(audio_devices_t device);
-    virtual sp<AudioOutputDescriptor> subOutput1() { return 0; }
-    virtual sp<AudioOutputDescriptor> subOutput2() { return 0; }
     virtual bool setVolume(float volume,
                            audio_stream_type_t stream,
                            audio_devices_t device,
@@ -119,7 +118,7 @@
         return mActiveClients;
     }
 
-    audio_devices_t mDevice = AUDIO_DEVICE_NONE; // current device this output is routed to
+    DeviceVector mDevices; /**< current devices this output is routed to */
     nsecs_t mStopTime[AUDIO_STREAM_CNT];
     int mMuteCount[AUDIO_STREAM_CNT];            // mute request counter
     bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible
@@ -151,14 +150,15 @@
     virtual ~SwAudioOutputDescriptor() {}
 
             void dump(String8 *dst) const override;
-    virtual audio_devices_t device() const;
-    virtual bool sharesHwModuleWith(const sp<AudioOutputDescriptor>& outputDesc);
-    virtual audio_devices_t supportedDevices();
+    virtual DeviceVector devices() const;
+    void setDevices(const DeviceVector &devices) { mDevices = devices; }
+    bool sharesHwModuleWith(const sp<SwAudioOutputDescriptor>& outputDesc);
+    virtual DeviceVector supportedDevices() const;
     virtual uint32_t latency();
     virtual bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
     virtual bool isFixedVolume(audio_devices_t device);
-    virtual sp<AudioOutputDescriptor> subOutput1() { return mOutput1; }
-    virtual sp<AudioOutputDescriptor> subOutput2() { return mOutput2; }
+    sp<SwAudioOutputDescriptor> subOutput1() { return mOutput1; }
+    sp<SwAudioOutputDescriptor> subOutput2() { return mOutput2; }
             void changeStreamActiveCount(
                     const sp<TrackClientDescriptor>& client, int delta) override;
     virtual bool setVolume(float volume,
@@ -171,22 +171,49 @@
                            const struct audio_port_config *srcConfig = NULL) const;
     virtual void toAudioPort(struct audio_port *port) const;
 
-            status_t open(const audio_config_t *config,
-                          audio_devices_t device,
-                          const String8& address,
-                          audio_stream_type_t stream,
-                          audio_output_flags_t flags,
-                          audio_io_handle_t *output);
-            // Called when a stream is about to be started
-            // Note: called before setClientActive(true);
-            status_t start();
-            // Called after a stream is stopped.
-            // Note: called after setClientActive(false);
-            void stop();
-            void close();
-            status_t openDuplicating(const sp<SwAudioOutputDescriptor>& output1,
-                                     const sp<SwAudioOutputDescriptor>& output2,
-                                     audio_io_handle_t *ioHandle);
+        status_t open(const audio_config_t *config,
+                      const DeviceVector &devices,
+                      audio_stream_type_t stream,
+                      audio_output_flags_t flags,
+                      audio_io_handle_t *output);
+
+        // Called when a stream is about to be started
+        // Note: called before setClientActive(true);
+        status_t start();
+        // Called after a stream is stopped.
+        // Note: called after setClientActive(false);
+        void stop();
+        void close();
+        status_t openDuplicating(const sp<SwAudioOutputDescriptor>& output1,
+                                 const sp<SwAudioOutputDescriptor>& output2,
+                                 audio_io_handle_t *ioHandle);
+
+    /**
+     * @brief supportsDevice
+     * @param device to be checked against
+     * @return true if the device is supported by type (for non bus / remote submix devices),
+     *         true if the device is supported (both type and address) for bus / remote submix
+     *         false otherwise
+     */
+    bool supportsDevice(const sp<DeviceDescriptor> &device) const;
+
+    /**
+     * @brief supportsAllDevices
+     * @param devices to be checked against
+     * @return true if the device is weakly supported by type (e.g. for non bus / rsubmix devices),
+     *         true if the device is supported (both type and address) for bus / remote submix
+     *         false otherwise
+     */
+    bool supportsAllDevices(const DeviceVector &devices) const;
+
+    /**
+     * @brief filterSupportedDevices takes a vector of devices and filters them according to the
+     * device supported by this output (the profile from which this output derives from)
+     * @param devices reference device vector to be filtered
+     * @return vector of devices filtered from the supported devices of this output (weakly or not
+     * depending on the device type)
+     */
+    DeviceVector filterSupportedDevices(const DeviceVector &devices) const;
 
     const sp<IOProfile> mProfile;          // I/O profile this output derives from
     audio_io_handle_t mIoHandle;           // output handle
@@ -208,7 +235,6 @@
 
             void dump(String8 *dst) const override;
 
-    virtual audio_devices_t supportedDevices();
     virtual bool setVolume(float volume,
                            audio_stream_type_t stream,
                            audio_devices_t device,
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
index 955e87b..e6a62d9 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
@@ -16,6 +16,7 @@
 
 #pragma once
 
+#include "DeviceDescriptor.h"
 #include <utils/RefBase.h>
 #include <media/AudioPolicy.h>
 #include <utils/KeyedVector.h>
@@ -74,9 +75,9 @@
     status_t getOutputForAttr(audio_attributes_t attributes, uid_t uid,
             sp<SwAudioOutputDescriptor> &desc);
 
-    audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource,
-                                                  audio_devices_t availableDeviceTypes,
-                                                  AudioMix **policyMix);
+    sp<DeviceDescriptor> getDeviceAndMixForInputSource(audio_source_t inputSource,
+                                                       const DeviceVector &availableDeviceTypes,
+                                                       AudioMix **policyMix);
 
     status_t getInputMixForAttr(audio_attributes_t attr, AudioMix **policyMix);
 
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
index bb9cad8..1b5a2d6 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
@@ -65,6 +65,7 @@
     uint32_t getFlags() const { return mFlags; }
 
     virtual void attach(const sp<HwModule>& module);
+    virtual void detach();
     bool isAttached() { return mModule != 0; }
 
     // Audio port IDs are in a different namespace than AudioFlinger unique IDs
@@ -161,7 +162,7 @@
                                    const struct audio_port_config *srcConfig = NULL) const = 0;
     virtual sp<AudioPort> getAudioPort() const = 0;
     virtual bool hasSameHwModuleAs(const sp<AudioPortConfig>& other) const {
-        return (other != 0) &&
+        return (other != 0) && (other->getAudioPort() != 0) && (getAudioPort() != 0) &&
                 (other->getAudioPort()->getModuleHandle() == getAudioPort()->getModuleHandle());
     }
     unsigned int mSamplingRate = 0u;
diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
index d02123c..b581665 100644
--- a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
@@ -53,6 +53,8 @@
 
     // AudioPort
     virtual void attach(const sp<HwModule>& module);
+    virtual void detach();
+
     virtual void toAudioPort(struct audio_port *port) const;
     virtual void importAudioPort(const sp<AudioPort>& port, bool force = false);
 
@@ -164,6 +166,23 @@
         return !operator==(right);
     }
 
+    /**
+     * @brief getFirstValidAddress
+     * @return the first valid address of a list of device, "" if no device with valid address
+     * found.
+     * This helper function helps maintaining compatibility with legacy where we used to have a
+     * devices mask and an address.
+     */
+    String8 getFirstValidAddress() const
+    {
+        for (const auto &device : *this) {
+            if (device->address() != "") {
+                return device->address();
+            }
+        }
+        return String8("");
+    }
+
     std::string toString() const;
 
     void dump(String8 *dst, const String8 &tag, int spaces = 0, bool verbose = true) const;
diff --git a/services/audiopolicy/common/managerdefinitions/include/HwModule.h b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
index 2b57fa9..d7dc4b0 100644
--- a/services/audiopolicy/common/managerdefinitions/include/HwModule.h
+++ b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
@@ -46,6 +46,22 @@
 
     const DeviceVector &getDeclaredDevices() const { return mDeclaredDevices; }
     void setDeclaredDevices(const DeviceVector &devices);
+    DeviceVector getAllDevices() const
+    {
+        DeviceVector devices = mDeclaredDevices;
+        devices.merge(mDynamicDevices);
+        return devices;
+    }
+    void addDynamicDevice(const sp<DeviceDescriptor> &device)
+    {
+        mDynamicDevices.add(device);
+    }
+
+    bool removeDynamicDevice(const sp<DeviceDescriptor> &device)
+    {
+        return mDynamicDevices.remove(device) >= 0;
+    }
+    DeviceVector getDynamicDevices() const { return mDynamicDevices; }
 
     const InputProfileCollection &getInputProfiles() const { return mInputProfiles; }
     const OutputProfileCollection &getOutputProfiles() const { return mOutputProfiles; }
@@ -104,6 +120,7 @@
     InputProfileCollection mInputProfiles;  // input profiles exposed by this module
     uint32_t mHalVersion; // audio HAL API version
     DeviceVector mDeclaredDevices; // devices declared in audio_policy configuration file.
+    DeviceVector mDynamicDevices; /**< devices that can be added/removed at runtime (e.g. rsbumix)*/
     AudioRouteVector mRoutes;
     AudioPortVector mPorts;
 };
@@ -113,13 +130,58 @@
 public:
     sp<HwModule> getModuleFromName(const char *name) const;
 
-    sp<HwModule> getModuleForDevice(audio_devices_t device) const;
+    sp<HwModule> getModuleForDeviceTypes(audio_devices_t device) const;
 
-    sp<DeviceDescriptor> getDeviceDescriptor(const audio_devices_t device,
-                                             const char *device_address,
-                                             const char *device_name,
+    sp<HwModule> getModuleForDevice(const sp<DeviceDescriptor> &device) const;
+
+    DeviceVector getAvailableDevicesFromModuleName(const char *name,
+                                                   const DeviceVector &availableDevices) const;
+
+    /**
+     * @brief getDeviceDescriptor returns a device descriptor associated to the device type and
+     * device address (if matchAddress is true).
+     * It may loop twice on all modules to check if allowToCreate is true
+     *      -first loop will check if the device is found on a module since declared in the list
+     * of device port in configuration file
+     *      -(allowToCreate is true)second loop will check if the device is weakly supported by one
+     * or more profiles on a given module and will add as a supported device for this module.
+     *       The device will also be added to the dynamic list of device of this module
+     * @param type of the device requested
+     * @param address of the device requested
+     * @param name of the device that requested
+     * @param matchAddress true if a strong match is required
+     * @param allowToCreate true if allowed to create dynamic device (e.g. hdmi, usb...)
+     * @return device descriptor associated to the type (and address if matchAddress is true)
+     */
+    sp<DeviceDescriptor> getDeviceDescriptor(const audio_devices_t type,
+                                             const char *address,
+                                             const char *name,
+                                             bool allowToCreate = false,
                                              bool matchAddress = true) const;
 
+    /**
+     * @brief createDevice creates a new device from the type and address given. It checks that
+     * according to the device type, a module is supporting this device (weak check).
+     * This concerns only dynamic device, aka device with a specific address and not
+     * already supported by module/underlying profiles.
+     * @param type of the device to be created
+     * @param address of the device to be created
+     * @param name of the device to be created
+     * @return device descriptor if a module is supporting this type, nullptr otherwise.
+     */
+    sp<DeviceDescriptor> createDevice(const audio_devices_t type,
+                                      const char *address,
+                                      const char *name) const;
+
+    /**
+     * @brief cleanUpForDevice: loop on all profiles of all modules to remove device from
+     * the list of supported device. If this device is a dynamic device (aka a device not in the
+     * xml file with a runtime address), it is also removed from the module collection of dynamic
+     * devices.
+     * @param device that has been disconnected
+     */
+    void cleanUpForDevice(const sp<DeviceDescriptor> &device);
+
     void dump(String8 *dst) const;
 };
 
diff --git a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
index ca6ca56..d0c05a5 100644
--- a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
+++ b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
@@ -57,12 +57,25 @@
         }
     }
 
-    // This method is used for input and direct output, and is not used for other output.
-    // If parameter updatedSamplingRate is non-NULL, it is assigned the actual sample rate.
-    // For input, flags is interpreted as audio_input_flags_t.
-    // TODO: merge audio_output_flags_t and audio_input_flags_t.
-    bool isCompatibleProfile(audio_devices_t device,
-                             const String8& address,
+    /**
+     * @brief isCompatibleProfile: This method is used for input and direct output,
+     * and is not used for other output.
+     * Checks if the IO profile is compatible with specified parameters.
+     * For input, flags is interpreted as audio_input_flags_t.
+     * TODO: merge audio_output_flags_t and audio_input_flags_t.
+     *
+     * @param devices vector of devices to be checked for compatibility
+     * @param samplingRate to be checked for compatibility. Must be specified
+     * @param updatedSamplingRate if non-NULL, it is assigned the actual sample rate.
+     * @param format to be checked for compatibility. Must be specified
+     * @param updatedFormat if non-NULL, it is assigned the actual format
+     * @param channelMask to be checked for compatibility. Must be specified
+     * @param updatedChannelMask if non-NULL, it is assigned the actual channel mask
+     * @param flags to be checked for compatibility
+     * @param exactMatchRequiredForInputFlags true if exact match is required on flags
+     * @return true if the profile is compatible, false otherwise.
+     */
+    bool isCompatibleProfile(const DeviceVector &devices,
                              uint32_t samplingRate,
                              uint32_t *updatedSamplingRate,
                              audio_format_t format,
@@ -78,7 +91,7 @@
 
     bool hasSupportedDevices() const { return !mSupportedDevices.isEmpty(); }
 
-    bool supportDevice(audio_devices_t device) const
+    bool supportsDeviceTypes(audio_devices_t device) const
     {
         if (audio_is_output_devices(device)) {
             return mSupportedDevices.types() & device;
@@ -86,41 +99,37 @@
         return mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN);
     }
 
-    bool supportDeviceAddress(const String8 &address) const
+    /**
+     * @brief supportsDevice
+     * @param device to be checked against
+     *        forceCheckOnAddress if true, check on type and address whatever the type, otherwise
+     *        the address enforcement is limited to "offical devices" that distinguishe on address
+     * @return true if the device is supported by type (for non bus / remote submix devices),
+     *         true if the device is supported (both type and address) for bus / remote submix
+     *         false otherwise
+     */
+    bool supportsDevice(const sp<DeviceDescriptor> &device, bool forceCheckOnAddress = false) const
     {
-        return mSupportedDevices[0]->address() == address;
-    }
-
-    // chose first device present in mSupportedDevices also part of deviceType
-    audio_devices_t getSupportedDeviceForType(audio_devices_t deviceType) const
-    {
-        for (size_t k = 0; k  < mSupportedDevices.size(); k++) {
-            audio_devices_t profileType = mSupportedDevices[k]->type();
-            if (profileType & deviceType) {
-                return profileType;
-            }
+        if (!device_distinguishes_on_address(device->type()) && !forceCheckOnAddress) {
+            return supportsDeviceTypes(device->type());
         }
-        return AUDIO_DEVICE_NONE;
+        return mSupportedDevices.contains(device);
     }
 
-    audio_devices_t getSupportedDevicesType() const { return mSupportedDevices.types(); }
-
     void clearSupportedDevices() { mSupportedDevices.clear(); }
     void addSupportedDevice(const sp<DeviceDescriptor> &device)
     {
         mSupportedDevices.add(device);
     }
-
+    void removeSupportedDevice(const sp<DeviceDescriptor> &device)
+    {
+        mSupportedDevices.remove(device);
+    }
     void setSupportedDevices(const DeviceVector &devices)
     {
         mSupportedDevices = devices;
     }
 
-    sp<DeviceDescriptor> getSupportedDeviceByAddress(audio_devices_t type, String8 address) const
-    {
-        return mSupportedDevices.getDevice(type, address);
-    }
-
     const DeviceVector &getSupportedDevices() const { return mSupportedDevices; }
 
     bool canOpenNewIo() {
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
index 0bc88a5..55d4db4 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
@@ -21,7 +21,6 @@
 #include <policy.h>
 #include <AudioPolicyInterface.h>
 #include "AudioInputDescriptor.h"
-#include "IOProfile.h"
 #include "AudioGain.h"
 #include "HwModule.h"
 
@@ -55,30 +54,7 @@
 
 audio_source_t AudioInputDescriptor::source() const
 {
-    audio_source_t source = AUDIO_SOURCE_DEFAULT;
-
-    for (bool activeOnly : { true, false }) {
-        int32_t topPriority = -1;
-        app_state_t topState = APP_STATE_IDLE;
-        for (const auto &client : getClientIterable()) {
-            if (activeOnly && !client->active()) {
-                continue;
-            }
-            app_state_t curState = client->appState();
-            if (curState >= topState) {
-                int32_t curPriority = source_priority(client->source());
-                if (curPriority > topPriority) {
-                    source = client->source();
-                    topPriority = curPriority;
-                }
-                topState = curState;
-            }
-        }
-        if (source != AUDIO_SOURCE_DEFAULT) {
-            break;
-        }
-    }
-    return source;
+    return getHighestPriorityAttributes().source;
 }
 
 void AudioInputDescriptor::toAudioPortConfig(struct audio_port_config *dstConfig,
@@ -148,6 +124,34 @@
     return false;
 }
 
+audio_attributes_t AudioInputDescriptor::getHighestPriorityAttributes() const
+{
+    audio_attributes_t attributes = { .source = AUDIO_SOURCE_DEFAULT };
+
+    for (bool activeOnly : { true, false }) {
+        int32_t topPriority = -1;
+        app_state_t topState = APP_STATE_IDLE;
+        for (const auto &client : getClientIterable()) {
+            if (activeOnly && !client->active()) {
+              continue;
+            }
+            app_state_t curState = client->appState();
+            if (curState >= topState) {
+                int32_t curPriority = source_priority(client->source());
+                if (curPriority > topPriority) {
+                    attributes = client->attributes();
+                    topPriority = curPriority;
+                }
+                topState = curState;
+            }
+        }
+        if (attributes.source != AUDIO_SOURCE_DEFAULT) {
+            break;
+        }
+    }
+    return attributes;
+}
+
 bool AudioInputDescriptor::isSoundTrigger() const {
     // sound trigger and non sound trigger clients are not mixed on a given input
     // so check only first client
@@ -180,8 +184,7 @@
 }
 
 status_t AudioInputDescriptor::open(const audio_config_t *config,
-                                       audio_devices_t device,
-                                       const String8& address,
+                                       const sp<DeviceDescriptor> &device,
                                        audio_source_t source,
                                        audio_input_flags_t flags,
                                        audio_io_handle_t *input)
@@ -198,24 +201,26 @@
 
     mDevice = device;
 
-    ALOGV("opening input for device %08x address %s profile %p name %s",
-          mDevice, address.string(), mProfile.get(), mProfile->getName().string());
+    ALOGV("opening input for device %s profile %p name %s",
+          mDevice->toString().c_str(), mProfile.get(), mProfile->getName().string());
+
+    audio_devices_t deviceType = mDevice->type();
 
     status_t status = mClientInterface->openInput(mProfile->getModuleHandle(),
                                                   input,
                                                   &lConfig,
-                                                  &mDevice,
-                                                  address,
+                                                  &deviceType,
+                                                  mDevice->address(),
                                                   source,
                                                   flags);
-    LOG_ALWAYS_FATAL_IF(mDevice != device,
+    LOG_ALWAYS_FATAL_IF(mDevice->type() != deviceType,
                         "%s openInput returned device %08x when given device %08x",
-                        __FUNCTION__, mDevice, device);
+                        __FUNCTION__, mDevice->type(), deviceType);
 
     if (status == NO_ERROR) {
         LOG_ALWAYS_FATAL_IF(*input == AUDIO_IO_HANDLE_NONE,
-                            "%s openInput returned input handle %d for device %08x",
-                            __FUNCTION__, *input, device);
+                            "%s openInput returned input handle %d for device %s",
+                            __FUNCTION__, *input, mDevice->toString().c_str());
         mSamplingRate = lConfig.sample_rate;
         mChannelMask = lConfig.channel_mask;
         mFormat = lConfig.format;
@@ -423,7 +428,7 @@
     dst->appendFormat(" Sampling rate: %d\n", mSamplingRate);
     dst->appendFormat(" Format: %d\n", mFormat);
     dst->appendFormat(" Channels: %08x\n", mChannelMask);
-    dst->appendFormat(" Devices %08x\n", mDevice);
+    dst->appendFormat(" Devices %s\n", mDevice->toString().c_str());
     getEnabledEffects().dump(dst, 1 /*spaces*/, false /*verbose*/);
     dst->append(" AudioRecord Clients:\n");
     ClientMapHandler<RecordClientDescriptor>::dump(dst);
@@ -452,14 +457,13 @@
     return NULL;
 }
 
-uint32_t AudioInputCollection::activeInputsCountOnDevices(audio_devices_t devices) const
+uint32_t AudioInputCollection::activeInputsCountOnDevices(const DeviceVector &devices) const
 {
     uint32_t count = 0;
     for (size_t i = 0; i < size(); i++) {
         const sp<AudioInputDescriptor>  inputDescriptor = valueAt(i);
         if (inputDescriptor->isActive() &&
-                ((devices == AUDIO_DEVICE_IN_DEFAULT) ||
-                 ((inputDescriptor->mDevice & devices & ~AUDIO_DEVICE_BIT_IN) != 0))) {
+                (devices.isEmpty() || devices.contains(inputDescriptor->getDevice()))) {
             count++;
         }
     }
@@ -479,13 +483,6 @@
     return activeInputs;
 }
 
-audio_devices_t AudioInputCollection::getSupportedDevices(audio_io_handle_t handle) const
-{
-    sp<AudioInputDescriptor> inputDesc = valueFor(handle);
-    audio_devices_t devices = inputDesc->mProfile->getSupportedDevicesType();
-    return devices;
-}
-
 sp<AudioInputDescriptor> AudioInputCollection::getInputForClient(audio_port_handle_t portId)
 {
     for (size_t i = 0; i < size(); i++) {
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 97504ab..643cbd1 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -82,25 +82,10 @@
     return mId;
 }
 
-audio_devices_t AudioOutputDescriptor::device() const
-{
-    return mDevice;
-}
-
-audio_devices_t AudioOutputDescriptor::supportedDevices()
-{
-    return mDevice;
-}
-
 bool AudioOutputDescriptor::sharesHwModuleWith(
         const sp<AudioOutputDescriptor>& outputDesc)
 {
-    if (outputDesc->isDuplicated()) {
-        return sharesHwModuleWith(outputDesc->subOutput1()) ||
-                    sharesHwModuleWith(outputDesc->subOutput2());
-    } else {
-        return hasSameHwModuleAs(outputDesc);
-    }
+    return hasSameHwModuleAs(outputDesc);
 }
 
 void AudioOutputDescriptor::changeStreamActiveCount(const sp<TrackClientDescriptor>& client,
@@ -282,7 +267,7 @@
     dst->appendFormat(" Sampling rate: %d\n", mSamplingRate);
     dst->appendFormat(" Format: %08x\n", mFormat);
     dst->appendFormat(" Channels: %08x\n", mChannelMask);
-    dst->appendFormat(" Devices: %08x\n", device());
+    dst->appendFormat(" Devices: %s\n", devices().toString().c_str());
     dst->appendFormat(" Global active count: %u\n", mGlobalActiveCount);
     dst->append(" Stream volume activeCount muteCount\n");
     for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
@@ -330,17 +315,18 @@
     AudioOutputDescriptor::dump(dst);
 }
 
-audio_devices_t SwAudioOutputDescriptor::device() const
+DeviceVector SwAudioOutputDescriptor::devices() const
 {
     if (isDuplicated()) {
-        return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
-    } else {
-        return mDevice;
+        DeviceVector devices = mOutput1->devices();
+        devices.merge(mOutput2->devices());
+        return devices;
     }
+    return mDevices;
 }
 
 bool SwAudioOutputDescriptor::sharesHwModuleWith(
-        const sp<AudioOutputDescriptor>& outputDesc)
+        const sp<SwAudioOutputDescriptor>& outputDesc)
 {
     if (isDuplicated()) {
         return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
@@ -352,13 +338,30 @@
     }
 }
 
