diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index cb4bcfc..e260fd8 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -1238,18 +1238,18 @@
 
 status_t AudioSystem::startAudioSource(const struct audio_port_config *source,
                                        const audio_attributes_t *attributes,
-                                       audio_patch_handle_t *handle)
+                                       audio_port_handle_t *portId)
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
     if (aps == 0) return PERMISSION_DENIED;
-    return aps->startAudioSource(source, attributes, handle);
+    return aps->startAudioSource(source, attributes, portId);
 }
 
-status_t AudioSystem::stopAudioSource(audio_patch_handle_t handle)
+status_t AudioSystem::stopAudioSource(audio_port_handle_t portId)
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
     if (aps == 0) return PERMISSION_DENIED;
-    return aps->stopAudioSource(handle);
+    return aps->stopAudioSource(portId);
 }
 
 status_t AudioSystem::setMasterMono(bool mono)
diff --git a/media/libaudioclient/IAudioPolicyService.cpp b/media/libaudioclient/IAudioPolicyService.cpp
index e229f4c..73a8b74 100644
--- a/media/libaudioclient/IAudioPolicyService.cpp
+++ b/media/libaudioclient/IAudioPolicyService.cpp
@@ -740,11 +740,11 @@
 
     virtual status_t startAudioSource(const struct audio_port_config *source,
                                       const audio_attributes_t *attributes,
-                                      audio_patch_handle_t *handle)
+                                      audio_port_handle_t *portId)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
-        if (source == NULL || attributes == NULL || handle == NULL) {
+        if (source == NULL || attributes == NULL || portId == NULL) {
             return BAD_VALUE;
         }
         data.write(source, sizeof(struct audio_port_config));
@@ -757,15 +757,15 @@
         if (status != NO_ERROR) {
             return status;
         }
-        *handle = (audio_patch_handle_t)reply.readInt32();
+        *portId = (audio_port_handle_t)reply.readInt32();
         return status;
     }
 
-    virtual status_t stopAudioSource(audio_patch_handle_t handle)
+    virtual status_t stopAudioSource(audio_port_handle_t portId)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
-        data.writeInt32(handle);
+        data.writeInt32(portId);
         status_t status = remote()->transact(STOP_AUDIO_SOURCE, data, &reply);
         if (status != NO_ERROR) {
             return status;
@@ -1472,17 +1472,17 @@
             audio_attributes_t attributes = {};
             data.read(&attributes, sizeof(audio_attributes_t));
             sanetizeAudioAttributes(&attributes);
-            audio_patch_handle_t handle = AUDIO_PATCH_HANDLE_NONE;
-            status_t status = startAudioSource(&source, &attributes, &handle);
+            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
+            status_t status = startAudioSource(&source, &attributes, &portId);
             reply->writeInt32(status);
-            reply->writeInt32(handle);
+            reply->writeInt32(portId);
             return NO_ERROR;
         } break;
 
         case STOP_AUDIO_SOURCE: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            audio_patch_handle_t handle = (audio_patch_handle_t) data.readInt32();
-            status_t status = stopAudioSource(handle);
+            audio_port_handle_t portId = (audio_port_handle_t) data.readInt32();
+            status_t status = stopAudioSource(portId);
             reply->writeInt32(status);
             return NO_ERROR;
         } break;
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index 10d6e92..adfee8b 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -322,9 +322,9 @@
     static status_t registerPolicyMixes(const Vector<AudioMix>& mixes, bool registration);
 
     static status_t startAudioSource(const struct audio_port_config *source,
-                                      const audio_attributes_t *attributes,
-                                      audio_patch_handle_t *handle);
-    static status_t stopAudioSource(audio_patch_handle_t handle);
+                                     const audio_attributes_t *attributes,
+                                     audio_port_handle_t *portId);
+    static status_t stopAudioSource(audio_port_handle_t portId);
 
     static status_t setMasterMono(bool mono);
     static status_t getMasterMono(bool *mono);
diff --git a/media/libaudioclient/include/media/IAudioPolicyService.h b/media/libaudioclient/include/media/IAudioPolicyService.h
index 6c017a3..cdbb876 100644
--- a/media/libaudioclient/include/media/IAudioPolicyService.h
+++ b/media/libaudioclient/include/media/IAudioPolicyService.h
@@ -153,8 +153,8 @@