-audio_devices_t SwAudioOutputDescriptor::supportedDevices()
+DeviceVector SwAudioOutputDescriptor::supportedDevices() const
 {
     if (isDuplicated()) {
-        return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
-    } else {
-        return mProfile->getSupportedDevicesType();
+        DeviceVector supportedDevices = mOutput1->supportedDevices();
+        supportedDevices.merge(mOutput2->supportedDevices());
+        return supportedDevices;
     }
+    return mProfile->getSupportedDevices();
+}
+
+bool SwAudioOutputDescriptor::supportsDevice(const sp<DeviceDescriptor> &device) const
+{
+    return supportedDevices().contains(device);
+}
+
+bool SwAudioOutputDescriptor::supportsAllDevices(const DeviceVector &devices) const
+{
+    return supportedDevices().containsAllDevices(devices);
+}
+
+DeviceVector SwAudioOutputDescriptor::filterSupportedDevices(const DeviceVector &devices) const
+{
+    DeviceVector filteredDevices = supportedDevices();
+    return filteredDevices.filter(devices);
 }
 
 uint32_t SwAudioOutputDescriptor::latency()
@@ -443,12 +446,15 @@
 }
 
 status_t SwAudioOutputDescriptor::open(const audio_config_t *config,
-                                       audio_devices_t device,
-                                       const String8& address,
+                                       const DeviceVector &devices,
                                        audio_stream_type_t stream,
                                        audio_output_flags_t flags,
                                        audio_io_handle_t *output)
 {
+    mDevices = devices;
+    const String8& address = devices.getFirstValidAddress();
+    audio_devices_t device = devices.types();
+
     audio_config_t lConfig;
     if (config == nullptr) {
         lConfig = AUDIO_CONFIG_INITIALIZER;
@@ -459,7 +465,6 @@
         lConfig = *config;
     }
 
-    mDevice = device;
     // if the selected profile is offloaded and no offload info was specified,
     // create a default one
     if ((mProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
@@ -477,19 +482,19 @@
 
     mFlags = (audio_output_flags_t)(mFlags | flags);
 
-    ALOGV("opening output for device %08x address %s profile %p name %s",
-          mDevice, address.string(), mProfile.get(), mProfile->getName().string());
+    ALOGV("opening output for device %s profile %p name %s",
+          mDevices.toString().c_str(), mProfile.get(), mProfile->getName().string());
 
     status_t status = mClientInterface->openOutput(mProfile->getModuleHandle(),
                                                    output,
                                                    &lConfig,
-                                                   &mDevice,
+                                                   &device,
                                                    address,
                                                    &mLatency,
                                                    mFlags);
-    LOG_ALWAYS_FATAL_IF(mDevice != device,
+    LOG_ALWAYS_FATAL_IF(mDevices.types() != device,
                         "%s openOutput returned device %08x when given device %08x",
-                        __FUNCTION__, mDevice, device);
+                        __FUNCTION__, mDevices.types(), device);
 
     if (status == NO_ERROR) {
         LOG_ALWAYS_FATAL_IF(*output == AUDIO_IO_HANDLE_NONE,
@@ -605,11 +610,6 @@
     mSource->dump(dst, 0, 0);
 }
 
-audio_devices_t HwAudioOutputDescriptor::supportedDevices()
-{
-    return mDevice;
-}
-
 void HwAudioOutputDescriptor::toAudioPortConfig(
                                                  struct audio_port_config *dstConfig,
                                                  const struct audio_port_config *srcConfig) const
@@ -657,7 +657,7 @@
     for (size_t i = 0; i < this->size(); i++) {
         const sp<SwAudioOutputDescriptor> outputDesc = this->valueAt(i);
         if (outputDesc->isStreamActive(stream, inPastMs, sysTime)
-                && ((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) == 0)) {
+                && ((outputDesc->devices().types() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) == 0)) {
             return true;
         }
     }
@@ -670,7 +670,7 @@
     nsecs_t sysTime = systemTime();
     for (size_t i = 0; i < size(); i++) {
         const sp<SwAudioOutputDescriptor> outputDesc = valueAt(i);
-        if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
+        if (((outputDesc->devices().types() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
                 outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
             // do not consider re routing (when the output is going to a dynamic policy)
             // as "remote playback"
@@ -686,7 +686,8 @@
 {
     for (size_t i = 0; i < size(); i++) {
         sp<SwAudioOutputDescriptor> outputDesc = valueAt(i);
-        if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) {
+        if (!outputDesc->isDuplicated() &&
+                outputDesc->devices().types() & AUDIO_DEVICE_OUT_ALL_A2DP) {
             return this->keyAt(i);
         }
     }
@@ -700,10 +701,9 @@
     if ((primaryOutput != NULL) && (primaryOutput->mProfile != NULL)
         && (primaryOutput->mProfile->getModule() != NULL)) {
         sp<HwModule> primaryHwModule = primaryOutput->mProfile->getModule();
-        Vector <sp<IOProfile>> primaryHwModuleOutputProfiles =
-                                   primaryHwModule->getOutputProfiles();
-        for (size_t i = 0; i < primaryHwModuleOutputProfiles.size(); i++) {
-            if (primaryHwModuleOutputProfiles[i]->supportDevice(AUDIO_DEVICE_OUT_ALL_A2DP)) {
+
+        for (const auto &outputProfile : primaryHwModule->getOutputProfiles()) {
+            if (outputProfile->supportsDeviceTypes(AUDIO_DEVICE_OUT_ALL_A2DP)) {
                 return true;
             }
         }
@@ -754,13 +754,6 @@
     return false;
 }
 
-audio_devices_t SwAudioOutputCollection::getSupportedDevices(audio_io_handle_t handle) const
-{
-    sp<SwAudioOutputDescriptor> outputDesc = valueFor(handle);
-    audio_devices_t devices = outputDesc->mProfile->getSupportedDevicesType();
-    return devices;
-}
-
 sp<SwAudioOutputDescriptor> SwAudioOutputCollection::getOutputForClient(audio_port_handle_t portId)
 {
     for (size_t i = 0; i < size(); i++) {
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
index 4d0916e..d18091c 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
@@ -280,13 +280,11 @@
     return BAD_VALUE;
 }
 
-audio_devices_t AudioPolicyMixCollection::getDeviceAndMixForInputSource(audio_source_t inputSource,
-                                                                        audio_devices_t availDevices,
-                                                                        AudioMix **policyMix)
+sp<DeviceDescriptor> AudioPolicyMixCollection::getDeviceAndMixForInputSource(
+        audio_source_t inputSource, const DeviceVector &availDevices, AudioMix **policyMix)
 {
     for (size_t i = 0; i < size(); i++) {
         AudioMix *mix = valueAt(i)->getMix();
-
         if (mix->mMixType != MIX_TYPE_RECORDERS) {
             continue;
         }
@@ -295,17 +293,21 @@
                     mix->mCriteria[j].mValue.mSource == inputSource) ||
                (RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET == mix->mCriteria[j].mRule &&
                     mix->mCriteria[j].mValue.mSource != inputSource)) {
-                if (availDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
+                // assuming PolicyMix only for remote submix for input
+                // so mix->mDeviceType can only be AUDIO_DEVICE_OUT_REMOTE_SUBMIX
+                audio_devices_t device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+                auto mixDevice = availDevices.getDevice(device, mix->mDeviceAddress);
+                if (mixDevice != nullptr) {
                     if (policyMix != NULL) {
                         *policyMix = mix;
                     }
-                    return AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+                    return mixDevice;
                 }
                 break;
             }
         }
     }
-    return AUDIO_DEVICE_NONE;
+    return nullptr;
 }
 
 status_t AudioPolicyMixCollection::getInputMixForAttr(audio_attributes_t attr, AudioMix **policyMix)
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
index 19dde6a..9fcf5e7 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
@@ -31,9 +31,15 @@
 // --- AudioPort class implementation
 void AudioPort::attach(const sp<HwModule>& module)
 {
+    ALOGV("%s: attaching module %s to port %s", __FUNCTION__, getModuleName(), mName.string());
     mModule = module;
 }
 
+void AudioPort::detach()
+{
+    mModule = nullptr;
+}
+
 // Note that is a different namespace than AudioFlinger unique IDs
 audio_port_handle_t AudioPort::getNextUniqueId()
 {
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
index 04cbcd1..01111c5 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -58,6 +58,12 @@
     mId = getNextUniqueId();
 }
 
+void DeviceDescriptor::detach()
+{
+    mId = AUDIO_PORT_HANDLE_NONE;
+    AudioPort::detach();
+}
+
 bool DeviceDescriptor::equals(const sp<DeviceDescriptor>& other) const
 {
     // Devices are considered equal if they:
diff --git a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
index 80af88d..7d2d094 100644
--- a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
@@ -52,6 +52,9 @@
 
     sp<DeviceDescriptor> devDesc = new DeviceDescriptor(device);
     devDesc->setAddress(address);
+    addDynamicDevice(devDesc);
+    // Reciprocally attach the device to the module
+    devDesc->attach(this);
     profile->addSupportedDevice(devDesc);
 
     return addOutputProfile(profile);
@@ -97,6 +100,9 @@
 {
     for (size_t i = 0; i < mOutputProfiles.size(); i++) {
         if (mOutputProfiles[i]->getName() == name) {
+            for (const auto &device : mOutputProfiles[i]->getSupportedDevices()) {
+                removeDynamicDevice(device);
+            }
             mOutputProfiles.removeAt(i);
             break;
         }
@@ -114,6 +120,9 @@
 
     sp<DeviceDescriptor> devDesc = new DeviceDescriptor(device);
     devDesc->setAddress(address);
+    addDynamicDevice(devDesc);
+    // Reciprocally attach the device to the module
+    devDesc->attach(this);
     profile->addSupportedDevice(devDesc);
 
     ALOGV("addInputProfile() name %s rate %d mask 0x%08x",
@@ -126,6 +135,9 @@
 {
     for (size_t i = 0; i < mInputProfiles.size(); i++) {
         if (mInputProfiles[i]->getName() == name) {
+            for (const auto &device : mInputProfiles[i]->getSupportedDevices()) {
+                removeDynamicDevice(device);
+            }
             mInputProfiles.removeAt(i);
             break;
         }
@@ -247,6 +259,7 @@
         }
     }
     mDeclaredDevices.dump(dst, String8("Declared"), 2, true);
+    mDynamicDevices.dump(dst, String8("Dynamic"),  2, true);
     mRoutes.dump(dst, 2);
 }
 
@@ -260,13 +273,13 @@
     return nullptr;
 }
 
-sp <HwModule> HwModuleCollection::getModuleForDevice(audio_devices_t device) const
+sp <HwModule> HwModuleCollection::getModuleForDeviceTypes(audio_devices_t device) const
 {
     for (const auto& module : *this) {
         const auto& profiles = audio_is_output_device(device) ?
                 module->getOutputProfiles() : module->getInputProfiles();
         for (const auto& profile : profiles) {
-            if (profile->supportDevice(device)) {
+            if (profile->supportsDeviceTypes(device)) {
                 return module;
             }
         }
@@ -274,30 +287,127 @@
     return nullptr;
 }
 
-sp<DeviceDescriptor> HwModuleCollection::getDeviceDescriptor(const audio_devices_t device,
-                                                             const char *device_address,
-                                                             const char *device_name,
+sp <HwModule> HwModuleCollection::getModuleForDevice(const sp<DeviceDescriptor> &device) const
+{
+    for (const auto& module : *this) {
+        const auto& profiles = audio_is_output_device(device->type()) ?
+                module->getOutputProfiles() : module->getInputProfiles();
+        for (const auto& profile : profiles) {
+            if (profile->supportsDevice(device)) {
+                return module;
+            }
+        }
+    }
+    return nullptr;
+}
+
+DeviceVector HwModuleCollection::getAvailableDevicesFromModuleName(
+        const char *name, const DeviceVector &availableDevices) const
+{
+    sp<HwModule> module = getModuleFromName(name);
+    if (module == nullptr) {
+        return DeviceVector();
+    }
+    return availableDevices.getDevicesFromHwModule(module->getHandle());
+}
+
+sp<DeviceDescriptor> HwModuleCollection::getDeviceDescriptor(const audio_devices_t deviceType,
+                                                             const char *address,
+                                                             const char *name,
+                                                             bool allowToCreate,
                                                              bool matchAddress) const
 {
-    String8 address = (device_address == nullptr || !matchAddress) ?
-            String8("") : String8(device_address);
+    String8 devAddress = (address == nullptr || !matchAddress) ? String8("") : String8(address);
     // handle legacy remote submix case where the address was not always specified
-    if (device_distinguishes_on_address(device) && (address.length() == 0)) {
-        address = String8("0");
+    if (device_distinguishes_on_address(deviceType) && (devAddress.length() == 0)) {
+        devAddress = String8("0");
     }
 
     for (const auto& hwModule : *this) {
-        DeviceVector declaredDevices = hwModule->getDeclaredDevices();
-        sp<DeviceDescriptor> deviceDesc = declaredDevices.getDevice(device, address);
-        if (deviceDesc) {
-            return deviceDesc;
+        DeviceVector moduleDevices = hwModule->getAllDevices();
+        auto moduleDevice = moduleDevices.getDevice(deviceType, devAddress);
+        if (moduleDevice) {
+            if (allowToCreate) {
+                moduleDevice->attach(hwModule);
+            }
+            return moduleDevice;
         }
     }
+    if (!allowToCreate) {
+        ALOGE("%s: could not find HW module for device %s %04x address %s", __FUNCTION__,
+              name, deviceType, address);
+        return nullptr;
+    }
+    return createDevice(deviceType, address, name);
+}
 
-    sp<DeviceDescriptor> devDesc = new DeviceDescriptor(device);
-    devDesc->setName(String8(device_name));
-    devDesc->setAddress(address);
-    return devDesc;
+sp<DeviceDescriptor> HwModuleCollection::createDevice(const audio_devices_t type,
+                                                      const char *address,
+                                                      const char *name) const
+{
+    sp<HwModule> hwModule = getModuleForDeviceTypes(type);
+    if (hwModule == 0) {
+        ALOGE("%s: could not find HW module for device %04x address %s", __FUNCTION__, type,
+              address);
+        return nullptr;
+    }
+    sp<DeviceDescriptor> device = new DeviceDescriptor(type, String8(name));
+    device->setName(String8(name));
+    device->setAddress(String8(address));
+
+    // Add the device to the list of dynamic devices
+    hwModule->addDynamicDevice(device);
+    // Reciprocally attach the device to the module
+    device->attach(hwModule);
+    ALOGD("%s: adding dynamic device %s to module %s", __FUNCTION__,
+          device->toString().c_str(), hwModule->getName());
+
+    const auto &profiles = (audio_is_output_device(type) ? hwModule->getOutputProfiles() :
+                                                             hwModule->getInputProfiles());
+    for (const auto &profile : profiles) {
+        // Add the device as supported to all profile supporting "weakly" or not the device
+        // according to its type
+        if (profile->supportsDevice(device, false /*matchAdress*/)) {
+
+            // @todo quid of audio profile? import the profile from device of the same type?
+            const auto &isoTypeDeviceForProfile = profile->getSupportedDevices().getDevice(type);
+            device->importAudioPort(isoTypeDeviceForProfile, true /* force */);
+
+            ALOGV("%s: adding device %s to profile %s", __FUNCTION__,
+                  device->toString().c_str(), profile->getTagName().c_str());
+            profile->addSupportedDevice(device);
+        }
+    }
+    return device;
+}
+
+void HwModuleCollection::cleanUpForDevice(const sp<DeviceDescriptor> &device)
+{
+    for (const auto& hwModule : *this) {
+        DeviceVector moduleDevices = hwModule->getAllDevices();
+        if (!moduleDevices.contains(device)) {
+            continue;
+        }
+        device->detach();
+        // Only remove from dynamic list, not from declared list!!!
+        if (!hwModule->getDynamicDevices().contains(device)) {
+            return;
+        }
+        hwModule->removeDynamicDevice(device);
+        ALOGV("%s: removed dynamic device %s from module %s", __FUNCTION__,
+              device->toString().c_str(), hwModule->getName());
+
+        const IOProfileCollection &profiles = audio_is_output_device(device->type()) ?
+                    hwModule->getOutputProfiles() : hwModule->getInputProfiles();
+        for (const auto &profile : profiles) {
+            // For cleanup, strong match is required
+            if (profile->supportsDevice(device, true /*matchAdress*/)) {
+                ALOGV("%s: removing device %s from profile %s", __FUNCTION__,
+                      device->toString().c_str(), profile->getTagName().c_str());
+                profile->removeSupportedDevice(device);
+            }
+        }
+    }
 }
 
 void HwModuleCollection::dump(String8 *dst) const
diff --git a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
index 3788244..fe2eaee 100644
--- a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
@@ -25,11 +25,7 @@
 
 namespace android {
 
-// checks if the IO profile is compatible with specified parameters.
-// Sampling rate, format and channel mask must be specified in order to
-// get a valid a match
-bool IOProfile::isCompatibleProfile(audio_devices_t device,
-                                    const String8& address,
+bool IOProfile::isCompatibleProfile(const DeviceVector &devices,
                                     uint32_t samplingRate,
                                     uint32_t *updatedSamplingRate,
                                     audio_format_t format,
@@ -46,14 +42,8 @@
             getType() == AUDIO_PORT_TYPE_MIX && getRole() == AUDIO_PORT_ROLE_SINK;
     ALOG_ASSERT(isPlaybackThread != isRecordThread);
 
-
-    if (device != AUDIO_DEVICE_NONE) {
-        // just check types if multiple devices are selected
-        if (popcount(device & ~AUDIO_DEVICE_BIT_IN) > 1) {
-            if ((mSupportedDevices.types() & device) != device) {
-                return false;
-            }
-        } else if (mSupportedDevices.getDevice(device, address) == 0) {
+    if (!devices.isEmpty()) {
+        if (!mSupportedDevices.containsAllDevices(devices)) {
             return false;
         }
     }
diff --git a/services/audiopolicy/engineconfigurable/wrapper/ParameterManagerWrapper.cpp b/services/audiopolicy/engineconfigurable/wrapper/ParameterManagerWrapper.cpp
index fc6c1e4..1934fa4 100644
--- a/services/audiopolicy/engineconfigurable/wrapper/ParameterManagerWrapper.cpp
+++ b/services/audiopolicy/engineconfigurable/wrapper/ParameterManagerWrapper.cpp
@@ -295,8 +295,8 @@
 
     auto criterionType = criterion->getCriterionType();
     int deviceAddressId;
-    if (not criterionType->getNumericalValue(devDesc->mAddress.string(), deviceAddressId)) {
-        ALOGE("%s: unknown device address reported (%s)", __FUNCTION__, devDesc->mAddress.c_str());
+    if (not criterionType->getNumericalValue(devDesc->address().string(), deviceAddressId)) {
+        ALOGW("%s: unknown device address reported (%s)", __FUNCTION__, devDesc->address().c_str());
         return BAD_TYPE;
     }
     int currentValueMask = criterion->getCriterionState();
diff --git a/services/audiopolicy/enginedefault/src/Engine.cpp b/services/audiopolicy/enginedefault/src/Engine.cpp
index 0ef6f52..3d68cd8 100644
--- a/services/audiopolicy/enginedefault/src/Engine.cpp
+++ b/services/audiopolicy/enginedefault/src/Engine.cpp
@@ -322,7 +322,7 @@
             // a primary device
             // FIXME: this is not the right way of solving this problem
             audio_devices_t availPrimaryOutputDevices =
-                (primaryOutput->supportedDevices() | AUDIO_DEVICE_OUT_HEARING_AID) &
+                (primaryOutput->supportedDevices().types() | AUDIO_DEVICE_OUT_HEARING_AID) &
                 availableOutputDevices.types();
 
             if (((availableInputDevices.types() &
@@ -475,7 +475,7 @@
             // compressed format as they would likely not be mixed and dropped.
             for (size_t i = 0; i < outputs.size(); i++) {
                 sp<AudioOutputDescriptor> desc = outputs.valueAt(i);
-                audio_devices_t devices = desc->device() &
+                audio_devices_t devices = desc->devices().types() &
                     (AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_HDMI_ARC);
                 if (desc->isActive() && !audio_is_linear_pcm(desc->mFormat) &&
                         devices != AUDIO_DEVICE_NONE) {
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 5544821..5c8a799 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -88,36 +88,39 @@
     return status;
 }
 
-void AudioPolicyManager::broadcastDeviceConnectionState(audio_devices_t device,
-                                                        audio_policy_dev_state_t state,
-                                                        const String8 &device_address)
+void AudioPolicyManager::broadcastDeviceConnectionState(const sp<DeviceDescriptor> &device,
+                                                        audio_policy_dev_state_t state)
 {
-    AudioParameter param(device_address);
+    AudioParameter param(device->address());
     const String8 key(state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE ?
                 AudioParameter::keyStreamConnect : AudioParameter::keyStreamDisconnect);
-    param.addInt(key, device);
+    param.addInt(key, device->type());
     mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
 }
 
-status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device,
+status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t deviceType,
                                                          audio_policy_dev_state_t state,
                                                          const char *device_address,
                                                          const char *device_name)
 {
     ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s",
-            device, state, device_address, device_name);
+            deviceType, state, device_address, device_name);
 
     // connect/disconnect only 1 device at a time
-    if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
+    if (!audio_is_output_device(deviceType) && !audio_is_input_device(deviceType)) return BAD_VALUE;
 
-    sp<DeviceDescriptor> devDesc =
-            mHwModules.getDeviceDescriptor(device, device_address, device_name);
+    sp<DeviceDescriptor> device =
+            mHwModules.getDeviceDescriptor(deviceType, device_address, device_name,
+                                           state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
+    if (device == 0) {
+        return INVALID_OPERATION;
+    }
 
     // handle output devices
-    if (audio_is_output_device(device)) {
+    if (audio_is_output_device(deviceType)) {
         SortedVector <audio_io_handle_t> outputs;
 
-        ssize_t index = mAvailableOutputDevices.indexOf(devDesc);
+        ssize_t index = mAvailableOutputDevices.indexOf(device);
 
         // save a copy of the opened output descriptors before any output is opened or closed
         // by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
@@ -127,70 +130,60 @@
         // handle output device connection
         case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
             if (index >= 0) {
-                ALOGW("setDeviceConnectionState() device already connected: %x", device);
+                ALOGW("%s() device already connected: %s", __func__, device->toString().c_str());
                 return INVALID_OPERATION;
             }
-            ALOGV("setDeviceConnectionState() connecting device %x", device);
+            ALOGV("%s() connecting device %s", __func__, device->toString().c_str());
 
             // register new device as available
-            index = mAvailableOutputDevices.add(devDesc);
-            if (index >= 0) {
-                sp<HwModule> module = mHwModules.getModuleForDevice(device);
-                if (module == 0) {
-                    ALOGD("setDeviceConnectionState() could not find HW module for device %08x",
-                          device);
-                    mAvailableOutputDevices.remove(devDesc);
-                    return INVALID_OPERATION;
-                }
-                mAvailableOutputDevices[index]->attach(module);
-            } else {
+            if (mAvailableOutputDevices.add(device) < 0) {
                 return NO_MEMORY;
             }
 
             // Before checking outputs, broadcast connect event to allow HAL to retrieve dynamic
             // parameters on newly connected devices (instead of opening the outputs...)
-            broadcastDeviceConnectionState(device, state, devDesc->address());
+            broadcastDeviceConnectionState(device, state);
 
-            if (checkOutputsForDevice(devDesc, state, outputs, devDesc->address()) != NO_ERROR) {
-                mAvailableOutputDevices.remove(devDesc);
+            if (checkOutputsForDevice(device, state, outputs) != NO_ERROR) {
+                mAvailableOutputDevices.remove(device);
 
-                broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
-                                               devDesc->address());
+                mHwModules.cleanUpForDevice(device);
+
+                broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE);
                 return INVALID_OPERATION;
             }
             // Propagate device availability to Engine
-            mEngine->setDeviceConnectionState(devDesc, state);
+            mEngine->setDeviceConnectionState(device, state);
 
             // outputs should never be empty here
             ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():"
                     "checkOutputsForDevice() returned no outputs but status OK");
-            ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
-                  outputs.size());
+            ALOGV("%s() checkOutputsForDevice() returned %zu outputs", __func__, outputs.size());
 
             } break;
         // handle output device disconnection
         case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
             if (index < 0) {
-                ALOGW("setDeviceConnectionState() device not connected: %x", device);
+                ALOGW("%s() device not connected: %s", __func__, device->toString().c_str());
                 return INVALID_OPERATION;
             }
 
-            ALOGV("setDeviceConnectionState() disconnecting output device %x", device);
+            ALOGV("%s() disconnecting output device %s", __func__, device->toString().c_str());
 
             // Send Disconnect to HALs
-            broadcastDeviceConnectionState(device, state, devDesc->address());
+            broadcastDeviceConnectionState(device, state);
 
             // remove device from available output devices
-            mAvailableOutputDevices.remove(devDesc);
+            mAvailableOutputDevices.remove(device);
 
-            checkOutputsForDevice(devDesc, state, outputs, devDesc->address());
+            checkOutputsForDevice(device, state, outputs);
 
             // Propagate device availability to Engine
-            mEngine->setDeviceConnectionState(devDesc, state);
+            mEngine->setDeviceConnectionState(device, state);
             } break;
 
         default:
-            ALOGE("setDeviceConnectionState() invalid state: %x", state);
+            ALOGE("%s() invalid state: %x", __func__, state);
             return BAD_VALUE;
         }
 
@@ -199,8 +192,8 @@
             if (!outputs.isEmpty()) {
                 for (audio_io_handle_t output : outputs) {
                     sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(output);
-                    // close unused outputs after device disconnection or direct outputs that have been
-                    // opened by checkOutputsForDevice() to query dynamic parameters
+                    // close unused outputs after device disconnection or direct outputs that have
+                    // been opened by checkOutputsForDevice() to query dynamic parameters
                     if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
                             (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
                              (desc->mDirectOpenCount == 0))) {
@@ -214,29 +207,28 @@
         });
 
         if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
-            audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
-            updateCallRouting(newDevice);
+            DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/);
+            updateCallRouting(newDevices);
         }
-        const audio_devices_t msdOutDevice = getModuleDeviceTypes(
-                mAvailableOutputDevices, AUDIO_HARDWARE_MODULE_ID_MSD);
+        const DeviceVector msdOutDevices = getMsdAudioOutDevices();
         for (size_t i = 0; i < mOutputs.size(); i++) {
             sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
             if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) {
-                audio_devices_t newDevice = getNewOutputDevice(desc, true /*fromCache*/);
+                DeviceVector newDevices = getNewOutputDevices(desc, true /*fromCache*/);
                 // do not force device change on duplicated output because if device is 0, it will
                 // also force a device 0 for the two outputs it is duplicated to which may override
                 // a valid device selection on those outputs.
-                bool force = (msdOutDevice == AUDIO_DEVICE_NONE || msdOutDevice != desc->device())
+                bool force = (msdOutDevices.isEmpty() || msdOutDevices != desc->devices())
                         && !desc->isDuplicated()
-                        && (!device_distinguishes_on_address(device)
+                        && (!device_distinguishes_on_address(deviceType)
                                 // always force when disconnecting (a non-duplicated device)
                                 || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
-                setOutputDevice(desc, newDevice, force, 0);
+                setOutputDevices(desc, newDevices, force, 0);
             }
         }
 
         if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
-            cleanUpForDevice(devDesc);
+            cleanUpForDevice(device);
         }
 
         mpClientInterface->onAudioPortListUpdate();
@@ -244,67 +236,59 @@
     }  // end if is output device
 
     // handle input devices
-    if (audio_is_input_device(device)) {
+    if (audio_is_input_device(deviceType)) {
         SortedVector <audio_io_handle_t> inputs;
 
-        ssize_t index = mAvailableInputDevices.indexOf(devDesc);
+        ssize_t index = mAvailableInputDevices.indexOf(device);
         switch (state)
         {
         // handle input device connection
         case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
             if (index >= 0) {
-                ALOGW("setDeviceConnectionState() device already connected: %d", device);
+                ALOGW("%s() device already connected: %s", __func__, device->toString().c_str());
                 return INVALID_OPERATION;
             }
-            sp<HwModule> module = mHwModules.getModuleForDevice(device);
-            if (module == NULL) {
-                ALOGW("setDeviceConnectionState(): could not find HW module for device %08x",
-                      device);
-                return INVALID_OPERATION;
-            }
-
             // Before checking intputs, broadcast connect event to allow HAL to retrieve dynamic
             // parameters on newly connected devices (instead of opening the inputs...)
-            broadcastDeviceConnectionState(device, state, devDesc->address());
+            broadcastDeviceConnectionState(device, state);
 
-            if (checkInputsForDevice(devDesc, state, inputs, devDesc->address()) != NO_ERROR) {
-                broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
-                                               devDesc->address());
+            if (checkInputsForDevice(device, state, inputs) != NO_ERROR) {
+                broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE);
+
+                mHwModules.cleanUpForDevice(device);
+
                 return INVALID_OPERATION;
             }
 
-            index = mAvailableInputDevices.add(devDesc);
-            if (index >= 0) {
-                mAvailableInputDevices[index]->attach(module);
-            } else {
+            if (mAvailableInputDevices.add(device) < 0) {
                 return NO_MEMORY;
             }
 
             // Propagate device availability to Engine
-            mEngine->setDeviceConnectionState(devDesc, state);
+            mEngine->setDeviceConnectionState(device, state);
         } break;
 
         // handle input device disconnection
         case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
             if (index < 0) {
-                ALOGW("setDeviceConnectionState() device not connected: %d", device);
+                ALOGW("%s() device not connected: %s", __func__, device->toString().c_str());
                 return INVALID_OPERATION;
             }
 
-            ALOGV("setDeviceConnectionState() disconnecting input device %x", device);
+            ALOGV("%s() disconnecting input device %s", __func__, device->toString().c_str());
 
             // Set Disconnect to HALs
-            broadcastDeviceConnectionState(device, state, devDesc->address());
+            broadcastDeviceConnectionState(device, state);
 
-            checkInputsForDevice(devDesc, state, inputs, devDesc->address());
-            mAvailableInputDevices.remove(devDesc);
+            checkInputsForDevice(device, state, inputs);
+            mAvailableInputDevices.remove(device);
 
             // Propagate device availability to Engine
-            mEngine->setDeviceConnectionState(devDesc, state);
+            mEngine->setDeviceConnectionState(device, state);
         } break;
 
         default:
-            ALOGE("setDeviceConnectionState() invalid state: %x", state);
+            ALOGE("%s() invalid state: %x", __func__, state);
             return BAD_VALUE;
         }
 
@@ -314,19 +298,19 @@
         updateDevicesAndOutputs();
 
         if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
-            audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
-            updateCallRouting(newDevice);
+            DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/);
+            updateCallRouting(newDevices);
         }
 
         if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
-            cleanUpForDevice(devDesc);
+            cleanUpForDevice(device);
         }
 
         mpClientInterface->onAudioPortListUpdate();
         return NO_ERROR;
     } // end if is input device
 
-    ALOGW("setDeviceConnectionState() invalid device: %x", device);
+    ALOGW("%s() invalid device: %s", __func__, device->toString().c_str());
     return BAD_VALUE;
 }
 
@@ -334,7 +318,7 @@
                                                                       const char *device_address)
 {
     sp<DeviceDescriptor> devDesc =
-            mHwModules.getDeviceDescriptor(device, device_address, "",
+            mHwModules.getDeviceDescriptor(device, device_address, "", false /* allowToCreate */,
                                            (strlen(device_address) != 0)/*matchAddress*/);
 
     if (devDesc == 0) {
@@ -350,7 +334,7 @@
     } else if (audio_is_input_device(device)) {
         deviceVector = &mAvailableInputDevices;
     } else {
-        ALOGW("getDeviceConnectionState() invalid device type %08x", device);
+        ALOGW("%s() invalid device type %08x", __func__, device);
         return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
     }
 
@@ -376,8 +360,7 @@
     // Check if the device is currently connected
     sp<DeviceDescriptor> devDesc =
             mHwModules.getDeviceDescriptor(device, device_address, device_name);
-    ssize_t index = mAvailableOutputDevices.indexOf(devDesc);
-    if (index < 0) {
+    if (devDesc == 0 || mAvailableOutputDevices.indexOf(devDesc) < 0) {
         // Nothing to do: device is not connected
         return NO_ERROR;
     }
@@ -425,16 +408,20 @@
     return NO_ERROR;
 }
 
-uint32_t AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, uint32_t delayMs)
+uint32_t AudioPolicyManager::updateCallRouting(const DeviceVector &rxDevices, uint32_t delayMs)
 {
     bool createTxPatch = false;
     uint32_t muteWaitMs = 0;
 
-    if(!hasPrimaryOutput() || mPrimaryOutput->device() == AUDIO_DEVICE_OUT_STUB) {
+    if(!hasPrimaryOutput() || mPrimaryOutput->devices().types() == AUDIO_DEVICE_OUT_STUB) {
         return muteWaitMs;
     }
-    audio_devices_t txDevice = getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
-    ALOGV("updateCallRouting device rxDevice %08x txDevice %08x", rxDevice, txDevice);
+    ALOG_ASSERT(!rxDevices.isEmpty(), "updateCallRouting() no selected output device");
+
+    audio_attributes_t attr = { .source = AUDIO_SOURCE_VOICE_COMMUNICATION };
+    auto txDevice = getDeviceAndMixForAttributes(attr);
+    ALOGV("updateCallRouting device rxDevice %s txDevice %s", 
+          rxDevices.toString().c_str(), txDevice->toString().c_str());
 
     // release existing RX patch if any
     if (mCallRxPatch != 0) {
@@ -450,16 +437,15 @@
     // If the RX device is on the primary HW module, then use legacy routing method for voice calls
     // via setOutputDevice() on primary output.
     // Otherwise, create two audio patches for TX and RX path.
-    if (availablePrimaryOutputDevices() & rxDevice) {
-        muteWaitMs = setOutputDevice(mPrimaryOutput, rxDevice, true, delayMs);
+    if (availablePrimaryOutputDevices().contains(rxDevices.itemAt(0))) {
+        muteWaitMs = setOutputDevices(mPrimaryOutput, rxDevices, true, delayMs);
         // If the TX device is also on the primary HW module, setOutputDevice() will take care
         // of it due to legacy implementation. If not, create a patch.
-        if ((availablePrimaryInputDevices() & txDevice & ~AUDIO_DEVICE_BIT_IN)
-                == AUDIO_DEVICE_NONE) {
+        if (!availablePrimaryModuleInputDevices().contains(txDevice)) {
             createTxPatch = true;
         }
     } else { // create RX path audio patch
-        mCallRxPatch = createTelephonyPatch(true /*isRx*/, rxDevice, delayMs);
+        mCallRxPatch = createTelephonyPatch(true /*isRx*/, rxDevices.itemAt(0), delayMs);
         createTxPatch = true;
     }
     if (createTxPatch) { // create TX path audio patch
@@ -470,26 +456,31 @@
 }
 
 sp<AudioPatch> AudioPolicyManager::createTelephonyPatch(
-        bool isRx, audio_devices_t device, uint32_t delayMs) {
+        bool isRx, const sp<DeviceDescriptor> &device, uint32_t delayMs) {
     PatchBuilder patchBuilder;
 
-    sp<DeviceDescriptor> txSourceDeviceDesc;
+    if (device == nullptr) {
+        return nullptr;
+    }
     if (isRx) {
-        patchBuilder.addSink(findDevice(mAvailableOutputDevices, device)).
-                addSource(findDevice(mAvailableInputDevices, AUDIO_DEVICE_IN_TELEPHONY_RX));
+        patchBuilder.addSink(device).
+                addSource(mAvailableInputDevices.getDevice(AUDIO_DEVICE_IN_TELEPHONY_RX));
     } else {
-        patchBuilder.addSource(txSourceDeviceDesc = findDevice(mAvailableInputDevices, device)).
-                addSink(findDevice(mAvailableOutputDevices, AUDIO_DEVICE_OUT_TELEPHONY_TX));
+        patchBuilder.addSource(device).
+                addSink(mAvailableOutputDevices.getDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX));
     }
 
-    audio_devices_t outputDevice = isRx ? device : AUDIO_DEVICE_OUT_TELEPHONY_TX;
-    SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(outputDevice, mOutputs);
-    audio_io_handle_t output = selectOutput(outputs);
+    // @TODO: still ignoring the address, or not dealing platform with mutliple telephonydevices
+    const sp<DeviceDescriptor> outputDevice = isRx ?
+                device : mAvailableOutputDevices.getDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX);
+    SortedVector<audio_io_handle_t> outputs =
+            getOutputsForDevices(DeviceVector(outputDevice), mOutputs);
+    audio_io_handle_t output = selectOutput(outputs, AUDIO_OUTPUT_FLAG_NONE, AUDIO_FORMAT_INVALID);
     // request to reuse existing output stream if one is already opened to reach the target device
     if (output != AUDIO_IO_HANDLE_NONE) {
         sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
-        ALOG_ASSERT(!outputDesc->isDuplicated(),
-                "%s() %#x device output %d is duplicated", __func__, outputDevice, output);
+        ALOG_ASSERT(!outputDesc->isDuplicated(), "%s() %s device output %d is duplicated", __func__,
+                    outputDevice->toString().c_str(), output);
         patchBuilder.addSource(outputDesc, { .stream = AUDIO_STREAM_PATCH });
     }
 
@@ -499,7 +490,7 @@
         // call TX device but this information is not in the audio patch and logic here must be
         // symmetric to the one in startInput()
         for (const auto& activeDesc : mInputs.getActiveInputs()) {
-            if (activeDesc->hasSameHwModuleAs(txSourceDeviceDesc)) {
+            if (activeDesc->hasSameHwModuleAs(device)) {
                 closeActiveClients(activeDesc);
             }
         }
@@ -599,17 +590,17 @@
     }
 
     if (hasPrimaryOutput()) {
-        // Note that despite the fact that getNewOutputDevice() is called on the primary output,
+        // Note that despite the fact that getNewOutputDevices() is called on the primary output,
         // the device returned is not necessarily reachable via this output
-        audio_devices_t rxDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+        DeviceVector rxDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/);
         // force routing command to audio hardware when ending call
         // even if no device change is needed
-        if (isStateInCall(oldState) && rxDevice == AUDIO_DEVICE_NONE) {
-            rxDevice = mPrimaryOutput->device();
+        if (isStateInCall(oldState) && rxDevices.isEmpty()) {
+            rxDevices = mPrimaryOutput->devices();
         }
 
         if (state == AUDIO_MODE_IN_CALL) {
-            updateCallRouting(rxDevice, delayMs);
+            updateCallRouting(rxDevices, delayMs);
         } else if (oldState == AUDIO_MODE_IN_CALL) {
             if (mCallRxPatch != 0) {
                 mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
@@ -619,18 +610,18 @@
                 mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
                 mCallTxPatch.clear();
             }
-            setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
+            setOutputDevices(mPrimaryOutput, rxDevices, force, 0);
         } else {
-            setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
+            setOutputDevices(mPrimaryOutput, rxDevices, force, 0);
         }
     }
 
     // reevaluate routing on all outputs in case tracks have been started during the call
     for (size_t i = 0; i < mOutputs.size(); i++) {
         sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
-        audio_devices_t newDevice = getNewOutputDevice(desc, true /*fromCache*/);
+        DeviceVector newDevices = getNewOutputDevices(desc, true /*fromCache*/);
         if (state != AUDIO_MODE_IN_CALL || desc != mPrimaryOutput) {
-            setOutputDevice(desc, newDevice, (newDevice != AUDIO_DEVICE_NONE), 0 /*delayMs*/);
+            setOutputDevices(desc, newDevices, !newDevices.isEmpty(), 0 /*delayMs*/);
         }
     }
 
@@ -654,7 +645,7 @@
 }
 
 void AudioPolicyManager::setForceUse(audio_policy_force_use_t usage,
-                                         audio_policy_forced_cfg_t config)
+                                     audio_policy_forced_cfg_t config)
 {
     ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState());
     if (config == mEngine->getForceUse(usage)) {
@@ -680,26 +671,24 @@
         delayMs = TOUCH_SOUND_FIXED_DELAY_MS;
     }
     if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
-        audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/);
-        waitMs = updateCallRouting(newDevice, delayMs);
+        DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, true /*fromCache*/);
+        waitMs = updateCallRouting(newDevices, delayMs);
     }
     for (size_t i = 0; i < mOutputs.size(); i++) {
         sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
-        audio_devices_t newDevice = getNewOutputDevice(outputDesc, true /*fromCache*/);
+        DeviceVector newDevices = getNewOutputDevices(outputDesc, true /*fromCache*/);
         if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput)) {
-            waitMs = setOutputDevice(outputDesc, newDevice, (newDevice != AUDIO_DEVICE_NONE),
-                                     delayMs);
+            waitMs = setOutputDevices(outputDesc, newDevices, !newDevices.isEmpty(), delayMs);
         }
-        if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
-            applyStreamVolumes(outputDesc, newDevice, waitMs, true);
+        if (forceVolumeReeval && !newDevices.isEmpty()) {
+            applyStreamVolumes(outputDesc, newDevices.types(), waitMs, true);
         }
     }
 
     for (const auto& activeDesc : mInputs.getActiveInputs()) {
-        audio_devices_t newDevice = getNewInputDevice(activeDesc);
+        auto newDevice = getNewInputDevice(activeDesc);
         // Force new input selection if the new device can not be reached via current input
-        if (activeDesc->mProfile->getSupportedDevices().types() &
-                (newDevice & ~AUDIO_DEVICE_BIT_IN)) {
+        if (activeDesc->mProfile->getSupportedDevices().contains(newDevice)) {
             setInputDevice(activeDesc->mIoHandle, newDevice);
         } else {
             closeInput(activeDesc->mIoHandle);
@@ -715,7 +704,7 @@
 // Find an output profile compatible with the parameters passed. When "directOnly" is set, restrict
 // search to profiles for direct outputs.
 sp<IOProfile> AudioPolicyManager::getProfileForOutput(
-                                                   audio_devices_t device,
+                                                   const DeviceVector& devices,
                                                    uint32_t samplingRate,
                                                    audio_format_t format,
                                                    audio_channel_mask_t channelMask,
@@ -736,7 +725,7 @@
 
     for (const auto& hwModule : mHwModules) {
         for (const auto& curProfile : hwModule->getOutputProfiles()) {
-            if (!curProfile->isCompatibleProfile(device, String8(""),
+            if (!curProfile->isCompatibleProfile(devices,
                     samplingRate, NULL /*updatedSamplingRate*/,
                     format, NULL /*updatedFormat*/,
                     channelMask, NULL /*updatedChannelMask*/,
@@ -744,7 +733,7 @@
                 continue;
             }
             // reject profiles not corresponding to a device currently available
-            if ((mAvailableOutputDevices.types() & curProfile->getSupportedDevicesType()) == 0) {
+            if (!mAvailableOutputDevices.containsAtLeastOne(curProfile->getSupportedDevices())) {
                 continue;
             }
             if (!directOnly) return curProfile;
@@ -765,7 +754,7 @@
 audio_io_handle_t AudioPolicyManager::getOutput(audio_stream_type_t stream)
 {
     routing_strategy strategy = getStrategy(stream);
-    audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+    DeviceVector devices = getDevicesForStrategy(strategy, false /*fromCache*/);
 
     // Note that related method getOutputForAttr() uses getOutputForDevice() not selectOutput().
     // We use selectOutput() here since we don't have the desired AudioTrack sample rate,
@@ -773,10 +762,11 @@
     // getOutput() solely on audio_stream_type such as AudioSystem::getOutputFrameCount()
     // and AudioSystem::getOutputSamplingRate().
 
-    SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
-    audio_io_handle_t output = selectOutput(outputs);
+    SortedVector<audio_io_handle_t> outputs = getOutputsForDevices(devices, mOutputs);
+    audio_io_handle_t output = selectOutput(outputs, AUDIO_OUTPUT_FLAG_NONE, AUDIO_FORMAT_INVALID);
 
-    ALOGV("getOutput() stream %d selected device %08x, output %d", stream, device, output);
+    ALOGV("getOutput() stream %d selected devices %s, output %d", stream,
+          devices.toString().c_str(), output);
     return output;
 }
 
@@ -813,12 +803,11 @@
                                                  audio_output_flags_t *flags,
                                                  audio_port_handle_t *selectedDeviceId)
 {
-    DeviceVector outputDevices;
+    DeviceVector devices;
     routing_strategy strategy;
-    audio_devices_t device;
-    const audio_port_handle_t requestedDeviceId = *selectedDeviceId;
-    audio_devices_t msdDevice =
-            getModuleDeviceTypes(mAvailableOutputDevices, AUDIO_HARDWARE_MODULE_ID_MSD);
+    audio_devices_t deviceType = AUDIO_DEVICE_NONE;
+    const audio_port_handle_t requestedPortId = *selectedDeviceId;
+    DeviceVector msdDevices = getMsdAudioOutDevices();
 
     status_t status = getAudioAttributes(resultAttr, attr, *stream);
     if (status != NO_ERROR) {
@@ -829,17 +818,16 @@
           " session %d selectedDeviceId %d",
           __func__,
           resultAttr->usage, resultAttr->content_type, resultAttr->tags, resultAttr->flags,
-          session, requestedDeviceId);
+          session, requestedPortId);
 
     *stream = streamTypefromAttributesInt(resultAttr);
 
     strategy = getStrategyForAttr(resultAttr);
 
     // First check for explicit routing (eg. setPreferredDevice)
-    if (requestedDeviceId != AUDIO_PORT_HANDLE_NONE) {
-        sp<DeviceDescriptor> deviceDesc =
-            mAvailableOutputDevices.getDeviceFromId(requestedDeviceId);
-        device = deviceDesc->type();
+    sp<DeviceDescriptor> requestedDevice = mAvailableOutputDevices.getDeviceFromId(requestedPortId);
+    if (requestedDevice != nullptr) {
+        deviceType = requestedDevice->type();
     } else {
         // If no explict route, is there a matching dynamic policy that applies?
         sp<SwAudioOutputDescriptor> desc;
@@ -863,7 +851,7 @@
             ALOGW("%s no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE", __func__);
             return BAD_VALUE;
         }
-        device = getDeviceForStrategy(strategy, false /*fromCache*/);
+        deviceType = getDeviceForStrategy(strategy, false /*fromCache*/);
     }
 
     if ((resultAttr->flags & AUDIO_FLAG_HW_AV_SYNC) != 0) {
@@ -875,42 +863,44 @@
     // FIXME: provide a more generic approach which is not device specific and move this back
     // to getOutputForDevice.
     // TODO: Remove check of AUDIO_STREAM_MUSIC once migration is completed on the app side.
-    if (device == AUDIO_DEVICE_OUT_TELEPHONY_TX &&
+    if (deviceType == AUDIO_DEVICE_OUT_TELEPHONY_TX &&
         (*stream == AUDIO_STREAM_MUSIC || resultAttr->usage == AUDIO_USAGE_VOICE_COMMUNICATION) &&
         audio_is_linear_pcm(config->format) &&
         isInCall()) {
-        if (requestedDeviceId != AUDIO_PORT_HANDLE_NONE) {
+        if (requestedPortId != AUDIO_PORT_HANDLE_NONE) {
             *flags = (audio_output_flags_t)AUDIO_OUTPUT_FLAG_INCALL_MUSIC;
         } else {
             // Get the devce type directly from the engine to bypass preferred route logic
-            device = mEngine->getDeviceForStrategy(strategy);
+            deviceType = mEngine->getDeviceForStrategy(strategy);
         }
     }
 
     ALOGV("%s device 0x%x, sampling rate %d, format %#x, channel mask %#x, "
           "flags %#x",
-          __func__, device, config->sample_rate, config->format, config->channel_mask, *flags);
+          __func__,
+          deviceType, config->sample_rate, config->format, config->channel_mask, *flags);
 
     *output = AUDIO_IO_HANDLE_NONE;
-    if (msdDevice != AUDIO_DEVICE_NONE) {
-        *output = getOutputForDevice(msdDevice, session, *stream, config, flags);
-        if (*output != AUDIO_IO_HANDLE_NONE && setMsdPatch(device) == NO_ERROR) {
-            ALOGV("%s() Using MSD device 0x%x instead of device 0x%x",
-                    __func__, msdDevice, device);
-            device = msdDevice;
+    if (!msdDevices.isEmpty()) {
+        *output = getOutputForDevices(msdDevices, session, *stream, config, flags);
+        sp<DeviceDescriptor> deviceDesc = mAvailableOutputDevices.getDevice(deviceType);
+        if (*output != AUDIO_IO_HANDLE_NONE && setMsdPatch(deviceDesc) == NO_ERROR) {
+            ALOGV("%s() Using MSD devices %s instead of device %s",
+                    __func__, msdDevices.toString().c_str(), deviceDesc->toString().c_str());
+            deviceType = msdDevices.types();
         } else {
             *output = AUDIO_IO_HANDLE_NONE;
         }
     }
+    devices = mAvailableOutputDevices.getDevicesFromTypeMask(deviceType);
     if (*output == AUDIO_IO_HANDLE_NONE) {
-        *output = getOutputForDevice(device, session, *stream, config, flags);
+        *output = getOutputForDevices(devices, session, *stream, config, flags);
     }
     if (*output == AUDIO_IO_HANDLE_NONE) {
         return INVALID_OPERATION;
     }
 
-    outputDevices = mAvailableOutputDevices.getDevicesFromTypeMask(device);
-    *selectedDeviceId = getFirstDeviceId(outputDevices);
+    *selectedDeviceId = getFirstDeviceId(devices);
 
     ALOGV("%s returns output %d selectedDeviceId %d", __func__, *output, *selectedDeviceId);
 
@@ -931,7 +921,7 @@
     if (*portId != AUDIO_PORT_HANDLE_NONE) {
         return INVALID_OPERATION;
     }
-    const audio_port_handle_t requestedDeviceId = *selectedDeviceId;
+    const audio_port_handle_t requestedPortId = *selectedDeviceId;
     audio_attributes_t resultAttr;
     status_t status = getOutputForAttrInt(&resultAttr, output, session, attr, stream, uid,
             config, flags, selectedDeviceId);
@@ -946,20 +936,20 @@
 
     sp<TrackClientDescriptor> clientDesc =
         new TrackClientDescriptor(*portId, uid, session, resultAttr, clientConfig,
-                                  requestedDeviceId, *stream,
+                                  requestedPortId, *stream,
                                   getStrategyForAttr(&resultAttr),
                                   *flags);
     sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(*output);
     outputDesc->addClient(clientDesc);
 
     ALOGV("%s returns output %d selectedDeviceId %d for port ID %d",
-          __func__, *output, requestedDeviceId, *portId);
+          __func__, *output, requestedPortId, *portId);
 
     return NO_ERROR;
 }
 
-audio_io_handle_t AudioPolicyManager::getOutputForDevice(
-        audio_devices_t device,
+audio_io_handle_t AudioPolicyManager::getOutputForDevices(
+        const DeviceVector &devices,
         audio_session_t session,
         audio_stream_type_t stream,
         const audio_config_t *config,
@@ -1017,7 +1007,7 @@
 
     if (((*flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
             !(mEffects.isNonOffloadableEffectEnabled() || mMasterMono)) {
-        profile = getProfileForOutput(device,
+        profile = getProfileForOutput(devices,
                                    config->sample_rate,
                                    config->format,
                                    config->channel_mask,
@@ -1037,7 +1027,7 @@
                     (config->channel_mask == desc->mChannelMask) &&
                     (session == desc->mDirectClientSession)) {
                     desc->mDirectOpenCount++;
-                    ALOGI("getOutputForDevice() reusing direct output %d for session %d",
+                    ALOGI("%s reusing direct output %d for session %d", __func__, 
                         mOutputs.keyAt(i), session);
                     return mOutputs.keyAt(i);
                 }
@@ -1051,8 +1041,7 @@
         sp<SwAudioOutputDescriptor> outputDesc =
                 new SwAudioOutputDescriptor(profile, mpClientInterface);
 
-        DeviceVector outputDevices = mAvailableOutputDevices.getDevicesFromTypeMask(device);
-        String8 address = getFirstDeviceAddress(outputDevices);
+        String8 address = getFirstDeviceAddress(devices);
 
         // MSD patch may be using the only output stream that can service this request. Release
         // MSD patch to prioritize this request over any active output on MSD.
@@ -1062,7 +1051,7 @@
             for (size_t j = 0; j < patch->mPatch.num_sinks; ++j) {
                 const struct audio_port_config *sink = &patch->mPatch.sinks[j];
                 if (sink->type == AUDIO_PORT_TYPE_DEVICE &&
-                        (sink->ext.device.type & device) != AUDIO_DEVICE_NONE &&
+                        (sink->ext.device.type & devices.types()) != AUDIO_DEVICE_NONE &&
                         (address.isEmpty() || strncmp(sink->ext.device.address, address.string(),
                                 AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0)) {
                     releaseAudioPatch(patch->mHandle, mUidCached);
@@ -1071,15 +1060,15 @@
             }
         }
 
-        status = outputDesc->open(config, device, address, stream, *flags, &output);
+        status = outputDesc->open(config, devices, stream, *flags, &output);
 
         // only accept an output with the requested parameters
         if (status != NO_ERROR ||
             (config->sample_rate != 0 && config->sample_rate != outputDesc->mSamplingRate) ||
             (config->format != AUDIO_FORMAT_DEFAULT && config->format != outputDesc->mFormat) ||
             (config->channel_mask != 0 && config->channel_mask != outputDesc->mChannelMask)) {
-            ALOGV("getOutputForDevice() failed opening direct output: output %d sample rate %d %d,"
-                    "format %d %d, channel mask %04x %04x", output, config->sample_rate,
+            ALOGV("%s failed opening direct output: output %d sample rate %d %d," 
+                    "format %d %d, channel mask %04x %04x", __func__, output, config->sample_rate,
                     outputDesc->mSamplingRate, config->format, outputDesc->mFormat,
                     config->channel_mask, outputDesc->mChannelMask);
             if (output != AUDIO_IO_HANDLE_NONE) {
@@ -1097,7 +1086,7 @@
 
         addOutput(output, outputDesc);
         mPreviousOutputs = mOutputs;
-        ALOGV("getOutputForDevice() returns new direct output %d", output);
+        ALOGV("%s returns new direct output %d", __func__, output);
         mpClientInterface->onAudioPortListUpdate();
         return output;
     }
@@ -1118,14 +1107,14 @@
     if (audio_is_linear_pcm(config->format)) {
         // get which output is suitable for the specified stream. The actual
         // routing change will happen when startOutput() will be called
-        SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+        SortedVector<audio_io_handle_t> outputs = getOutputsForDevices(devices, mOutputs);
 
         // at this stage we should ignore the DIRECT flag as no direct output could be found earlier
         *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_DIRECT);
         output = selectOutput(outputs, *flags, config->format,
                 config->channel_mask, config->sample_rate);
     }
-    ALOGW_IF((output == 0), "getOutputForDevice() could not find output for stream %d, "
+    ALOGW_IF((output == 0), "getOutputForDevices() could not find output for stream %d, "
             "sampling rate %d, format %#x, channels %#x, flags %#x",
             stream, config->sample_rate, config->format, config->channel_mask, *flags);
 
@@ -1133,13 +1122,14 @@
 }
 
 sp<DeviceDescriptor> AudioPolicyManager::getMsdAudioInDevice() const {
-    sp<HwModule> msdModule = mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD);
-    if (msdModule != 0) {
-        DeviceVector msdInputDevices = mAvailableInputDevices.getDevicesFromHwModule(
-                msdModule->getHandle());
-        if (!msdInputDevices.isEmpty()) return msdInputDevices.itemAt(0);
-    }
-    return 0;
+    auto msdInDevices = mHwModules.getAvailableDevicesFromModuleName(AUDIO_HARDWARE_MODULE_ID_MSD,
+                                                                     mAvailableInputDevices);
+    return msdInDevices.isEmpty()? nullptr : msdInDevices.itemAt(0);
+}
+
+DeviceVector AudioPolicyManager::getMsdAudioOutDevices() const {
+    return mHwModules.getAvailableDevicesFromModuleName(AUDIO_HARDWARE_MODULE_ID_MSD,
+                                                        mAvailableOutputDevices);
 }
 
 const AudioPatchCollection AudioPolicyManager::getMsdPatches() const {
@@ -1160,7 +1150,7 @@
     return msdPatches;
 }
 
-status_t AudioPolicyManager::getBestMsdAudioProfileFor(audio_devices_t outputDevice,
+status_t AudioPolicyManager::getBestMsdAudioProfileFor(const sp<DeviceDescriptor> &outputDevice,
         bool hwAvSync, audio_port_config *sourceConfig, audio_port_config *sinkConfig) const
 {
     sp<HwModule> msdModule = mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD);
@@ -1170,7 +1160,7 @@
     }
     sp<HwModule> deviceModule = mHwModules.getModuleForDevice(outputDevice);
     if (deviceModule == nullptr) {
-        ALOGE("%s() unable to get module for %#x", __func__, outputDevice);
+        ALOGE("%s() unable to get module for %s", __func__, outputDevice->toString().c_str());
         return NO_INIT;
     }
     const InputProfileCollection &inputProfiles = msdModule->getInputProfiles();
@@ -1180,7 +1170,7 @@
     }
     const OutputProfileCollection &outputProfiles = deviceModule->getOutputProfiles();
     if (outputProfiles.isEmpty()) {
-        ALOGE("%s() no output profiles for device %#x", __func__, outputDevice);
+        ALOGE("%s() no output profiles for device %s", __func__, outputDevice->toString().c_str());
         return NO_INIT;
     }
     AudioProfileVector msdProfiles;
@@ -1201,8 +1191,8 @@
             compressedFormatsOrder, surroundChannelMasksOrder, true /*preferHigherSamplingRates*/,
             &bestSinkConfig);
     if (result != NO_ERROR) {
-        ALOGD("%s() no matching profiles found for device: %#x, hwAvSync: %d",
-                __func__, outputDevice, hwAvSync);
+        ALOGD("%s() no matching profiles found for device: %s, hwAvSync: %d",
+                __func__, outputDevice->toString().c_str(), hwAvSync);
         return result;
     }
     sinkConfig->sample_rate = bestSinkConfig.sample_rate;
@@ -1231,11 +1221,10 @@
     return NO_ERROR;
 }
 
-PatchBuilder AudioPolicyManager::buildMsdPatch(audio_devices_t outputDevice) const
+PatchBuilder AudioPolicyManager::buildMsdPatch(const sp<DeviceDescriptor> &outputDevice) const
 {
     PatchBuilder patchBuilder;
-    patchBuilder.addSource(getMsdAudioInDevice()).
-            addSink(findDevice(mAvailableOutputDevices, outputDevice));
+    patchBuilder.addSource(getMsdAudioInDevice()).addSink(outputDevice);
     audio_port_config sourceConfig = patchBuilder.patch()->sources[0];
     audio_port_config sinkConfig = patchBuilder.patch()->sinks[0];
     // TODO: Figure out whether MSD module has HW_AV_SYNC flag set in the AP config file.
@@ -1253,15 +1242,18 @@
     return patchBuilder;
 }
 
-status_t AudioPolicyManager::setMsdPatch(audio_devices_t outputDevice) {
-    ALOGV("%s() for outputDevice %#x", __func__, outputDevice);
-    if (outputDevice == AUDIO_DEVICE_NONE) {
+status_t AudioPolicyManager::setMsdPatch(const sp<DeviceDescriptor> &outputDevice) {
+    sp<DeviceDescriptor> device = outputDevice;
+    if (device == nullptr) {
         // Use media strategy for unspecified output device. This should only
         // occur on checkForDeviceAndOutputChanges(). Device connection events may
         // therefore invalidate explicit routing requests.
-        outputDevice = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+        DeviceVector devices = getDevicesForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+        LOG_ALWAYS_FATAL_IF(devices.isEmpty(), "no outpudevice to set Msd Patch");
+        device = devices.itemAt(0);
     }
-    PatchBuilder patchBuilder = buildMsdPatch(outputDevice);
+    ALOGV("%s() for device %s", __func__, device->toString().c_str());
+    PatchBuilder patchBuilder = buildMsdPatch(device);
     const struct audio_patch* patch = patchBuilder.patch();
     const AudioPatchCollection msdPatches = getMsdPatches();
     if (!msdPatches.isEmpty()) {
@@ -1277,8 +1269,9 @@
             patch, 0 /*delayMs*/, mUidCached, nullptr /*patchDescPtr*/);
     ALOGE_IF(status != NO_ERROR, "%s() error %d creating MSD audio patch", __func__, status);
     ALOGI_IF(status == NO_ERROR, "%s() Patch created from MSD_IN to "
-           "device:%#x (format:%#x channels:%#x samplerate:%d)", __func__, outputDevice,
-           patch->sources[0].format, patch->sources[0].channel_mask, patch->sources[0].sample_rate);
+           "device:%s (format:%#x channels:%#x samplerate:%d)", __func__,
+             device->toString().c_str(), patch->sources[0].format,
+             patch->sources[0].channel_mask, patch->sources[0].sample_rate);
     return status;
 }
 
@@ -1289,7 +1282,7 @@
                                                        uint32_t samplingRate)
 {
     // select one output among several that provide a path to a particular device or set of
-    // devices (the list was previously build by getOutputsForDevice()).
+    // devices (the list was previously build by getOutputsForDevices()).
     // The priority is as follows:
     // 1: the output supporting haptic playback when requesting haptic playback
     // 2: the output with the highest number of requested policy flags
@@ -1451,17 +1444,19 @@
     bool force = !outputDesc->isActive() &&
             (outputDesc->getPatchHandle() == AUDIO_PATCH_HANDLE_NONE);
 
-    audio_devices_t device = AUDIO_DEVICE_NONE;
+    DeviceVector devices;
     AudioMix *policyMix = NULL;
     const char *address = NULL;
     if (outputDesc->mPolicyMix != NULL) {
         policyMix = outputDesc->mPolicyMix;
+        audio_devices_t newDeviceType;
         address = policyMix->mDeviceAddress.string();
         if ((policyMix->mRouteFlags & MIX_ROUTE_FLAG_RENDER) == MIX_ROUTE_FLAG_RENDER) {
-            device = policyMix->mDeviceType;
+            newDeviceType = policyMix->mDeviceType;
         } else {
-            device = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+            newDeviceType = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
         }
+        devices.add(mAvailableOutputDevices.getDevice(newDeviceType, String8(address)));
     }
 
     // requiresMuteCheck is false when we can bypass mute strategy.
@@ -1476,8 +1471,8 @@
     outputDesc->setClientActive(client, true);
 
     if (client->hasPreferredDevice(true)) {
-        device = getNewOutputDevice(outputDesc, false /*fromCache*/);
-        if (device != outputDesc->device()) {
+        devices = getNewOutputDevices(outputDesc, false /*fromCache*/);
+        if (devices != outputDesc->devices()) {
             checkStrategyRoute(getStrategy(stream), outputDesc->mIoHandle);
         }
     }
@@ -1486,10 +1481,10 @@
         selectOutputForMusicEffects();
     }
 
-    if (outputDesc->streamActiveCount(stream) == 1 || device != AUDIO_DEVICE_NONE) {
+    if (outputDesc->streamActiveCount(stream) == 1 || !devices.isEmpty()) {
         // starting an output being rerouted?
-        if (device == AUDIO_DEVICE_NONE) {
-            device = getNewOutputDevice(outputDesc, false /*fromCache*/);
+        if (devices.isEmpty()) {
+            devices = getNewOutputDevices(outputDesc, false /*fromCache*/);
         }
 
         routing_strategy strategy = getStrategy(stream);
@@ -1498,13 +1493,13 @@
                             (beaconMuteLatency > 0);
         uint32_t waitMs = beaconMuteLatency;
         for (size_t i = 0; i < mOutputs.size(); i++) {
-            sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+            sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
             if (desc != outputDesc) {
                 // An output has a shared device if
                 // - managed by the same hw module
                 // - supports the currently selected device
                 const bool sharedDevice = outputDesc->sharesHwModuleWith(desc)
-                        && (desc->supportedDevices() & device) != AUDIO_DEVICE_NONE;
+                        && (!desc->filterSupportedDevices(devices).isEmpty());
 
                 // force a device change if any other output is:
                 // - managed by the same hw module
@@ -1514,7 +1509,7 @@
                 // In this case, the audio HAL must receive the new device selection so that it can
                 // change the device currently selected by the other output.
                 if (sharedDevice &&
-                        desc->device() != device &&
+                        desc->devices() != devices &&
                         desc->getPatchHandle() != AUDIO_PATCH_HANDLE_NONE) {
                     force = true;
                 }
@@ -1537,13 +1532,13 @@
         }
 
         const uint32_t muteWaitMs =
-                setOutputDevice(outputDesc, device, force, 0, NULL, address, requiresMuteCheck);
+                setOutputDevices(outputDesc, devices, force, 0, NULL, requiresMuteCheck);
 
         // apply volume rules for current stream and device if necessary
         checkAndSetVolume(stream,
-                          mVolumeCurves->getVolumeIndex(stream, outputDesc->device()),
+                          mVolumeCurves->getVolumeIndex(stream, outputDesc->devices().types()),
                           outputDesc,
-                          outputDesc->device());
+                          outputDesc->devices().types());
 
         // update the outputs if starting an output with a stream that can affect notification
         // routing
@@ -1574,7 +1569,7 @@
 
     // Automatically enable the remote submix input when output is started on a re routing mix
     // of type MIX_TYPE_RECORDERS
-    if (audio_is_remote_submix_device(device) && policyMix != NULL &&
+    if (audio_is_remote_submix_device(devices.types()) && policyMix != NULL &&
         policyMix->mMixType == MIX_TYPE_RECORDERS) {
         setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
                                     AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
@@ -1619,7 +1614,7 @@
         if (outputDesc->streamActiveCount(stream) == 1) {
             // Automatically disable the remote submix input when output is stopped on a
             // re routing mix of type MIX_TYPE_RECORDERS
-            if (audio_is_remote_submix_device(outputDesc->mDevice) &&
+            if (audio_is_remote_submix_device(outputDesc->devices().types()) &&
                 outputDesc->mPolicyMix != NULL &&
                 outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
                 setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
@@ -1640,33 +1635,31 @@
         // store time at which the stream was stopped - see isStreamActive()
         if (outputDesc->streamActiveCount(stream) == 0 || forceDeviceUpdate) {
             outputDesc->mStopTime[stream] = systemTime();
-            audio_devices_t newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/);
+            DeviceVector newDevices = getNewOutputDevices(outputDesc, false /*fromCache*/);
             // delay the device switch by twice the latency because stopOutput() is executed when
             // the track stop() command is received and at that time the audio track buffer can
             // still contain data that needs to be drained. The latency only covers the audio HAL
             // and kernel buffers. Also the latency does not always include additional delay in the
             // audio path (audio DSP, CODEC ...)
-            setOutputDevice(outputDesc, newDevice, false, outputDesc->latency()*2);
+            setOutputDevices(outputDesc, newDevices, false, outputDesc->latency()*2);
 
             // force restoring the device selection on other active outputs if it differs from the
             // one being selected for this output
             uint32_t delayMs = outputDesc->latency()*2;
             for (size_t i = 0; i < mOutputs.size(); i++) {
-                sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+                sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
                 if (desc != outputDesc &&
                         desc->isActive() &&
                         outputDesc->sharesHwModuleWith(desc) &&
-                        (newDevice != desc->device())) {
-                    audio_devices_t newDevice2 = getNewOutputDevice(desc, false /*fromCache*/);
-                    bool force = desc->device() != newDevice2;
+                        (newDevices != desc->devices())) {
+                    DeviceVector newDevices2 = getNewOutputDevices(desc, false /*fromCache*/);
+                    bool force = desc->devices() != newDevices2;
 
-                    setOutputDevice(desc,
-                                    newDevice2,
-                                    force,
-                                    delayMs);
+                    setOutputDevices(desc, newDevices2, force, delayMs);
+
                     // re-apply device specific volume if not done by setOutputDevice()
                     if (!force) {
-                        applyStreamVolumes(desc, newDevice2, delayMs);
+                        applyStreamVolumes(desc, newDevices2.types(), delayMs);
                     }
                 }
             }
@@ -1739,29 +1732,27 @@
           attr->source, config->sample_rate, config->format, config->channel_mask, session, flags);
 
     status_t status = NO_ERROR;
-    // handle legacy remote submix case where the address was not always specified
-    String8 address = String8("");
     audio_source_t halInputSource;
-    audio_source_t inputSource = attr->source;
+    audio_attributes_t attributes = *attr;
     AudioMix *policyMix = NULL;
-    DeviceVector inputDevices;
+    sp<DeviceDescriptor> device;
     sp<AudioInputDescriptor> inputDesc;
     sp<RecordClientDescriptor> clientDesc;
     audio_port_handle_t requestedDeviceId = *selectedDeviceId;
     bool isSoundTrigger;
-    audio_devices_t device;
 
     // The supplied portId must be AUDIO_PORT_HANDLE_NONE
     if (*portId != AUDIO_PORT_HANDLE_NONE) {
         return INVALID_OPERATION;
     }
 
-    if (inputSource == AUDIO_SOURCE_DEFAULT) {
-        inputSource = AUDIO_SOURCE_MIC;
+    if (attr->source == AUDIO_SOURCE_DEFAULT) {
+        attributes.source = AUDIO_SOURCE_MIC;
     }
 
     // Explicit routing?
-    sp<DeviceDescriptor> deviceDesc = mAvailableInputDevices.getDeviceFromId(*selectedDeviceId);
+    sp<DeviceDescriptor> explicitRoutingDevice = 
+            mAvailableInputDevices.getDeviceFromId(*selectedDeviceId);
 
     // special case for mmap capture: if an input IO handle is specified, we reuse this input if
     // possible
@@ -1802,7 +1793,7 @@
             }
         }
         *inputType = API_INPUT_LEGACY;
-        device = inputDesc->mDevice;
+        device = inputDesc->getDevice();
 
         ALOGI("%s reusing MMAP input %d for session %d", __FUNCTION__, *input, session);
         goto exit;
@@ -1811,44 +1802,38 @@
     *input = AUDIO_IO_HANDLE_NONE;
     *inputType = API_INPUT_INVALID;
 
-    halInputSource = inputSource;
+    halInputSource = attributes.source;
 
-    if (inputSource == AUDIO_SOURCE_REMOTE_SUBMIX &&
-            strncmp(attr->tags, "addr=", strlen("addr=")) == 0) {
-        status = mPolicyMixes.getInputMixForAttr(*attr, &policyMix);
+    if (attributes.source == AUDIO_SOURCE_REMOTE_SUBMIX &&
+            strncmp(attributes.tags, "addr=", strlen("addr=")) == 0) {
+        status = mPolicyMixes.getInputMixForAttr(attributes, &policyMix);
         if (status != NO_ERROR) {
             goto error;
         }
         *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;
-        device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
-        address = String8(attr->tags + strlen("addr="));
+        device = mAvailableInputDevices.getDevice(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+                                                  String8(attr->tags + strlen("addr=")));
     } else {
-        if (deviceDesc != 0) {
-            device = deviceDesc->type();
+        if (explicitRoutingDevice != nullptr) {
+            device = explicitRoutingDevice;
         } else {
-            device = getDeviceAndMixForInputSource(inputSource, &policyMix);
+            device = getDeviceAndMixForAttributes(attributes, &policyMix);
         }
-        if (device == AUDIO_DEVICE_NONE) {
-            ALOGW("getInputForAttr() could not find device for source %d", inputSource);
+        if (device == nullptr) {
+            ALOGW("getInputForAttr() could not find device for source %d", attributes.source);
             status = BAD_VALUE;
             goto error;
         }
-        if (policyMix != NULL) {
-            address = policyMix->mDeviceAddress;
-            if (policyMix->mMixType == MIX_TYPE_RECORDERS) {
-                // there is an external policy, but this input is attached to a mix of recorders,
-                // meaning it receives audio injected into the framework, so the recorder doesn't
-                // know about it and is therefore considered "legacy"
-                *inputType = API_INPUT_LEGACY;
-            } else {
-                // recording a mix of players defined by an external policy, we're rerouting for
-                // an external policy
-                *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;
-            }
-        } else if (audio_is_remote_submix_device(device)) {
-            address = String8("0");
+        if (policyMix != nullptr) {
+            ALOG_ASSERT(policyMix->mMixType == MIX_TYPE_RECORDERS, "Invalid Mix Type");
+            // there is an external policy, but this input is attached to a mix of recorders,
+            // meaning it receives audio injected into the framework, so the recorder doesn't
+            // know about it and is therefore considered "legacy"
+            *inputType = API_INPUT_LEGACY;
+        } else if (audio_is_remote_submix_device(device->type())) {
+            device = mAvailableInputDevices.getDevice(AUDIO_DEVICE_IN_REMOTE_SUBMIX, String8("0"));
             *inputType = API_INPUT_MIX_CAPTURE;
-        } else if (device == AUDIO_DEVICE_IN_TELEPHONY_RX) {
+        } else if (device->type() == AUDIO_DEVICE_IN_TELEPHONY_RX) {
             *inputType = API_INPUT_TELEPHONY_RX;
         } else {
             *inputType = API_INPUT_LEGACY;
@@ -1856,7 +1841,7 @@
 
     }
 
-    *input = getInputForDevice(device, address, session, inputSource,
+    *input = getInputForDevice(device, session, attributes.source,
                                config, flags,
                                policyMix);
     if (*input == AUDIO_IO_HANDLE_NONE) {
@@ -1866,16 +1851,16 @@
 
 exit:
 
-    inputDevices = mAvailableInputDevices.getDevicesFromTypeMask(device);
-    *selectedDeviceId = getFirstDeviceId(inputDevices);
+    *selectedDeviceId = mAvailableInputDevices.contains(device) ? 
+            device->getId() : AUDIO_PORT_HANDLE_NONE;
 
-    isSoundTrigger = inputSource == AUDIO_SOURCE_HOTWORD &&
+    isSoundTrigger = attributes.source == AUDIO_SOURCE_HOTWORD &&
         mSoundTriggerSessions.indexOfKey(session) > 0;
     *portId = AudioPort::getNextUniqueId();
 
-    clientDesc = new RecordClientDescriptor(*portId, uid, session,
-                                  *attr, *config, requestedDeviceId,
-                                  inputSource,flags, isSoundTrigger);
+    clientDesc = new RecordClientDescriptor(*portId, uid, session, *attr, *config,
+                                            requestedDeviceId, attributes.source, flags,
+                                            isSoundTrigger);
     inputDesc = mInputs.valueFor(*input);
     inputDesc->addClient(clientDesc);
 
@@ -1889,8 +1874,7 @@
 }
 
 
-audio_io_handle_t AudioPolicyManager::getInputForDevice(audio_devices_t device,
-                                                        String8 address,
+audio_io_handle_t AudioPolicyManager::getInputForDevice(const sp<DeviceDescriptor> &device,
                                                         audio_session_t session,
                                                         audio_source_t inputSource,
                                                         const audio_config_base_t *config,
@@ -1926,8 +1910,7 @@
     audio_input_flags_t profileFlags = flags;
     for (;;) {
         profileFormat = config->format; // reset each time through loop, in case it is updated
-        profile = getInputProfile(device, address,
-                                  profileSamplingRate, profileFormat, profileChannelMask,
+        profile = getInputProfile(device, profileSamplingRate, profileFormat, profileChannelMask,
                                   profileFlags);
         if (profile != 0) {
             break; // success
@@ -1936,9 +1919,9 @@
         } else if (profileFlags != AUDIO_INPUT_FLAG_NONE) {
             profileFlags = AUDIO_INPUT_FLAG_NONE; // retry
         } else { // fail
-            ALOGW("getInputForDevice() could not find profile for device 0x%X, "
-                  "sampling rate %u, format %#x, channel mask 0x%X, flags %#x",
-                    device, config->sample_rate, config->format, config->channel_mask, flags);
+            ALOGW("%s could not find profile for device %s, sampling rate %u, format %#x, "
+                  "channel mask 0x%X, flags %#x", __func__, device->toString().c_str(), 
+                  config->sample_rate, config->format, config->channel_mask, flags);
             return input;
         }
     }
@@ -1995,14 +1978,7 @@
     lConfig.channel_mask = profileChannelMask;
     lConfig.format = profileFormat;
 
-    if (address == "") {
-        DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromTypeMask(device);
-        // the inputs vector must be of size >= 1, but we don't want to crash here
-        address = getFirstDeviceAddress(inputDevices);
-    }
-
-    status_t status = inputDesc->open(&lConfig, device, address,
-            halInputSource, profileFlags, &input);
+    status_t status = inputDesc->open(&lConfig, device, halInputSource, profileFlags, &input);
 
     // only accept input with the exact requested set of parameters
     if (status != NO_ERROR || input == AUDIO_IO_HANDLE_NONE ||
@@ -2059,7 +2035,7 @@
 
     // indicate active capture to sound trigger service if starting capture from a mic on
     // primary HW module
-    audio_devices_t device = getNewInputDevice(inputDesc);
+    sp<DeviceDescriptor> device = getNewInputDevice(inputDesc);
     setInputDevice(input, device, true /* force */);
 
     if (inputDesc->activeCount()  == 1) {
@@ -2070,8 +2046,8 @@
                     MIX_STATE_MIXING);
         }
 
-        audio_devices_t primaryInputDevices = availablePrimaryInputDevices();
-        if (((device & primaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) &&
+        DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
+        if (primaryInputDevices.contains(device) &&
                 mInputs.activeInputsCountOnDevices(primaryInputDevices) == 1) {
             SoundTrigger::setCaptureState(true);
         }
@@ -2079,7 +2055,7 @@
         // automatically enable the remote submix output when input is started if not
         // used by a policy mix of type MIX_TYPE_RECORDERS
         // For remote submix (a virtual device), we open only one input per capture request.
-        if (audio_is_remote_submix_device(inputDesc->mDevice)) {
+        if (audio_is_remote_submix_device(inputDesc->getDeviceType())) {
             String8 address = String8("");
             if (inputDesc->mPolicyMix == NULL) {
                 address = String8("0");
@@ -2130,7 +2106,7 @@
 
         // automatically disable the remote submix output when input is stopped if not
         // used by a policy mix of type MIX_TYPE_RECORDERS
-        if (audio_is_remote_submix_device(inputDesc->mDevice)) {
+        if (audio_is_remote_submix_device(inputDesc->getDeviceType())) {
             String8 address = String8("");
             if (inputDesc->mPolicyMix == NULL) {
                 address = String8("0");
@@ -2143,14 +2119,12 @@
                                          address, "remote-submix");
             }
         }
-
-        audio_devices_t device = inputDesc->mDevice;
         resetInputDevice(input);
 
         // indicate inactive capture to sound trigger service if stopping capture from a mic on
         // primary HW module
-        audio_devices_t primaryInputDevices = availablePrimaryInputDevices();
-        if (((device & primaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) &&
+        DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
+        if (primaryInputDevices.contains(inputDesc->getDevice()) &&
                 mInputs.activeInputsCountOnDevices(primaryInputDevices) == 0) {
             SoundTrigger::setCaptureState(false);
         }
@@ -2280,7 +2254,7 @@
     status_t status = NO_ERROR;
     for (size_t i = 0; i < mOutputs.size(); i++) {
         sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
-        audio_devices_t curDevice = desc->device();
+        audio_devices_t curDevice = desc->devices().types();
         for (int curStream = 0; curStream < AUDIO_STREAM_FOR_POLICY_CNT; curStream++) {
             if (!(streamsMatchForvolume(stream, (audio_stream_type_t)curStream))) {
                 continue;
@@ -2356,8 +2330,8 @@
     // 4: the first output in the list
 
     routing_strategy strategy = getStrategy(AUDIO_STREAM_MUSIC);
-    audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
-    SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+    DeviceVector devices = getDevicesForStrategy(strategy, false /*fromCache*/);
+    SortedVector<audio_io_handle_t> outputs = getOutputsForDevices(devices, mOutputs);
 
     if (outputs.size() == 0) {
         return AUDIO_IO_HANDLE_NONE;
@@ -2693,8 +2667,7 @@
                             devices[i].mType, devices[i].mAddress, String8());
             SortedVector<audio_io_handle_t> outputs;
             if (checkOutputsForDevice(devDesc, AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
-                    outputs,
-                    devDesc->address()) != NO_ERROR) {
+                    outputs) != NO_ERROR) {
                 ALOGE("setUidDeviceAffinities() error in checkOutputsForDevice for device=%08x"
                         " addr=%s", devices[i].mType, devices[i].mAddress.string());
                 return INVALID_OPERATION;
@@ -2715,8 +2688,7 @@
                     devices[i].mType, devices[i].mAddress, String8());
             SortedVector<audio_io_handle_t> outputs;
             if (checkOutputsForDevice(devDesc, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
-                    outputs,
-                    devDesc->address()) != NO_ERROR) {
+                    outputs) != NO_ERROR) {
                 ALOGE("%s() error in checkOutputsForDevice for device=%08x addr=%s",
                         __FUNCTION__, devices[i].mType, devices[i].mAddress.string());
                 return INVALID_OPERATION;
@@ -2836,7 +2808,7 @@
 
     // See if there is a profile to support this.
     // AUDIO_DEVICE_NONE
-    sp<IOProfile> profile = getProfileForOutput(AUDIO_DEVICE_NONE /*ignore device */,
+    sp<IOProfile> profile = getProfileForOutput(DeviceVector() /*ignore device */,
                                             offloadInfo.sample_rate,
                                             offloadInfo.format,
                                             offloadInfo.channel_mask,
@@ -2850,7 +2822,7 @@
                                                  const audio_attributes_t& attributes) {
     audio_output_flags_t output_flags = AUDIO_OUTPUT_FLAG_NONE;
     audio_attributes_flags_to_audio_output_flags(attributes.flags, output_flags);
-    sp<IOProfile> profile = getProfileForOutput(AUDIO_DEVICE_NONE /*ignore device */,
+    sp<IOProfile> profile = getProfileForOutput(DeviceVector() /*ignore device */,
                                             config.sample_rate,
                                             config.format,
                                             config.channel_mask,
@@ -3044,8 +3016,7 @@
                 return BAD_VALUE;
             }
 
-            if (!outputDesc->mProfile->isCompatibleProfile(devDesc->type(),
-                                                           devDesc->address(),
+            if (!outputDesc->mProfile->isCompatibleProfile(DeviceVector(devDesc),
                                                            patch->sources[0].sample_rate,
                                                            NULL,  // updatedSamplingRate
                                                            patch->sources[0].format,
@@ -3066,7 +3037,7 @@
         // TODO: reconfigure output format and channels here
         ALOGV("createAudioPatch() setting device %08x on output %d",
               devices.types(), outputDesc->mIoHandle);
-        setOutputDevice(outputDesc, devices.types(), true, 0, handle);
+        setOutputDevices(outputDesc, devices, true, 0, handle);
         index = mAudioPatches.indexOfKey(*handle);
         if (index >= 0) {
             if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
@@ -3095,14 +3066,13 @@
                     return BAD_VALUE;
                 }
             }
-            sp<DeviceDescriptor> devDesc =
+            sp<DeviceDescriptor> device =
                     mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
-            if (devDesc == 0) {
+            if (device == 0) {
                 return BAD_VALUE;
             }
 
-            if (!inputDesc->mProfile->isCompatibleProfile(devDesc->type(),
-                                                          devDesc->address(),
+            if (!inputDesc->mProfile->isCompatibleProfile(DeviceVector(device),
                                                           patch->sinks[0].sample_rate,
                                                           NULL, /*updatedSampleRate*/
                                                           patch->sinks[0].format,
@@ -3116,9 +3086,9 @@
                 return INVALID_OPERATION;
             }
             // TODO: reconfigure output format and channels here
-            ALOGV("createAudioPatch() setting device %08x on output %d",
-                                                  devDesc->type(), inputDesc->mIoHandle);
-            setInputDevice(inputDesc->mIoHandle, devDesc->type(), true, handle);
+            ALOGV("%s() setting device %s on output %d", __func__,
+                  device->toString().c_str(), inputDesc->mIoHandle);
+            setInputDevice(inputDesc->mIoHandle, device, true, handle);
             index = mAudioPatches.indexOfKey(*handle);
             if (index >= 0) {
                 if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
@@ -3138,16 +3108,16 @@
                     return BAD_VALUE;
                 }
             }
-            sp<DeviceDescriptor> srcDeviceDesc =
+            sp<DeviceDescriptor> srcDevice =
                     mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
-            if (srcDeviceDesc == 0) {
+            if (srcDevice == 0) {
                 return BAD_VALUE;
             }
 
             //update source and sink with our own data as the data passed in the patch may
             // be incomplete.
             struct audio_patch newPatch = *patch;
-            srcDeviceDesc->toAudioPortConfig(&newPatch.sources[0], &patch->sources[0]);
+            srcDevice->toAudioPortConfig(&newPatch.sources[0], &patch->sources[0]);
 
             for (size_t i = 0; i < patch->num_sinks; i++) {
                 if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
@@ -3155,26 +3125,26 @@
                     return INVALID_OPERATION;
                 }
 
-                sp<DeviceDescriptor> sinkDeviceDesc =
+                sp<DeviceDescriptor> sinkDevice =
                         mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
-                if (sinkDeviceDesc == 0) {
+                if (sinkDevice == 0) {
                     return BAD_VALUE;
                 }
-                sinkDeviceDesc->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]);
+                sinkDevice->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]);
 
                 // create a software bridge in PatchPanel if:
                 // - source and sink devices are on different HW modules OR
                 // - audio HAL version is < 3.0
                 // - audio HAL version is >= 3.0 but no route has been declared between devices
-                if (!srcDeviceDesc->hasSameHwModuleAs(sinkDeviceDesc) ||
-                        (srcDeviceDesc->getModuleVersionMajor() < 3) ||
-                        !srcDeviceDesc->getModule()->supportsPatch(srcDeviceDesc, sinkDeviceDesc)) {
+                if (!srcDevice->hasSameHwModuleAs(sinkDevice) ||
+                        (srcDevice->getModuleVersionMajor() < 3) ||
+                        !srcDevice->getModule()->supportsPatch(srcDevice, sinkDevice)) {
                     // support only one sink device for now to simplify output selection logic
                     if (patch->num_sinks > 1) {
                         return INVALID_OPERATION;
                     }
                     SortedVector<audio_io_handle_t> outputs =
-                                            getOutputsForDevice(sinkDeviceDesc->type(), mOutputs);
+                            getOutputsForDevices(DeviceVector(sinkDevice), mOutputs);
                     // if the sink device is reachable via an opened output stream, request to go via
                     // this output stream by adding a second source to the patch description
                     audio_io_handle_t output = selectOutput(outputs);
@@ -3232,11 +3202,11 @@
             return BAD_VALUE;
         }
 
-        setOutputDevice(outputDesc,
-                        getNewOutputDevice(outputDesc, true /*fromCache*/),
-                       true,
-                       0,
-                       NULL);
+        setOutputDevices(outputDesc,
+                         getNewOutputDevices(outputDesc, true /*fromCache*/),
+                         true,
+                         0,
+                         NULL);
     } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
         if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
             sp<AudioInputDescriptor> inputDesc = mInputs.getInputFromId(patch->sinks[0].id);
@@ -3359,8 +3329,8 @@
 void AudioPolicyManager::checkStrategyRoute(routing_strategy strategy,
                                             audio_io_handle_t ouptutToSkip)
 {
-    audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
-    SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+    DeviceVector devices = getDevicesForStrategy(strategy, false /*fromCache*/);
+    SortedVector<audio_io_handle_t> outputs = getOutputsForDevices(devices, mOutputs);
     for (size_t j = 0; j < mOutputs.size(); j++) {
         if (mOutputs.keyAt(j) == ouptutToSkip) {
             continue;
@@ -3379,8 +3349,8 @@
                 }
             }
         } else {
-            audio_devices_t newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/);
-            setOutputDevice(outputDesc, newDevice, false);
+            setOutputDevices(
+                        outputDesc, getNewOutputDevices(outputDesc, false /*fromCache*/), false);
         }
     }
 }
@@ -3443,7 +3413,8 @@
 {
     *session = (audio_session_t)mpClientInterface->newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
     *ioHandle = (audio_io_handle_t)mpClientInterface->newAudioUniqueId(AUDIO_UNIQUE_ID_USE_INPUT);
-    *device = getDeviceAndMixForInputSource(AUDIO_SOURCE_HOTWORD);
+    audio_attributes_t attr = { .source = AUDIO_SOURCE_HOTWORD };
+    *device = getDeviceAndMixForAttributes(attr)->type();
 
     return mSoundTriggerSessions.acquireSession(*session, *ioHandle);
 }
@@ -3469,10 +3440,10 @@
         return INVALID_OPERATION;
     }
 
-    sp<DeviceDescriptor> srcDeviceDesc =
+    sp<DeviceDescriptor> srcDevice =
             mAvailableInputDevices.getDevice(source->ext.device.type,
-                                              String8(source->ext.device.address));
-    if (srcDeviceDesc == 0) {
+                                             String8(source->ext.device.address));
+    if (srcDevice == 0) {
         ALOGW("%s source->ext.device.type %08x not found", __FUNCTION__, source->ext.device.type);
         return BAD_VALUE;
     }
@@ -3483,7 +3454,7 @@
     sp<AudioPatch> patchDesc = new AudioPatch(&dummyPatch, uid);
 
     sp<SourceClientDescriptor> sourceDesc =
-        new SourceClientDescriptor(*portId, uid, *attributes, patchDesc, srcDeviceDesc,
+        new SourceClientDescriptor(*portId, uid, *attributes, patchDesc, srcDevice,
                                    streamTypefromAttributesInt(attributes),
                                    getStrategyForAttr(attributes));
 
@@ -3504,18 +3475,20 @@
     audio_attributes_t attributes = sourceDesc->attributes();
     routing_strategy strategy = getStrategyForAttr(&attributes);
     audio_stream_type_t stream = sourceDesc->stream();
-    sp<DeviceDescriptor> srcDeviceDesc = sourceDesc->srcDevice();
+    sp<DeviceDescriptor> srcDevice = sourceDesc->srcDevice();
 
-    audio_devices_t sinkDevice = getDeviceForStrategy(strategy, true);
-    sp<DeviceDescriptor> sinkDeviceDesc =
-            mAvailableOutputDevices.getDevice(sinkDevice, String8(""));
+    DeviceVector sinkDevices = getDevicesForStrategy(strategy, true);
+    ALOG_ASSERT(!sinkDevices.isEmpty(), "connectAudioSource(): no device found for strategy");
+    sp<DeviceDescriptor> sinkDevice = sinkDevices.itemAt(0);
+    ALOG_ASSERT(mAvailableOutputDevices.contains(sinkDevice), "%s: Device %s not available",
+                __FUNCTION__, sinkDevice->toString().c_str());
 
     audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
 
-    if (srcDeviceDesc->hasSameHwModuleAs(sinkDeviceDesc) &&
-            srcDeviceDesc->getModuleVersionMajor() >= 3 &&
-            sinkDeviceDesc->getModule()->supportsPatch(srcDeviceDesc, sinkDeviceDesc) &&
-            srcDeviceDesc->getAudioPort()->mGains.size() > 0) {
+    if (srcDevice->hasSameHwModuleAs(sinkDevice) &&
+            srcDevice->getModuleVersionMajor() >= 3 &&
+            sinkDevice->getModule()->supportsPatch(srcDevice, sinkDevice) &&
+            srcDevice->getAudioPort()->mGains.size() > 0) {
         ALOGV("%s Device to Device route supported by >=3.0 HAL", __FUNCTION__);
         // TODO: may explicitly specify whether we should use HW or SW patch
         //   create patch between src device and output device
@@ -3532,12 +3505,12 @@
         getOutputForAttrInt(&resultAttr, &output, AUDIO_SESSION_NONE,
                 &attributes, &stream, sourceDesc->uid(), &config, &flags, &selectedDeviceId);
         if (output == AUDIO_IO_HANDLE_NONE) {
-            ALOGV("%s no output for device %08x", __FUNCTION__, sinkDevice);
+            ALOGV("%s no output for device %08x", __FUNCTION__, sinkDevices.types());
             return INVALID_OPERATION;
         }
         sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
         if (outputDesc->isDuplicated()) {
-            ALOGV("%s output for device %08x is duplicated", __FUNCTION__, sinkDevice);
+            ALOGV("%s output for device %08x is duplicated", __FUNCTION__, sinkDevices.types());
             return INVALID_OPERATION;
         }
         status_t status = outputDesc->start();
@@ -3551,7 +3524,7 @@
         // - the sink is defined by whatever output device is currently selected for the output
         // though which this patch is routed.
         PatchBuilder patchBuilder;
-        patchBuilder.addSource(srcDeviceDesc).addSource(outputDesc, { .stream = stream });
+        patchBuilder.addSource(srcDevice).addSource(outputDesc, { .stream = stream });
         status = mpClientInterface->createAudioPatch(patchBuilder.patch(),
                                                               &afPatchHandle,
                                                               0);
@@ -3978,8 +3951,6 @@
 
     // mAvailableOutputDevices and mAvailableInputDevices now contain all attached devices
     // open all output streams needed to access attached devices
-    audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types();
-    audio_devices_t inputDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
     for (const auto& hwModule : mHwModulesAll) {
         hwModule->setHandle(mpClientInterface->loadHwModule(hwModule->getName()));
         if (hwModule->getHandle() == AUDIO_MODULE_HANDLE_NONE) {
@@ -4008,51 +3979,49 @@
             if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
                 continue;
             }
-            audio_devices_t profileType = outProfile->getSupportedDevicesType();
-            if ((profileType & mDefaultOutputDevice->type()) != AUDIO_DEVICE_NONE) {
-                profileType = mDefaultOutputDevice->type();
+            const DeviceVector &supportedDevices = outProfile->getSupportedDevices();
+            DeviceVector availProfileDevices = supportedDevices.filter(mAvailableOutputDevices);
+            sp<DeviceDescriptor> supportedDevice = 0;
+            if (supportedDevices.contains(mDefaultOutputDevice)) {
+                supportedDevice = mDefaultOutputDevice;
             } else {
-                // chose first device present in profile's SupportedDevices also part of
-                // outputDeviceTypes
-                profileType = outProfile->getSupportedDeviceForType(outputDeviceTypes);
+                // choose first device present in profile's SupportedDevices also part of
+                // mAvailableOutputDevices.
+                if (availProfileDevices.isEmpty()) {
+                    continue;
+                }
+                supportedDevice = availProfileDevices.itemAt(0);
             }
-            if ((profileType & outputDeviceTypes) == 0) {
+            if (!mAvailableOutputDevices.contains(supportedDevice)) {
                 continue;
             }
             sp<SwAudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(outProfile,
                                                                                  mpClientInterface);
-            const DeviceVector &supportedDevices = outProfile->getSupportedDevices();
-            const DeviceVector &devicesForType = supportedDevices.getDevicesFromTypeMask(
-                    profileType);
-            String8 address = getFirstDeviceAddress(devicesForType);
             audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
-            status_t status = outputDesc->open(nullptr, profileType, address,
-                                           AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
-
+            status_t status = outputDesc->open(nullptr, DeviceVector(supportedDevice),
+                                               AUDIO_STREAM_DEFAULT,
+                                               AUDIO_OUTPUT_FLAG_NONE, &output);
             if (status != NO_ERROR) {
-                ALOGW("Cannot open output stream for device %08x on hw module %s",
-                      outputDesc->mDevice,
-                      hwModule->getName());
-            } else {
-                for (const auto& dev : supportedDevices) {
-                    ssize_t index = mAvailableOutputDevices.indexOf(dev);
-                    // give a valid ID to an attached device once confirmed it is reachable
-                    if (index >= 0 && !mAvailableOutputDevices[index]->isAttached()) {
-                        mAvailableOutputDevices[index]->attach(hwModule);
-                    }
-                }
-                if (mPrimaryOutput == 0 &&
-                        outProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) {
-                    mPrimaryOutput = outputDesc;
-                }
-                addOutput(output, outputDesc);
-                setOutputDevice(outputDesc,
-                                profileType,
-                                true,
-                                0,
-                                NULL,
-                                address);
+                ALOGW("Cannot open output stream for devices %s on hw module %s",
+                      supportedDevice->toString().c_str(), hwModule->getName());
+                continue;
             }
+            for (const auto &device : availProfileDevices) {
+                // give a valid ID to an attached device once confirmed it is reachable
+                if (!device->isAttached()) {
+                    device->attach(hwModule);
+                }
+            }
+            if (mPrimaryOutput == 0 &&
+                    outProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) {
+                mPrimaryOutput = outputDesc;
+            }
+            addOutput(output, outputDesc);
+            setOutputDevices(outputDesc,
+                             DeviceVector(supportedDevice),
+                             true,
+                             0,
+                             NULL);
         }
         // open input streams needed to access attached devices to validate
         // mAvailableInputDevices list
@@ -4067,75 +4036,59 @@
                 continue;
             }
             // chose first device present in profile's SupportedDevices also part of
-            // inputDeviceTypes
-            audio_devices_t profileType = inProfile->getSupportedDeviceForType(inputDeviceTypes);
-
-            if ((profileType & inputDeviceTypes) == 0) {
+            // available input devices
+            const DeviceVector &supportedDevices = inProfile->getSupportedDevices();
+            DeviceVector availProfileDevices = supportedDevices.filter(mAvailableInputDevices);
+            if (availProfileDevices.isEmpty()) {
+                ALOGE("%s: Input device list is empty!", __FUNCTION__);
                 continue;
             }
             sp<AudioInputDescriptor> inputDesc =
                     new AudioInputDescriptor(inProfile, mpClientInterface);
 
-            DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromTypeMask(profileType);
-            //   the inputs vector must be of size >= 1, but we don't want to crash here
-            String8 address = getFirstDeviceAddress(inputDevices);
-            ALOGV("  for input device 0x%x using address %s", profileType, address.string());
-            ALOGE_IF(inputDevices.size() == 0, "Input device list is empty!");
-
             audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
             status_t status = inputDesc->open(nullptr,
-                                              profileType,
-                                              address,
+                                              availProfileDevices.itemAt(0),
                                               AUDIO_SOURCE_MIC,
                                               AUDIO_INPUT_FLAG_NONE,
                                               &input);
-
-            if (status == NO_ERROR) {
-                for (const auto& dev : inProfile->getSupportedDevices()) {
-                    ssize_t index = mAvailableInputDevices.indexOf(dev);
-                    // give a valid ID to an attached device once confirmed it is reachable
-                    if (index >= 0) {
-                        sp<DeviceDescriptor> devDesc = mAvailableInputDevices[index];
-                        if (!devDesc->isAttached()) {
-                            devDesc->attach(hwModule);
-                            devDesc->importAudioPort(inProfile, true);
-                        }
-                    }
-                }
-                inputDesc->close();
-            } else {
-                ALOGW("Cannot open input stream for device %08x on hw module %s",
-                      profileType,
+            if (status != NO_ERROR) {
+                ALOGW("Cannot open input stream for device %s on hw module %s",
+                      availProfileDevices.toString().c_str(),
                       hwModule->getName());
+                continue;
             }
+            for (const auto &device : availProfileDevices) {
+                // give a valid ID to an attached device once confirmed it is reachable
+                if (!device->isAttached()) {
+                    device->attach(hwModule);
+                    device->importAudioPort(inProfile, true);
+                }
+            }
+            inputDesc->close();
         }
     }
     // make sure all attached devices have been allocated a unique ID
-    for (size_t i = 0; i  < mAvailableOutputDevices.size();) {
-        if (!mAvailableOutputDevices[i]->isAttached()) {
-            ALOGW("Output device %08x unreachable", mAvailableOutputDevices[i]->type());
-            mAvailableOutputDevices.remove(mAvailableOutputDevices[i]);
-            continue;
+    auto checkAndSetAvailable = [this](auto& devices) {
+        for (size_t i = 0; i < devices.size();) {
+            const auto &device = devices[i];
+            if (!device->isAttached()) {
+                ALOGW("device %s is unreachable", device->toString().c_str());
+                devices.remove(device);
+                continue;
+            }
+            // Device is now validated and can be appended to the available devices of the engine
+            mEngine->setDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
+            i++;
         }
-        // The device is now validated and can be appended to the available devices of the engine
-        mEngine->setDeviceConnectionState(mAvailableOutputDevices[i],
-                                          AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
-        i++;
-    }
-    for (size_t i = 0; i  < mAvailableInputDevices.size();) {
-        if (!mAvailableInputDevices[i]->isAttached()) {
-            ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->type());
-            mAvailableInputDevices.remove(mAvailableInputDevices[i]);
-            continue;
-        }
-        // The device is now validated and can be appended to the available devices of the engine
-        mEngine->setDeviceConnectionState(mAvailableInputDevices[i],
-                                          AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
-        i++;
-    }
+    };
+    checkAndSetAvailable(mAvailableOutputDevices);
+    checkAndSetAvailable(mAvailableInputDevices);
+
     // make sure default device is reachable
-    if (mDefaultOutputDevice == 0 || mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) {
-        ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->type());
+    if (mDefaultOutputDevice == 0 || !mAvailableOutputDevices.contains(mDefaultOutputDevice)) {
+        ALOGE_IF(mDefaultOutputDevice != 0, "Default device %s is unreachable",
+                 mDefaultOutputDevice->toString().c_str());
         status = NO_INIT;
     }
     // If microphones address is empty, set it according to device type
@@ -4208,44 +4161,27 @@
     nextAudioPortGeneration();
 }
 
-void AudioPolicyManager::findIoHandlesByAddress(const sp<SwAudioOutputDescriptor>& desc /*in*/,
-        const audio_devices_t device /*in*/,
-        const String8& address /*in*/,
-        SortedVector<audio_io_handle_t>& outputs /*out*/) {
-    sp<DeviceDescriptor> devDesc =
-        desc->mProfile->getSupportedDeviceByAddress(device, address);
-    if (devDesc != 0) {
-        ALOGV("findIoHandlesByAddress(): adding opened output %d on same address %s",
-              desc->mIoHandle, address.string());
-        outputs.add(desc->mIoHandle);
-    }
-}
-
-status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor>& devDesc,
+status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor>& device,
                                                    audio_policy_dev_state_t state,
-                                                   SortedVector<audio_io_handle_t>& outputs,
-                                                   const String8& address)
+                                                   SortedVector<audio_io_handle_t>& outputs)
 {
-    audio_devices_t device = devDesc->type();
+    audio_devices_t deviceType = device->type();
+    const String8 &address = device->address();
     sp<SwAudioOutputDescriptor> desc;
 
-    if (audio_device_is_digital(device)) {
+    if (audio_device_is_digital(deviceType)) {
         // erase all current sample rates, formats and channel masks
-        devDesc->clearAudioProfiles();
+        device->clearAudioProfiles();
     }
 
     if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
         // first list already open outputs that can be routed to this device
         for (size_t i = 0; i < mOutputs.size(); i++) {
             desc = mOutputs.valueAt(i);
-            if (!desc->isDuplicated() && (desc->supportedDevices() & device)) {
-                if (!device_distinguishes_on_address(device)) {
-                    ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i));
-                    outputs.add(mOutputs.keyAt(i));
-                } else {
-                    ALOGV("  checking address match due to device 0x%x", device);
-                    findIoHandlesByAddress(desc, device, address, outputs);
-                }
+            if (!desc->isDuplicated() && desc->supportsDevice(device)) {
+                ALOGV("checkOutputsForDevice(): adding opened output %d on device %s",
+                      mOutputs.keyAt(i), device->toString().c_str());
+                outputs.add(mOutputs.keyAt(i));
             }
         }
         // then look for output profiles that can be routed to this device
@@ -4253,13 +4189,10 @@
         for (const auto& hwModule : mHwModules) {
             for (size_t j = 0; j < hwModule->getOutputProfiles().size(); j++) {
                 sp<IOProfile> profile = hwModule->getOutputProfiles()[j];
-                if (profile->supportDevice(device)) {
-                    if (!device_distinguishes_on_address(device) ||
-                            profile->supportDeviceAddress(address)) {
-                        profiles.add(profile);
-                        ALOGV("checkOutputsForDevice(): adding profile %zu from module %s",
-                                j, hwModule->getName());
-                    }
+                if (profile->supportsDevice(device)) {
+                    profiles.add(profile);
+                    ALOGV("checkOutputsForDevice(): adding profile %zu from module %s",
+                          j, hwModule->getName());
                 }
             }
         }
@@ -4267,7 +4200,7 @@
         ALOGV("  found %zu profiles, %zu outputs", profiles.size(), outputs.size());
 
         if (profiles.isEmpty() && outputs.isEmpty()) {
-            ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
+            ALOGW("checkOutputsForDevice(): No output available for device %04x", deviceType);
             return BAD_VALUE;
         }
 
@@ -4283,8 +4216,8 @@
                 if (!desc->isDuplicated() && desc->mProfile == profile) {
                     // matching profile: save the sample rates, format and channel masks supported
                     // by the profile in our device descriptor
-                    if (audio_device_is_digital(device)) {
-                        devDesc->importAudioPort(profile);
+                    if (audio_device_is_digital(deviceType)) {
+                        device->importAudioPort(profile);
                     }
                     break;
                 }
@@ -4300,20 +4233,20 @@
             }
 
             ALOGV("opening output for device %08x with params %s profile %p name %s",
-                  device, address.string(), profile.get(), profile->getName().string());
+                  deviceType, address.string(), profile.get(), profile->getName().string());
             desc = new SwAudioOutputDescriptor(profile, mpClientInterface);
             audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
-            status_t status = desc->open(nullptr, device, address,
+            status_t status = desc->open(nullptr, DeviceVector(device),
                                          AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
 
             if (status == NO_ERROR) {
                 // Here is where the out_set_parameters() for card & device gets called
                 if (!address.isEmpty()) {
-                    char *param = audio_device_address_to_parameter(device, address);
+                    char *param = audio_device_address_to_parameter(deviceType, address);
                     mpClientInterface->setParameters(output, String8(param));
                     free(param);
                 }
-                updateAudioProfiles(devDesc, output, profile->getAudioProfiles());
+                updateAudioProfiles(device, output, profile->getAudioProfiles());
                 if (!profile->hasValidAudioProfile()) {
                     ALOGW("checkOutputsForDevice() missing param");
                     desc->close();
@@ -4328,7 +4261,8 @@
                     config.offload_info.channel_mask = config.channel_mask;
                     config.offload_info.format = config.format;
 
-                    status_t status = desc->open(&config, device, address, AUDIO_STREAM_DEFAULT,
+                    status_t status = desc->open(&config, DeviceVector(device),
+                                                 AUDIO_STREAM_DEFAULT,
                                                  AUDIO_OUTPUT_FLAG_NONE, &output);
                     if (status != NO_ERROR) {
                         output = AUDIO_IO_HANDLE_NONE;
@@ -4337,14 +4271,15 @@
 
                 if (output != AUDIO_IO_HANDLE_NONE) {
                     addOutput(output, desc);
-                    if (device_distinguishes_on_address(device) && address != "0") {
+                    if (device_distinguishes_on_address(deviceType) && address != "0") {
                         sp<AudioPolicyMix> policyMix;
-                        if (mPolicyMixes.getAudioPolicyMix(address, policyMix) != NO_ERROR) {
-                            ALOGE("checkOutputsForDevice() cannot find policy for address %s",
+                        if (mPolicyMixes.getAudioPolicyMix(address, policyMix) == NO_ERROR) {
+                            policyMix->setOutput(desc);
+                            desc->mPolicyMix = policyMix->getMix();
+                        } else {
+                            ALOGW("checkOutputsForDevice() cannot find policy for address %s",
                                   address.string());
                         }
-                        policyMix->setOutput(desc);
-                        desc->mPolicyMix = policyMix->getMix();
 
                     } else if (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) &&
                                     hasPrimaryOutput()) {
@@ -4376,28 +4311,28 @@
                 output = AUDIO_IO_HANDLE_NONE;
             }
             if (output == AUDIO_IO_HANDLE_NONE) {
-                ALOGW("checkOutputsForDevice() could not open output for device %x", device);
+                ALOGW("checkOutputsForDevice() could not open output for device %x", deviceType);
                 profiles.removeAt(profile_index);
                 profile_index--;
             } else {
                 outputs.add(output);
                 // Load digital format info only for digital devices
-                if (audio_device_is_digital(device)) {
-                    devDesc->importAudioPort(profile);
+                if (audio_device_is_digital(deviceType)) {
+                    device->importAudioPort(profile);
                 }
 
-                if (device_distinguishes_on_address(device)) {
-                    ALOGV("checkOutputsForDevice(): setOutputDevice(dev=0x%x, addr=%s)",
-                            device, address.string());
-                    setOutputDevice(desc, device, true/*force*/, 0/*delay*/,
-                            NULL/*patch handle*/, address.string());
+                if (device_distinguishes_on_address(deviceType)) {
+                    ALOGV("checkOutputsForDevice(): setOutputDevices %s",
+                            device->toString().c_str());
+                    setOutputDevices(desc, DeviceVector(device), true/*force*/, 0/*delay*/,
+                                     NULL/*patch handle*/);
                 }
                 ALOGV("checkOutputsForDevice(): adding output %d", output);
             }
         }
 
         if (profiles.isEmpty()) {
-            ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
+            ALOGW("checkOutputsForDevice(): No output available for device %04x", deviceType);
             return BAD_VALUE;
         }
     } else { // Disconnect
@@ -4406,10 +4341,9 @@
             desc = mOutputs.valueAt(i);
             if (!desc->isDuplicated()) {
                 // exact match on device
-                if (device_distinguishes_on_address(device) &&
-                        (desc->supportedDevices() == device)) {
-                    findIoHandlesByAddress(desc, device, address, outputs);
-                } else if (!(desc->supportedDevices() & mAvailableOutputDevices.types())) {
+                if (device_distinguishes_on_address(deviceType) && desc->supportsDevice(device)) {
+                    outputs.add(mOutputs.keyAt(i));
+                } else if (!mAvailableOutputDevices.containsAtLeastOne(desc->supportedDevices())) {
                     ALOGV("checkOutputsForDevice(): disconnecting adding output %d",
                             mOutputs.keyAt(i));
                     outputs.add(mOutputs.keyAt(i));
@@ -4420,7 +4354,7 @@
         for (const auto& hwModule : mHwModules) {
             for (size_t j = 0; j < hwModule->getOutputProfiles().size(); j++) {
                 sp<IOProfile> profile = hwModule->getOutputProfiles()[j];
-                if (profile->supportDevice(device)) {
+                if (profile->supportsDevice(device)) {
                     ALOGV("checkOutputsForDevice(): "
                             "clearing direct output profile %zu on module %s",
                             j, hwModule->getName());
@@ -4432,24 +4366,22 @@
     return NO_ERROR;
 }
 
-status_t AudioPolicyManager::checkInputsForDevice(const sp<DeviceDescriptor>& devDesc,
+status_t AudioPolicyManager::checkInputsForDevice(const sp<DeviceDescriptor>& device,
                                                   audio_policy_dev_state_t state,
-                                                  SortedVector<audio_io_handle_t>& inputs,
-                                                  const String8& address)
+                                                  SortedVector<audio_io_handle_t>& inputs)
 {
-    audio_devices_t device = devDesc->type();
     sp<AudioInputDescriptor> desc;
 
-    if (audio_device_is_digital(device)) {
+    if (audio_device_is_digital(device->type())) {
         // erase all current sample rates, formats and channel masks
-        devDesc->clearAudioProfiles();
+        device->clearAudioProfiles();
     }
 
     if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
         // first list already open inputs that can be routed to this device
         for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
             desc = mInputs.valueAt(input_index);
-            if (desc->mProfile->supportDevice(device)) {
+            if (desc->mProfile->supportsDeviceTypes(device->type())) {
                 ALOGV("checkInputsForDevice(): adding opened input %d", mInputs.keyAt(input_index));
                inputs.add(mInputs.keyAt(input_index));
             }
@@ -4463,19 +4395,16 @@
                  profile_index++) {
                 sp<IOProfile> profile = hwModule->getInputProfiles()[profile_index];
 
-                if (profile->supportDevice(device)) {
-                    if (!device_distinguishes_on_address(device) ||
-                            profile->supportDeviceAddress(address)) {
-                        profiles.add(profile);
-                        ALOGV("checkInputsForDevice(): adding profile %zu from module %s",
-                                profile_index, hwModule->getName());
-                    }
+                if (profile->supportsDevice(device)) {
+                    profiles.add(profile);
+                    ALOGV("checkInputsForDevice(): adding profile %zu from module %s",
+                          profile_index, hwModule->getName());
                 }
             }
         }
 
         if (profiles.isEmpty() && inputs.isEmpty()) {
-            ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
+            ALOGW("%s: No input available for device %s", __func__, device->toString().c_str());
             return BAD_VALUE;
         }
 
@@ -4490,8 +4419,8 @@
             for (input_index = 0; input_index < mInputs.size(); input_index++) {
                 desc = mInputs.valueAt(input_index);
                 if (desc->mProfile == profile) {
-                    if (audio_device_is_digital(device)) {
-                        devDesc->importAudioPort(profile);
+                    if (audio_device_is_digital(device->type())) {
+                        device->importAudioPort(profile);
                     }
                     break;
                 }
@@ -4510,18 +4439,18 @@
             audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
             status_t status = desc->open(nullptr,
                                          device,
-                                         address,
                                          AUDIO_SOURCE_MIC,
                                          AUDIO_INPUT_FLAG_NONE,
                                          &input);
 
             if (status == NO_ERROR) {
+                const String8& address = device->address();
                 if (!address.isEmpty()) {
-                    char *param = audio_device_address_to_parameter(device, address);
+                    char *param = audio_device_address_to_parameter(device->type(), address);
                     mpClientInterface->setParameters(input, String8(param));
                     free(param);
                 }
-                updateAudioProfiles(devDesc, input, profile->getAudioProfiles());
+                updateAudioProfiles(device, input, profile->getAudioProfiles());
                 if (!profile->hasValidAudioProfile()) {
                     ALOGW("checkInputsForDevice() direct input missing param");
                     desc->close();
@@ -4534,20 +4463,21 @@
             } // endif input != 0
 
             if (input == AUDIO_IO_HANDLE_NONE) {
-                ALOGW("checkInputsForDevice() could not open input for device 0x%X", device);
+                ALOGW("%s could not open input for device %s", __func__,  
+                       device->toString().c_str());
                 profiles.removeAt(profile_index);
                 profile_index--;
             } else {
                 inputs.add(input);
-                if (audio_device_is_digital(device)) {
-                    devDesc->importAudioPort(profile);
+                if (audio_device_is_digital(device->type())) {
+                    device->importAudioPort(profile);
                 }
                 ALOGV("checkInputsForDevice(): adding input %d", input);
             }
         } // end scan profiles
 
         if (profiles.isEmpty()) {
-            ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
+            ALOGW("%s: No input available for device %s", __func__,  device->toString().c_str());
             return BAD_VALUE;
         }
     } else {
@@ -4555,7 +4485,7 @@
         // check if one opened input is not needed any more after disconnecting one device
         for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
             desc = mInputs.valueAt(input_index);
-            if (!(desc->mProfile->supportDevice(mAvailableInputDevices.types()))) {
+            if (!mAvailableInputDevices.containsAtLeastOne(desc->supportedDevices())) {
                 ALOGV("checkInputsForDevice(): disconnecting adding input %d",
                       mInputs.keyAt(input_index));
                 inputs.add(mInputs.keyAt(input_index));
@@ -4567,7 +4497,7 @@
                  profile_index < hwModule->getInputProfiles().size();
                  profile_index++) {
                 sp<IOProfile> profile = hwModule->getInputProfiles()[profile_index];
-                if (profile->supportDevice(device)) {
+                if (profile->supportsDevice(device)) {
                     ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %s",
                             profile_index, hwModule->getName());
                     profile->clearAudioProfiles();
@@ -4641,7 +4571,7 @@
 
     // MSD patches may have been released to support a non-MSD direct output. Reset MSD patch if
     // no direct outputs are open.
-    if (mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD) != 0) {
+    if (!getMsdAudioOutDevices().isEmpty()) {
         bool directOutputOpen = false;
         for (size_t i = 0; i < mOutputs.size(); i++) {
             if (mOutputs[i]->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
@@ -4668,7 +4598,7 @@
 
     nextAudioPortGeneration();
 
-    audio_devices_t device = inputDesc->mDevice;
+    sp<DeviceDescriptor> device = inputDesc->getDevice();
     ssize_t index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
     if (index >= 0) {
         sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
@@ -4680,26 +4610,26 @@
     inputDesc->close();
     mInputs.removeItem(input);
 
-    audio_devices_t primaryInputDevices = availablePrimaryInputDevices();
-    if (((device & primaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) &&
+    DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
+    if (primaryInputDevices.contains(device) &&
             mInputs.activeInputsCountOnDevices(primaryInputDevices) == 0) {
         SoundTrigger::setCaptureState(false);
     }
 }
 
-SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(
-                                                                audio_devices_t device,
-                                                                const SwAudioOutputCollection& openOutputs)
+SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevices(
+            const DeviceVector &devices,
+            const SwAudioOutputCollection& openOutputs)
 {
     SortedVector<audio_io_handle_t> outputs;
 
-    ALOGVV("getOutputsForDevice() device %04x", device);
+    ALOGVV("%s() devices %s", __func__, devices.toString().c_str());
     for (size_t i = 0; i < openOutputs.size(); i++) {
-        ALOGVV("output %zu isDuplicated=%d device=%04x",
+        ALOGVV("output %zu isDuplicated=%d device=%s",
                 i, openOutputs.valueAt(i)->isDuplicated(),
-                openOutputs.valueAt(i)->supportedDevices());
-        if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) {
-            ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i));
+                openOutputs.valueAt(i)->supportedDevices().toString().c_str());
+        if (openOutputs.valueAt(i)->supportsAllDevices(devices)) {
+            ALOGVV("%s() found output %d", __func__, openOutputs.keyAt(i));
             outputs.add(openOutputs.keyAt(i));
         }
     }
@@ -4721,10 +4651,10 @@
 
 void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy)
 {
-    audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
-    audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
-    SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs);
-    SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
+    DeviceVector oldDevices = getDevicesForStrategy(strategy, true /*fromCache*/);
+    DeviceVector newDevices = getDevicesForStrategy(strategy, false /*fromCache*/);
+    SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevices(oldDevices, mPreviousOutputs);
+    SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevices(newDevices, mOutputs);
 
     // also take into account external policy-related changes: add all outputs which are
     // associated with policies in the "before" and "after" output vectors
@@ -4744,7 +4674,7 @@
         }
     }
 
-    if (srcOutputs != dstOutputs) {
+    if (!dstOutputs.isEmpty() && srcOutputs != dstOutputs) {
         // get maximum latency of all source outputs to determine the minimum mute time guaranteeing
         // audio from invalidated tracks will be rendered when unmuting
         uint32_t maxLatency = 0;
@@ -4754,14 +4684,16 @@
                 maxLatency = desc->latency();
             }
         }
-        ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d",
-              strategy, srcOutputs[0], dstOutputs[0]);
+        ALOGV("%s: strategy %d, moving from output %s to output %s", __func__, strategy,
+              (srcOutputs.isEmpty()? "none" : std::to_string(srcOutputs[0]).c_str()),
+              (dstOutputs.isEmpty()? "none" : std::to_string(dstOutputs[0]).c_str()));
         // mute strategy while moving tracks from one output to another
         for (audio_io_handle_t srcOut : srcOutputs) {
             sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueFor(srcOut);
             if (desc != 0 && isStrategyActive(desc, strategy)) {
                 setStrategyMute(strategy, true, desc);
-                setStrategyMute(strategy, false, desc, maxLatency * LATENCY_MUTE_FACTOR, newDevice);
+                setStrategyMute(strategy, false, desc, maxLatency * LATENCY_MUTE_FACTOR,
+                                newDevices.types());
             }
             sp<SourceClientDescriptor> source =
                     getSourceForStrategyOnOutput(srcOut, strategy);
@@ -4880,26 +4812,28 @@
     return device;
 }
 
-audio_devices_t AudioPolicyManager::getNewOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
-                                                       bool fromCache)
+DeviceVector AudioPolicyManager::getNewOutputDevices(const sp<SwAudioOutputDescriptor>& outputDesc,
+                                                     bool fromCache)
 {
+    DeviceVector devices;
+
     ssize_t index = mAudioPatches.indexOfKey(outputDesc->getPatchHandle());
     if (index >= 0) {
         sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
         if (patchDesc->mUid != mUidCached) {
-            ALOGV("getNewOutputDevice() device %08x forced by patch %d",
-                  outputDesc->device(), outputDesc->getPatchHandle());
-            return outputDesc->device();
+            ALOGV("%s device %s forced by patch %d", __func__,
+                  outputDesc->devices().toString().c_str(), outputDesc->getPatchHandle());
+            return  outputDesc->devices();
         }
     }
 
     // Honor explicit routing requests only if no client using default routing is active on this
     // input: a specific app can not force routing for other apps by setting a preferred device.
     bool active; // unused
-    sp<DeviceDescriptor> deviceDesc =
+    sp<DeviceDescriptor> device =
         findPreferredDevice(outputDesc, STRATEGY_NONE, active, mAvailableOutputDevices);
-    if (deviceDesc != nullptr) {
-        return deviceDesc->type();
+    if (device != nullptr) {
+        return DeviceVector(device);
     }
 
     // check the following by order of priority to request a routing change if necessary:
@@ -4925,66 +4859,65 @@
     // FIXME: extend use of isStrategyActiveOnSameModule() to all strategies
     // with a refined rule considering mutually exclusive devices (using same backend)
     // as opposed to all streams on the same audio HAL module.
-    audio_devices_t device = AUDIO_DEVICE_NONE;
     if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE) &&
         mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
-        device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
+        devices = getDevicesForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
     } else if (isInCall() ||
                isStrategyActiveOnSameModule(outputDesc, STRATEGY_PHONE)) {
-        device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
+        devices = getDevicesForStrategy(STRATEGY_PHONE, fromCache);
     } else if (isStrategyActiveOnSameModule(outputDesc, STRATEGY_SONIFICATION)) {
-        device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
+        devices = getDevicesForStrategy(STRATEGY_SONIFICATION, fromCache);
     } else if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE)) {
-        device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
+        devices = getDevicesForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
     } else if (isStrategyActive(outputDesc, STRATEGY_ACCESSIBILITY)) {
-        device = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, fromCache);
+        devices = getDevicesForStrategy(STRATEGY_ACCESSIBILITY, fromCache);
     } else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION_RESPECTFUL)) {
-        device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
+        devices = getDevicesForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
     } else if (isStrategyActive(outputDesc, STRATEGY_MEDIA)) {
-        device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
+        devices = getDevicesForStrategy(STRATEGY_MEDIA, fromCache);
     } else if (isStrategyActive(outputDesc, STRATEGY_DTMF)) {
-        device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
+        devices = getDevicesForStrategy(STRATEGY_DTMF, fromCache);
     } else if (isStrategyActive(outputDesc, STRATEGY_TRANSMITTED_THROUGH_SPEAKER)) {
-        device = getDeviceForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache);
+        devices = getDevicesForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache);
     } else if (isStrategyActive(outputDesc, STRATEGY_REROUTING)) {
-        device = getDeviceForStrategy(STRATEGY_REROUTING, fromCache);
+        devices = getDevicesForStrategy(STRATEGY_REROUTING, fromCache);
     }
 
-    ALOGV("getNewOutputDevice() selected device %x", device);
-    return device;
+    ALOGV("getNewOutputDevice() selected devices %s", devices.toString().c_str());
+    return devices;
 }
 
-audio_devices_t AudioPolicyManager::getNewInputDevice(const sp<AudioInputDescriptor>& inputDesc)
+sp<DeviceDescriptor> AudioPolicyManager::getNewInputDevice(
+        const sp<AudioInputDescriptor>& inputDesc)
 {
-    audio_devices_t device = AUDIO_DEVICE_NONE;
+    sp<DeviceDescriptor> device;
 
     ssize_t index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
     if (index >= 0) {
         sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
         if (patchDesc->mUid != mUidCached) {
-            ALOGV("getNewInputDevice() device %08x forced by patch %d",
-                  inputDesc->mDevice, inputDesc->getPatchHandle());
-            return inputDesc->mDevice;
+            ALOGV("getNewInputDevice() device %s forced by patch %d",
+                  inputDesc->getDevice()->toString().c_str(), inputDesc->getPatchHandle());
+            return inputDesc->getDevice();
         }
     }
 
     // Honor explicit routing requests only if no client using default routing is active on this
     // input: a specific app can not force routing for other apps by setting a preferred device.
     bool active;
-    sp<DeviceDescriptor> deviceDesc =
-        findPreferredDevice(inputDesc, AUDIO_SOURCE_DEFAULT, active, mAvailableInputDevices);
-    if (deviceDesc != nullptr) {
-        return deviceDesc->type();
+    device = findPreferredDevice(inputDesc, AUDIO_SOURCE_DEFAULT, active, mAvailableInputDevices);
+    if (device != nullptr) {
+        return device;
     }
 
     // If we are not in call and no client is active on this input, this methods returns
     // AUDIO_DEVICE_NONE, causing the patch on the input stream to be released.
-    audio_source_t source = inputDesc->source();
-    if (source == AUDIO_SOURCE_DEFAULT && isInCall()) {
-        source = AUDIO_SOURCE_VOICE_COMMUNICATION;
+    audio_attributes_t attributes = inputDesc->getHighestPriorityAttributes();
+    if (attributes.source == AUDIO_SOURCE_DEFAULT && isInCall()) {
+        attributes.source = AUDIO_SOURCE_VOICE_COMMUNICATION;
     }
-    if (source != AUDIO_SOURCE_DEFAULT) {
-        device = getDeviceAndMixForInputSource(source);
+    if (attributes.source != AUDIO_SOURCE_DEFAULT) {
+        device = getDeviceAndMixForAttributes(attributes);
     }
 
     return device;
@@ -5006,36 +4939,37 @@
     if (stream < (audio_stream_type_t) 0 || stream >= AUDIO_STREAM_PUBLIC_CNT) {
         return AUDIO_DEVICE_NONE;
     }
-    audio_devices_t activeDevices = AUDIO_DEVICE_NONE;
-    audio_devices_t devices = AUDIO_DEVICE_NONE;
+    DeviceVector activeDevices;
+    DeviceVector devices;
     for (int curStream = 0; curStream < AUDIO_STREAM_FOR_POLICY_CNT; curStream++) {
         if (!streamsMatchForvolume(stream, (audio_stream_type_t)curStream)) {
             continue;
         }
         routing_strategy curStrategy = getStrategy((audio_stream_type_t)curStream);
-        audio_devices_t curDevices =
-                getDeviceForStrategy((routing_strategy)curStrategy, false /*fromCache*/);
-        devices |= curDevices;
-        for (audio_io_handle_t output : getOutputsForDevice(curDevices, mOutputs)) {
+        DeviceVector curDevices =
+                getDevicesForStrategy((routing_strategy)curStrategy, false /*fromCache*/);
+        devices.merge(curDevices);
+        for (audio_io_handle_t output : getOutputsForDevices(curDevices, mOutputs)) {
             sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
             if (outputDesc->isStreamActive((audio_stream_type_t)curStream)) {
-                activeDevices |= outputDesc->device();
+                activeDevices.merge(outputDesc->devices());
             }
         }
     }
 
     // Favor devices selected on active streams if any to report correct device in case of
     // explicit device selection
-    if (activeDevices != AUDIO_DEVICE_NONE) {
+    if (!activeDevices.isEmpty()) {
         devices = activeDevices;
     }
     /*Filter SPEAKER_SAFE out of results, as AudioService doesn't know about it
       and doesn't really need to.*/
-    if (devices & AUDIO_DEVICE_OUT_SPEAKER_SAFE) {
-        devices |= AUDIO_DEVICE_OUT_SPEAKER;
-        devices &= ~AUDIO_DEVICE_OUT_SPEAKER_SAFE;
+    DeviceVector speakerSafeDevices = devices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER_SAFE);
+    if (!speakerSafeDevices.isEmpty()) {
+        devices.merge(mAvailableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER));
+        devices.remove(speakerSafeDevices);
     }
-    return devices;
+    return devices.types();
 }
 
 routing_strategy AudioPolicyManager::getStrategy(audio_stream_type_t stream) const
@@ -5126,34 +5060,33 @@
     return 0;
 }
 
-audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy,
-                                                         bool fromCache)
+DeviceVector AudioPolicyManager::getDevicesForStrategy(routing_strategy strategy, bool fromCache)
 {
     // Honor explicit routing requests only if all active clients have a preferred route in which
     // case the last active client route is used
-    sp<DeviceDescriptor> deviceDesc = findPreferredDevice(mOutputs, strategy, mAvailableOutputDevices);
-    if (deviceDesc != nullptr) {
-        return deviceDesc->type();
+    sp<DeviceDescriptor> device = findPreferredDevice(mOutputs, strategy, mAvailableOutputDevices);
+    if (device != nullptr) {
+        return DeviceVector(device);
     }
 
     if (fromCache) {
-        ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x",
-              strategy, mDeviceForStrategy[strategy]);
-        return mDeviceForStrategy[strategy];
+        ALOGVV("%s from cache strategy %d, device %s", __func__, strategy,
+               mDevicesForStrategy[strategy].toString().c_str());
+        return mDevicesForStrategy[strategy];
     }
-    return mEngine->getDeviceForStrategy(strategy);
+    return mAvailableOutputDevices.getDevicesFromTypeMask(mEngine->getDeviceForStrategy(strategy));
 }
 
 void AudioPolicyManager::updateDevicesAndOutputs()
 {
     for (int i = 0; i < NUM_STRATEGIES; i++) {
-        mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
+        mDevicesForStrategy[i] = getDevicesForStrategy((routing_strategy)i, false /*fromCache*/);
     }
     mPreviousOutputs = mOutputs;
 }
 
 uint32_t AudioPolicyManager::checkDeviceMuteStrategies(const sp<AudioOutputDescriptor>& outputDesc,
-                                                       audio_devices_t prevDevice,
+                                                       audio_devices_t prevDeviceType,
                                                        uint32_t delayMs)
 {
     // mute/unmute strategies using an incompatible device combination
@@ -5164,13 +5097,14 @@
     }
 
     uint32_t muteWaitMs = 0;
-    audio_devices_t device = outputDesc->device();
-    bool shouldMute = outputDesc->isActive() && (popcount(device) >= 2);
+    audio_devices_t deviceType = outputDesc->devices().types();
+    bool shouldMute = outputDesc->isActive() && (popcount(deviceType) >= 2);
 
     for (size_t i = 0; i < NUM_STRATEGIES; i++) {
-        audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
-        curDevice = curDevice & outputDesc->supportedDevices();
-        bool mute = shouldMute && (curDevice & device) && (curDevice != device);
+        audio_devices_t curDeviceType =
+                getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
+        curDeviceType = curDeviceType & outputDesc->supportedDevices().types();
+        bool mute = shouldMute && (curDeviceType & deviceType) && (curDeviceType != deviceType);
         bool doMute = false;
 
         if (mute && !outputDesc->mStrategyMutedByDevice[i]) {
@@ -5184,12 +5118,11 @@
             for (size_t j = 0; j < mOutputs.size(); j++) {
                 sp<AudioOutputDescriptor> desc = mOutputs.valueAt(j);
                 // skip output if it does not share any device with current output
-                if ((desc->supportedDevices() & outputDesc->supportedDevices())
-                        == AUDIO_DEVICE_NONE) {
+                if (!desc->supportedDevices().containsAtLeastOne(outputDesc->supportedDevices())) {
                     continue;
                 }
                 ALOGVV("checkDeviceMuteStrategies() %s strategy %zu (curDevice %04x)",
-                      mute ? "muting" : "unmuting", i, curDevice);
+                      mute ? "muting" : "unmuting", i, curDeviceType);
                 setStrategyMute((routing_strategy)i, mute, desc, mute ? 0 : delayMs);
                 if (isStrategyActive(desc, (routing_strategy)i)) {
                     if (mute) {
@@ -5209,7 +5142,7 @@
 
     // temporary mute output if device selection changes to avoid volume bursts due to
     // different per device volumes
-    if (outputDesc->isActive() && (device != prevDevice)) {
+    if (outputDesc->isActive() && (deviceType != prevDeviceType)) {
         uint32_t tempMuteWaitMs = outputDesc->latency() * 2;
         // temporary mute duration is conservatively set to 4 times the reported latency
         uint32_t tempMuteDurationMs = outputDesc->latency() * 4;
@@ -5223,7 +5156,7 @@
                 // delayed device change
                 setStrategyMute((routing_strategy)i, true, outputDesc, delayMs);
                 setStrategyMute((routing_strategy)i, false, outputDesc,
-                                delayMs + tempMuteDurationMs, device);
+                                delayMs + tempMuteDurationMs, deviceType);
             }
         }
     }
@@ -5237,46 +5170,45 @@
     return 0;
 }
 
-uint32_t AudioPolicyManager::setOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
-                                             audio_devices_t device,
-                                             bool force,
-                                             int delayMs,
-                                             audio_patch_handle_t *patchHandle,
-                                             const char *address,
-                                             bool requiresMuteCheck)
+uint32_t AudioPolicyManager::setOutputDevices(const sp<SwAudioOutputDescriptor>& outputDesc,
+                                              const DeviceVector &devices,
+                                              bool force,
+                                              int delayMs,
+                                              audio_patch_handle_t *patchHandle,
+                                              bool requiresMuteCheck)
 {
-    ALOGV("setOutputDevice() device %04x delayMs %d", device, delayMs);
-    AudioParameter param;
+    ALOGV("%s device %s delayMs %d", __func__, devices.toString().c_str(), delayMs);
     uint32_t muteWaitMs;
 
     if (outputDesc->isDuplicated()) {
-        muteWaitMs = setOutputDevice(outputDesc->subOutput1(), device, force, delayMs,
-                nullptr /* patchHandle */, nullptr /* address */, requiresMuteCheck);
-        muteWaitMs += setOutputDevice(outputDesc->subOutput2(), device, force, delayMs,
-                nullptr /* patchHandle */, nullptr /* address */, requiresMuteCheck);
+        muteWaitMs = setOutputDevices(outputDesc->subOutput1(), devices, force, delayMs,
+                nullptr /* patchHandle */, requiresMuteCheck);
+        muteWaitMs += setOutputDevices(outputDesc->subOutput2(), devices, force, delayMs,
+                nullptr /* patchHandle */, requiresMuteCheck);
         return muteWaitMs;
     }
-    // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
-    // output profile
-    if ((device != AUDIO_DEVICE_NONE) &&
-            ((device & outputDesc->supportedDevices()) == AUDIO_DEVICE_NONE)) {
-        return 0;
-    }
 
     // filter devices according to output selected
-    device = (audio_devices_t)(device & outputDesc->supportedDevices());
+    DeviceVector filteredDevices = outputDesc->filterSupportedDevices(devices);
 
-    audio_devices_t prevDevice = outputDesc->mDevice;
+    // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
+    // output profile
+    if (!devices.isEmpty() && filteredDevices.isEmpty()) {
+        ALOGV("%s: unsupported device %s for output", __func__, devices.toString().c_str());
+        return 0;
+    }
 
-    ALOGV("setOutputDevice() prevDevice 0x%04x", prevDevice);
+    DeviceVector prevDevices = outputDesc->devices();
 
-    if (device != AUDIO_DEVICE_NONE) {
-        outputDesc->mDevice = device;
+    ALOGV("setOutputDevices() prevDevice %s", prevDevices.toString().c_str());
+
+    if (!filteredDevices.isEmpty()) {
+        outputDesc->setDevices(filteredDevices);
     }
 
     // if the outputs are not materially active, there is no need to mute.
     if (requiresMuteCheck) {
-        muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs);
+        muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevices.types(), delayMs);
     } else {
         ALOGV("%s: suppressing checkDeviceMuteStrategies", __func__);
         muteWaitMs = 0;
@@ -5287,42 +5219,32 @@
     //      OR the requested device is the same as current device
     //  AND force is not specified
     //  AND the output is connected by a valid audio patch.
-    // Doing this check here allows the caller to call setOutputDevice() without conditions
-    if ((device == AUDIO_DEVICE_NONE || device == prevDevice) &&
-        !force &&
-        outputDesc->getPatchHandle() != 0) {
-        ALOGV("setOutputDevice() setting same device 0x%04x or null device", device);
+    // Doing this check here allows the caller to call setOutputDevices() without conditions
+    if ((!filteredDevices.isEmpty() || filteredDevices == prevDevices) &&
+            !force && outputDesc->getPatchHandle() != 0) {
+        ALOGV("%s setting same device %s or null device, force=%d, patch handle=%d", __func__,
+              filteredDevices.toString().c_str(), force, outputDesc->getPatchHandle());
         return muteWaitMs;
     }
 
-    ALOGV("setOutputDevice() changing device");
+    ALOGV("%s changing device to %s", __func__, filteredDevices.toString().c_str());
 
     // do the routing
-    if (device == AUDIO_DEVICE_NONE) {
+    if (filteredDevices.isEmpty()) {
         resetOutputDevice(outputDesc, delayMs, NULL);
     } else {
-        DeviceVector deviceList;
-        if ((address == NULL) || (strlen(address) == 0)) {
-            deviceList = mAvailableOutputDevices.getDevicesFromTypeMask(device);
-        } else {
-            sp<DeviceDescriptor> deviceDesc = mAvailableOutputDevices.getDevice(
-                    device, String8(address));
-            if (deviceDesc) deviceList.add(deviceDesc);
+        PatchBuilder patchBuilder;
+        patchBuilder.addSource(outputDesc);
+        ALOG_ASSERT(filteredDevices.size() <= AUDIO_PATCH_PORTS_MAX, "Too many sink ports");
+        for (const auto &filteredDevice : filteredDevices) {
+            patchBuilder.addSink(filteredDevice);
         }
 
-        if (!deviceList.isEmpty()) {
-            PatchBuilder patchBuilder;
-            patchBuilder.addSource(outputDesc);
-            ALOG_ASSERT(deviceList.size() <= AUDIO_PATCH_PORTS_MAX, "Too many sink ports");
-            for (const auto &device : deviceList) {
-                patchBuilder.addSink(device);
-            }
-            installPatch(__func__, patchHandle, outputDesc.get(), patchBuilder.patch(), delayMs);
-        }
+        installPatch(__func__, patchHandle, outputDesc.get(), patchBuilder.patch(), delayMs);
     }
 
     // update stream volumes according to new device
-    applyStreamVolumes(outputDesc, device, delayMs);
+    applyStreamVolumes(outputDesc, filteredDevices.types(), delayMs);
 
     return muteWaitMs;
 }
@@ -5351,18 +5273,17 @@
 }
 
 status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input,
-                                            audio_devices_t device,
+                                            const sp<DeviceDescriptor> &device,
                                             bool force,
                                             audio_patch_handle_t *patchHandle)
 {
     status_t status = NO_ERROR;
 
     sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
-    if ((device != AUDIO_DEVICE_NONE) && ((device != inputDesc->mDevice) || force)) {
-        inputDesc->mDevice = device;
+    if ((device != nullptr) && ((device != inputDesc->getDevice()) || force)) {
+        inputDesc->setDevice(device);
 
-        DeviceVector deviceList = mAvailableInputDevices.getDevicesFromTypeMask(device);
-        if (!deviceList.isEmpty()) {
+        if (mAvailableInputDevices.contains(device)) {
             PatchBuilder patchBuilder;
             patchBuilder.addSink(inputDesc,
             // AUDIO_SOURCE_HOTWORD is for internal use only:
@@ -5374,7 +5295,7 @@
                         }
                         return result; }).
             //only one input device for now
-                    addSource(deviceList.itemAt(0));
+                    addSource(device);
             status = installPatch(__func__, patchHandle, inputDesc.get(), patchBuilder.patch(), 0);
         }
     }
@@ -5404,8 +5325,7 @@
     return status;
 }
 
-sp<IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device,
-                                                  const String8& address,
+sp<IOProfile> AudioPolicyManager::getInputProfile(const sp<DeviceDescriptor> &device,
                                                   uint32_t& samplingRate,
                                                   audio_format_t& format,
                                                   audio_channel_mask_t& channelMask,
@@ -5425,7 +5345,7 @@
         for (const auto& profile : hwModule->getInputProfiles()) {
             // profile->log();
             //updatedFormat = format;
-            if (profile->isCompatibleProfile(device, address, samplingRate,
+            if (profile->isCompatibleProfile(DeviceVector(device), samplingRate,
                                              &samplingRate  /*updatedSamplingRate*/,
                                              format,
                                              &format,       /*updatedFormat*/
@@ -5436,7 +5356,7 @@
                                              true /*exactMatchRequiredForInputFlags*/)) {
                 return profile;
             }
-            if (firstInexact == nullptr && profile->isCompatibleProfile(device, address,
+            if (firstInexact == nullptr && profile->isCompatibleProfile(DeviceVector(device),
                                              samplingRate,
                                              &updatedSamplingRate,
                                              format,
@@ -5460,32 +5380,33 @@
     return NULL;
 }
 
-
-audio_devices_t AudioPolicyManager::getDeviceAndMixForInputSource(audio_source_t inputSource,
-                                                                  AudioMix **policyMix)
+sp<DeviceDescriptor> AudioPolicyManager::getDeviceAndMixForAttributes(
+        const audio_attributes_t &attributes, AudioMix **policyMix)
 {
     // Honor explicit routing requests only if all active clients have a preferred route in which
     // case the last active client route is used
-    sp<DeviceDescriptor> deviceDesc =
-        findPreferredDevice(mInputs, inputSource, mAvailableInputDevices);
-    if (deviceDesc != nullptr) {
-        return deviceDesc->type();
+    sp<DeviceDescriptor> device =
+        findPreferredDevice(mInputs, attributes.source, mAvailableInputDevices);
+    if (device != nullptr) {
+        return device;
     }
 
-
-    audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
-    audio_devices_t selectedDeviceFromMix =
-           mPolicyMixes.getDeviceAndMixForInputSource(inputSource, availableDeviceTypes, policyMix);
-
-    if (selectedDeviceFromMix != AUDIO_DEVICE_NONE) {
-        return selectedDeviceFromMix;
-    }
-    return getDeviceForInputSource(inputSource);
+    sp<DeviceDescriptor> selectedDeviceFromMix =
+           mPolicyMixes.getDeviceAndMixForInputSource(attributes.source, mAvailableInputDevices,
+                                                      policyMix);
+    return (selectedDeviceFromMix != nullptr) ?
+           selectedDeviceFromMix : getDeviceForAttributes(attributes);
 }
 
-audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource)
+sp<DeviceDescriptor> AudioPolicyManager::getDeviceForAttributes(const audio_attributes_t &attributes)
 {
-    return mEngine->getDeviceForInputSource(inputSource);
+    audio_devices_t device = mEngine->getDeviceForInputSource(attributes.source);
+    if (attributes.source == AUDIO_SOURCE_REMOTE_SUBMIX &&
+                strncmp(attributes.tags, "addr=", strlen("addr=")) == 0) {
+        return mAvailableInputDevices.getDevice(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+                                                String8(attributes.tags + strlen("addr=")));
+    }
+    return mAvailableInputDevices.getDevice(device);
 }
 
 float AudioPolicyManager::computeVolume(audio_stream_type_t stream,
@@ -5630,7 +5551,7 @@
     }
 
     if (device == AUDIO_DEVICE_NONE) {
-        device = outputDesc->device();
+        device = outputDesc->devices().types();
     }
 
     float volumeDb = computeVolume(stream, index, device);
@@ -5701,7 +5622,7 @@
                                            audio_devices_t device)
 {
     if (device == AUDIO_DEVICE_NONE) {
-        device = outputDesc->device();
+        device = outputDesc->devices().types();
     }
 
     ALOGVV("setStreamMute() stream %d, mute %d, mMuteCount %d device %04x",
@@ -5798,9 +5719,9 @@
     return false;
 }
 
-bool AudioPolicyManager::isStrategyActiveOnSameModule(const sp<AudioOutputDescriptor>& outputDesc,
-                                          routing_strategy strategy, uint32_t inPastMs,
-                                          nsecs_t sysTime) const
+bool AudioPolicyManager::isStrategyActiveOnSameModule(const sp<SwAudioOutputDescriptor>& outputDesc,
+                                                      routing_strategy strategy, uint32_t inPastMs,
+                                                      nsecs_t sysTime) const
 {
     for (size_t i = 0; i < mOutputs.size(); i++) {
         sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
@@ -5859,6 +5780,8 @@
             releaseAudioPatch(patchDesc->mHandle, patchDesc->mUid);
         }
     }
+
+    mHwModules.cleanUpForDevice(deviceDesc);
 }
 
 void AudioPolicyManager::modifySurroundFormats(
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 9eb1dcf..e99de16 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -313,36 +313,40 @@
         //  where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
         //  before updateDevicesAndOutputs() is called.
         virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy,
-                                                     bool fromCache);
+                                                     bool fromCache)
+        {
+            return getDevicesForStrategy(strategy, fromCache).types();
+        }
+
+        DeviceVector getDevicesForStrategy(routing_strategy strategy, bool fromCache);
 
         bool isStrategyActive(const sp<AudioOutputDescriptor>& outputDesc, routing_strategy strategy,
                               uint32_t inPastMs = 0, nsecs_t sysTime = 0) const;
 
-        bool isStrategyActiveOnSameModule(const sp<AudioOutputDescriptor>& outputDesc,
-                                                  routing_strategy strategy, uint32_t inPastMs = 0,
-                                                  nsecs_t sysTime = 0) const;
+        bool isStrategyActiveOnSameModule(const sp<SwAudioOutputDescriptor>& outputDesc,
+                                          routing_strategy strategy, uint32_t inPastMs = 0,
+                                          nsecs_t sysTime = 0) const;
 
         // change the route of the specified output. Returns the number of ms we have slept to
         // allow new routing to take effect in certain cases.
-        virtual uint32_t setOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
-                             audio_devices_t device,
-                             bool force = false,
-                             int delayMs = 0,
-                             audio_patch_handle_t *patchHandle = NULL,
-                             const char *address = nullptr,
-                             bool requiresMuteCheck = true);
+        uint32_t setOutputDevices(const sp<SwAudioOutputDescriptor>& outputDesc,
+                                  const DeviceVector &device,
+                                  bool force = false,
+                                  int delayMs = 0,
+                                  audio_patch_handle_t *patchHandle = NULL,
+                                  bool requiresMuteCheck = true);
         status_t resetOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
                                    int delayMs = 0,
                                    audio_patch_handle_t *patchHandle = NULL);
         status_t setInputDevice(audio_io_handle_t input,
-                                audio_devices_t device,
+                                const sp<DeviceDescriptor> &device,
                                 bool force = false,
                                 audio_patch_handle_t *patchHandle = NULL);
         status_t resetInputDevice(audio_io_handle_t input,
                                   audio_patch_handle_t *patchHandle = NULL);
 
         // select input device corresponding to requested audio source
-        virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource);
+        sp<DeviceDescriptor> getDeviceForAttributes(const audio_attributes_t &attributes);
 
         // compute the actual volume for a given stream according to the requested index and a particular
         // device
@@ -391,15 +395,13 @@
         // when a device is disconnected, checks if an output is not used any more and
         // returns its handle if any.
         // transfers the audio tracks and effects from one output thread to another accordingly.
-        status_t checkOutputsForDevice(const sp<DeviceDescriptor>& devDesc,
+        status_t checkOutputsForDevice(const sp<DeviceDescriptor>& device,
                                        audio_policy_dev_state_t state,
-                                       SortedVector<audio_io_handle_t>& outputs,
-                                       const String8& address);
+                                       SortedVector<audio_io_handle_t>& outputs);
 
-        status_t checkInputsForDevice(const sp<DeviceDescriptor>& devDesc,
+        status_t checkInputsForDevice(const sp<DeviceDescriptor>& device,
                                       audio_policy_dev_state_t state,
-                                      SortedVector<audio_io_handle_t>& inputs,
-                                      const String8& address);
+                                      SortedVector<audio_io_handle_t>& inputs);
 
         // close an output and its companion duplicating output.
         void closeOutput(audio_io_handle_t output);
@@ -437,8 +439,8 @@
         // must be called every time a condition that affects the device choice for a given output is
         // changed: connected device, phone state, force use, output start, output stop..
         // see getDeviceForStrategy() for the use of fromCache parameter
-        audio_devices_t getNewOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
-                                           bool fromCache);
+        DeviceVector getNewOutputDevices(const sp<SwAudioOutputDescriptor>& outputDesc,
+                                         bool fromCache);
 
         // updates cache of device used by all strategies (mDeviceForStrategy[])
         // must be called every time a condition that affects the device choice for a given strategy is
@@ -448,7 +450,7 @@
         void updateDevicesAndOutputs();
 
         // selects the most appropriate device on input for current state
-        audio_devices_t getNewInputDevice(const sp<AudioInputDescriptor>& inputDesc);
+        sp<DeviceDescriptor> getNewInputDevice(const sp<AudioInputDescriptor>& inputDesc);
 
         virtual uint32_t getMaxEffectsCpuLoad()
         {
@@ -460,16 +462,16 @@
             return mEffects.getMaxEffectsMemory();
         }
 
-        SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
-                                                            const SwAudioOutputCollection& openOutputs);
+        SortedVector<audio_io_handle_t> getOutputsForDevices(
+                const DeviceVector &devices, const SwAudioOutputCollection& openOutputs);
 
         // mute/unmute strategies using an incompatible device combination
         // if muting, wait for the audio in pcm buffer to be drained before proceeding
         // if unmuting, unmute only after the specified delay
         // Returns the number of ms waited
         virtual uint32_t  checkDeviceMuteStrategies(const sp<AudioOutputDescriptor>& outputDesc,
-                                            audio_devices_t prevDevice,
-                                            uint32_t delayMs);
+                                                    audio_devices_t prevDeviceType,
+                                                    uint32_t delayMs);
 
         audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
@@ -477,13 +479,22 @@
                                        audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE,
                                        uint32_t samplingRate = 0);
         // samplingRate, format, channelMask are in/out and so may be modified
-        sp<IOProfile> getInputProfile(audio_devices_t device,
-                                      const String8& address,
+        sp<IOProfile> getInputProfile(const sp<DeviceDescriptor> & device,
                                       uint32_t& samplingRate,
                                       audio_format_t& format,
                                       audio_channel_mask_t& channelMask,
                                       audio_input_flags_t flags);
-        sp<IOProfile> getProfileForOutput(audio_devices_t device,
+        /**
+         * @brief getProfileForOutput
+         * @param devices vector of descriptors, may be empty if ignoring the device is required
+         * @param samplingRate
+         * @param format
+         * @param channelMask
+         * @param flags
+         * @param directOnly
+         * @return IOProfile to be used if found, nullptr otherwise
+         */
+        sp<IOProfile> getProfileForOutput(const DeviceVector &devices,
                                           uint32_t samplingRate,
                                           audio_format_t format,
                                           audio_channel_mask_t channelMask,
@@ -501,19 +512,19 @@
             return mAudioPatches.removeAudioPatch(handle);
         }
 
-        audio_devices_t availablePrimaryOutputDevices() const
+        DeviceVector availablePrimaryOutputDevices() const
         {
             if (!hasPrimaryOutput()) {
-                return AUDIO_DEVICE_NONE;
+                return DeviceVector();
             }
-            return mPrimaryOutput->supportedDevices() & mAvailableOutputDevices.types();
+            return mAvailableOutputDevices.filter(mPrimaryOutput->supportedDevices());
         }
-        audio_devices_t availablePrimaryInputDevices() const
+        DeviceVector availablePrimaryModuleInputDevices() const
         {
             if (!hasPrimaryOutput()) {
-                return AUDIO_DEVICE_NONE;
+                return DeviceVector();
             }
-            return mAvailableInputDevices.getDeviceTypesFromHwModule(
+            return mAvailableInputDevices.getDevicesFromHwModule(
                     mPrimaryOutput->getModuleHandle());
         }
         /**
@@ -530,8 +541,9 @@
             return (devices.size() > 0) ? devices.itemAt(0)->address() : String8("");
         }
 
-        uint32_t updateCallRouting(audio_devices_t rxDevice, uint32_t delayMs = 0);
-        sp<AudioPatch> createTelephonyPatch(bool isRx, audio_devices_t device, uint32_t delayMs);
+        uint32_t updateCallRouting(const DeviceVector &rxDevices, uint32_t delayMs = 0);
+        sp<AudioPatch> createTelephonyPatch(bool isRx, const sp<DeviceDescriptor> &device,
+                                            uint32_t delayMs);
         sp<DeviceDescriptor> findDevice(
                 const DeviceVector& devices, audio_devices_t device) const;
         audio_devices_t getModuleDeviceTypes(
@@ -581,7 +593,16 @@
         DeviceVector  mAvailableInputDevices;  // all available input devices
 
         bool    mLimitRingtoneVolume;        // limit ringtone volume to music volume if headset connected
-        audio_devices_t mDeviceForStrategy[NUM_STRATEGIES];
+
+        /**
+         * @brief mDevicesForStrategy vector of devices that are assigned for a given strategy.
+         * Note: in case of removal of device (@see setDeviceConnectionState), the device descriptor
+         * will be removed from the @see mAvailableOutputDevices or @see mAvailableInputDevices
+         * but the devices for strategies will be reevaluated within the
+         * @see setDeviceConnectionState function.
+         */
+        DeviceVector mDevicesForStrategy[NUM_STRATEGIES];
+
         float   mLastVoiceVolume;            // last voice volume value sent to audio HAL
         bool    mA2dpSuspended;  // true if A2DP output is suspended
 
@@ -637,13 +658,14 @@
 
         // Support for Multi-Stream Decoder (MSD) module
         sp<DeviceDescriptor> getMsdAudioInDevice() const;
+        DeviceVector getMsdAudioOutDevices() const;
         const AudioPatchCollection getMsdPatches() const;
-        status_t getBestMsdAudioProfileFor(audio_devices_t outputDevice,
+        status_t getBestMsdAudioProfileFor(const sp<DeviceDescriptor> &outputDevice,
                                            bool hwAvSync,
                                            audio_port_config *sourceConfig,
                                            audio_port_config *sinkConfig) const;
-        PatchBuilder buildMsdPatch(audio_devices_t outputDevice) const;
-        status_t setMsdPatch(audio_devices_t outputDevice = AUDIO_DEVICE_NONE);
+        PatchBuilder buildMsdPatch(const sp<DeviceDescriptor> &outputDevice) const;
+        status_t setMsdPatch(const sp<DeviceDescriptor> &outputDevice = nullptr);
 
         // If any, resolve any "dynamic" fields of an Audio Profiles collection
         void updateAudioProfiles(const sp<DeviceDescriptor>& devDesc, audio_io_handle_t ioHandle,
@@ -654,22 +676,12 @@
         // It can give a chance to HAL implementer to retrieve dynamic capabilities associated
         // to this device for example.
         // TODO avoid opening stream to retrieve capabilities of a profile.
-        void broadcastDeviceConnectionState(audio_devices_t device,
-                                            audio_policy_dev_state_t state,
-                                            const String8 &device_address);
+        void broadcastDeviceConnectionState(const sp<DeviceDescriptor> &device,
+                                            audio_policy_dev_state_t state);
 
         // updates device caching and output for streams that can influence the
         //    routing of notifications
         void handleNotificationRoutingForStream(audio_stream_type_t stream);
-        // find the outputs on a given output descriptor that have the given address.
-        // to be called on an AudioOutputDescriptor whose supported devices (as defined
-        //   in mProfile->mSupportedDevices) matches the device whose address is to be matched.
-        // see deviceDistinguishesOnAddress(audio_devices_t) for whether the device type is one
-        //   where addresses are used to distinguish between one connected device and another.
-        void findIoHandlesByAddress(const sp<SwAudioOutputDescriptor>& desc /*in*/,
-                const audio_devices_t device /*in*/,
-                const String8& address /*in*/,
-                SortedVector<audio_io_handle_t>& outputs /*out*/);
         uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; }
         // internal method, get audio_attributes_t from either a source audio_attributes_t
         // or audio_stream_type_t, respectively.
@@ -687,15 +699,14 @@
                 audio_output_flags_t *flags,
                 audio_port_handle_t *selectedDeviceId);
         // internal method to return the output handle for the given device and format
-        audio_io_handle_t getOutputForDevice(
-                audio_devices_t device,
+        audio_io_handle_t getOutputForDevices(
+                const DeviceVector &devices,
                 audio_session_t session,
                 audio_stream_type_t stream,
                 const audio_config_t *config,
                 audio_output_flags_t *flags);
         // internal method to return the input handle for the given device and format
-        audio_io_handle_t getInputForDevice(audio_devices_t device,
-                String8 address,
+        audio_io_handle_t getInputForDevice(const sp<DeviceDescriptor> &device,
                 audio_session_t session,
                 audio_source_t inputSource,
                 const audio_config_base_t *config,
@@ -713,14 +724,14 @@
 
         // select input device corresponding to requested audio source and return associated policy
         // mix if any. Calls getDeviceForInputSource().
-        audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource,
-                                                        AudioMix **policyMix = NULL);
+        sp<DeviceDescriptor> getDeviceAndMixForAttributes(const audio_attributes_t &attributes,
+                                                          AudioMix **policyMix = NULL);
 
         // Called by setDeviceConnectionState().
-        status_t setDeviceConnectionStateInt(audio_devices_t device,
-                                                          audio_policy_dev_state_t state,
-                                                          const char *device_address,
-                                                          const char *device_name);
+        status_t setDeviceConnectionStateInt(audio_devices_t deviceType,
+                                             audio_policy_dev_state_t state,
+                                             const char *device_address,
+                                             const char *device_name);
         void updateMono(audio_io_handle_t output) {
             AudioParameter param;
             param.addInt(String8(AudioParameter::keyMonoOutput), (int)mMasterMono);
diff --git a/services/oboeservice/AAudioServiceEndpoint.h b/services/oboeservice/AAudioServiceEndpoint.h
index 43b0a37..3616fa2 100644
--- a/services/oboeservice/AAudioServiceEndpoint.h
+++ b/services/oboeservice/AAudioServiceEndpoint.h
@@ -121,7 +121,7 @@
     mutable std::mutex       mLockStreams;
     std::vector<android::sp<AAudioServiceStreamBase>> mRegisteredStreams;
 
-    SimpleDoubleBuffer<Timestamp>  mAtomicTimestamp;
+    SimpleDoubleBuffer<Timestamp>  mAtomicEndpointTimestamp;
 
     android::AudioClient     mMmapClient;   // set in open, used in open and startStream
 
diff --git a/services/oboeservice/AAudioServiceEndpointShared.cpp b/services/oboeservice/AAudioServiceEndpointShared.cpp
index 2f1ec7e..0a415fd 100644
--- a/services/oboeservice/AAudioServiceEndpointShared.cpp
+++ b/services/oboeservice/AAudioServiceEndpointShared.cpp
@@ -181,8 +181,8 @@
 // Get timestamp that was written by the real-time service thread, eg. mixer.
 aaudio_result_t AAudioServiceEndpointShared::getFreeRunningPosition(int64_t *positionFrames,
                                                                   int64_t *timeNanos) {
-    if (mAtomicTimestamp.isValid()) {
-        Timestamp timestamp = mAtomicTimestamp.read();
+    if (mAtomicEndpointTimestamp.isValid()) {
+        Timestamp timestamp = mAtomicEndpointTimestamp.read();
         *positionFrames = timestamp.getPosition();
         *timeNanos = timestamp.getNanoseconds();
         return AAUDIO_OK;
diff --git a/services/oboeservice/AAudioServiceStreamBase.cpp b/services/oboeservice/AAudioServiceStreamBase.cpp
index defbb7b..b16b5dc 100644
--- a/services/oboeservice/AAudioServiceStreamBase.cpp
+++ b/services/oboeservice/AAudioServiceStreamBase.cpp
@@ -43,7 +43,7 @@
 AAudioServiceStreamBase::AAudioServiceStreamBase(AAudioService &audioService)
         : mUpMessageQueue(nullptr)
         , mTimestampThread("AATime")
-        , mAtomicTimestamp()
+        , mAtomicStreamTimestamp()
         , mAudioService(audioService) {
     mMmapClient.clientUid = -1;
     mMmapClient.clientPid = -1;
@@ -182,7 +182,7 @@
     setSuspended(false);
 
     // Start with fresh presentation timestamps.
-    mAtomicTimestamp.clear();
+    mAtomicStreamTimestamp.clear();
 
     mClientHandle = AUDIO_PORT_HANDLE_NONE;
     result = startDevice();
@@ -291,16 +291,20 @@
 }
 
 // implement Runnable, periodically send timestamps to client
+__attribute__((no_sanitize("integer")))
 void AAudioServiceStreamBase::run() {
     ALOGD("%s() %s entering >>>>>>>>>>>>>> TIMESTAMPS", __func__, getTypeText());
     TimestampScheduler timestampScheduler;
     timestampScheduler.setBurstPeriod(mFramesPerBurst, getSampleRate());
     timestampScheduler.start(AudioClock::getNanoseconds());
     int64_t nextTime = timestampScheduler.nextAbsoluteTime();
+    int32_t loopCount = 0;
     while(mThreadEnabled.load()) {
+        loopCount++;
         if (AudioClock::getNanoseconds() >= nextTime) {
             aaudio_result_t result = sendCurrentTimestamp();
             if (result != AAUDIO_OK) {
+                ALOGE("%s() timestamp thread got result = %d", __func__, result);
                 break;
             }
             nextTime = timestampScheduler.nextAbsoluteTime();
@@ -310,7 +314,8 @@
             AudioClock::sleepUntilNanoTime(nextTime);
         }
     }
-    ALOGD("%s() %s exiting <<<<<<<<<<<<<< TIMESTAMPS", __func__, getTypeText());
+    ALOGD("%s() %s exiting after %d loops <<<<<<<<<<<<<< TIMESTAMPS",
+          __func__, getTypeText(), loopCount);
 }
 
 void AAudioServiceStreamBase::disconnect() {
diff --git a/services/oboeservice/AAudioServiceStreamBase.h b/services/oboeservice/AAudioServiceStreamBase.h
index 7904b25..ffc768b 100644
--- a/services/oboeservice/AAudioServiceStreamBase.h
+++ b/services/oboeservice/AAudioServiceStreamBase.h
@@ -301,7 +301,7 @@
     // TODO rename mClientHandle to mPortHandle to be more consistent with AudioFlinger.
     audio_port_handle_t     mClientHandle = AUDIO_PORT_HANDLE_NONE;
 
-    SimpleDoubleBuffer<Timestamp>  mAtomicTimestamp;
+    SimpleDoubleBuffer<Timestamp>  mAtomicStreamTimestamp;
 
     android::AAudioService &mAudioService;
 
diff --git a/services/oboeservice/AAudioServiceStreamMMAP.cpp b/services/oboeservice/AAudioServiceStreamMMAP.cpp
index 9377945..837b080 100644
--- a/services/oboeservice/AAudioServiceStreamMMAP.cpp
+++ b/services/oboeservice/AAudioServiceStreamMMAP.cpp
@@ -162,7 +162,7 @@
     aaudio_result_t result = serviceEndpointMMAP->getFreeRunningPosition(positionFrames, timeNanos);
     if (result == AAUDIO_OK) {
         Timestamp timestamp(*positionFrames, *timeNanos);
-        mAtomicTimestamp.write(timestamp);
+        mAtomicStreamTimestamp.write(timestamp);
         *positionFrames = timestamp.getPosition();
         *timeNanos = timestamp.getNanoseconds();
     } else if (result != AAUDIO_ERROR_UNAVAILABLE) {
@@ -184,8 +184,8 @@
             static_cast<AAudioServiceEndpointMMAP *>(endpoint.get());
 
     // TODO Get presentation timestamp from the HAL
-    if (mAtomicTimestamp.isValid()) {
-        Timestamp timestamp = mAtomicTimestamp.read();
+    if (mAtomicStreamTimestamp.isValid()) {
+        Timestamp timestamp = mAtomicStreamTimestamp.read();
         *positionFrames = timestamp.getPosition();
         *timeNanos = timestamp.getNanoseconds() + serviceEndpointMMAP->getHardwareTimeOffsetNanos();
         return AAUDIO_OK;
diff --git a/services/oboeservice/AAudioServiceStreamShared.cpp b/services/oboeservice/AAudioServiceStreamShared.cpp
index d5450fe..14742dd 100644
--- a/services/oboeservice/AAudioServiceStreamShared.cpp
+++ b/services/oboeservice/AAudioServiceStreamShared.cpp
@@ -238,15 +238,15 @@
 }
 
 void AAudioServiceStreamShared::markTransferTime(Timestamp &timestamp) {
-    mAtomicTimestamp.write(timestamp);
+    mAtomicStreamTimestamp.write(timestamp);
 }
 
 // Get timestamp that was written by mixer or distributor.
 aaudio_result_t AAudioServiceStreamShared::getFreeRunningPosition(int64_t *positionFrames,
                                                                   int64_t *timeNanos) {
     // TODO Get presentation timestamp from the HAL
-    if (mAtomicTimestamp.isValid()) {
-        Timestamp timestamp = mAtomicTimestamp.read();
+    if (mAtomicStreamTimestamp.isValid()) {
+        Timestamp timestamp = mAtomicStreamTimestamp.read();
         *positionFrames = timestamp.getPosition();
         *timeNanos = timestamp.getNanoseconds();
         return AAUDIO_OK;
diff --git a/services/soundtrigger/SoundTriggerHalHidl.cpp b/services/soundtrigger/SoundTriggerHalHidl.cpp
index 1d37a8e..68d54c7 100644
--- a/services/soundtrigger/SoundTriggerHalHidl.cpp
+++ b/services/soundtrigger/SoundTriggerHalHidl.cpp
@@ -168,18 +168,23 @@
     int ret;
     SoundModelHandle halHandle;
     sp<V2_1_ISoundTriggerHw> soundtrigger_2_1 = toService2_1(soundtrigger);
+    sp<V2_2_ISoundTriggerHw> soundtrigger_2_2 = toService2_2(soundtrigger);
     if (sound_model->type == SOUND_MODEL_TYPE_KEYPHRASE) {
-        if (!soundtrigger_2_1) {
-            ISoundTriggerHw::PhraseSoundModel halSoundModel;
-            convertPhraseSoundModelToHal(&halSoundModel, sound_model);
-            AutoMutex lock(mHalLock);
-            hidlReturn = soundtrigger->loadPhraseSoundModel(
-                    halSoundModel,
-                    this, modelId, [&](int32_t retval, auto res) {
-                        ret = retval;
-                        halHandle = res;
-                    });
-        } else {
+        if (soundtrigger_2_2) {
+            V2_2_ISoundTriggerHw::PhraseSoundModel halSoundModel;
+            auto result = convertPhraseSoundModelToHal(&halSoundModel, sound_model);
+            if (result.first) {
+                AutoMutex lock(mHalLock);
+                hidlReturn = soundtrigger_2_2->loadPhraseSoundModel_2_1(
+                        halSoundModel,
+                        this, modelId, [&](int32_t retval, auto res) {
+                            ret = retval;
+                            halHandle = res;
+                        });
+            } else {
+                return NO_MEMORY;
+            }
+        } else if (soundtrigger_2_1) {
             V2_1_ISoundTriggerHw::PhraseSoundModel halSoundModel;
             auto result = convertPhraseSoundModelToHal(&halSoundModel, sound_model);
             if (result.first) {
@@ -193,18 +198,32 @@
             } else {
                 return NO_MEMORY;
             }
-        }
-    } else {
-        if (!soundtrigger_2_1) {
-            ISoundTriggerHw::SoundModel halSoundModel;
-            convertSoundModelToHal(&halSoundModel, sound_model);
+        } else {
+            ISoundTriggerHw::PhraseSoundModel halSoundModel;
+            convertPhraseSoundModelToHal(&halSoundModel, sound_model);
             AutoMutex lock(mHalLock);
-            hidlReturn = soundtrigger->loadSoundModel(halSoundModel,
+            hidlReturn = soundtrigger->loadPhraseSoundModel(
+                    halSoundModel,
                     this, modelId, [&](int32_t retval, auto res) {
                         ret = retval;
                         halHandle = res;
                     });
-        } else {
+        }
+    } else {
+        if (soundtrigger_2_2) {
+            V2_2_ISoundTriggerHw::SoundModel halSoundModel;
+            auto result = convertSoundModelToHal(&halSoundModel, sound_model);
+            if (result.first) {
+                AutoMutex lock(mHalLock);
+                hidlReturn = soundtrigger_2_2->loadSoundModel_2_1(halSoundModel,
+                        this, modelId, [&](int32_t retval, auto res) {
+                            ret = retval;
+                            halHandle = res;
+                        });
+            } else {
+                return NO_MEMORY;
+            }
+        } else if (soundtrigger_2_1) {
             V2_1_ISoundTriggerHw::SoundModel halSoundModel;
             auto result = convertSoundModelToHal(&halSoundModel, sound_model);
             if (result.first) {
@@ -217,6 +236,15 @@
             } else {
                 return NO_MEMORY;
             }
+        } else {
+            ISoundTriggerHw::SoundModel halSoundModel;
+            convertSoundModelToHal(&halSoundModel, sound_model);
+            AutoMutex lock(mHalLock);
+            hidlReturn = soundtrigger->loadSoundModel(halSoundModel,
+                    this, modelId, [&](int32_t retval, auto res) {
+                        ret = retval;
+                        halHandle = res;
+                    });
         }
     }
 
@@ -282,16 +310,20 @@
     model->mRecognitionCookie = cookie;
 
     sp<V2_1_ISoundTriggerHw> soundtrigger_2_1 = toService2_1(soundtrigger);
+    sp<V2_2_ISoundTriggerHw> soundtrigger_2_2 = toService2_2(soundtrigger);
     Return<int32_t> hidlReturn(0);
 
-    if (!soundtrigger_2_1) {
-        ISoundTriggerHw::RecognitionConfig halConfig;
-        convertRecognitionConfigToHal(&halConfig, config);
-        {
+    if (soundtrigger_2_2) {
+        V2_2_ISoundTriggerHw::RecognitionConfig halConfig;
+        auto result = convertRecognitionConfigToHal(&halConfig, config);
+        if (result.first) {
             AutoMutex lock(mHalLock);
-            hidlReturn = soundtrigger->startRecognition(model->mHalHandle, halConfig, this, handle);
+            hidlReturn = soundtrigger_2_2->startRecognition_2_1(
+                    model->mHalHandle, halConfig, this, handle);
+        } else {
+            return NO_MEMORY;
         }
-    } else {
+    } else if (soundtrigger_2_1) {
         V2_1_ISoundTriggerHw::RecognitionConfig halConfig;
         auto result = convertRecognitionConfigToHal(&halConfig, config);
         if (result.first) {
@@ -301,6 +333,13 @@
         } else {
             return NO_MEMORY;
         }
+    } else {
+        ISoundTriggerHw::RecognitionConfig halConfig;
+        convertRecognitionConfigToHal(&halConfig, config);
+        {
+            AutoMutex lock(mHalLock);
+            hidlReturn = soundtrigger->startRecognition(model->mHalHandle, halConfig, this, handle);
+        }
     }
 
     if (!hidlReturn.isOk()) {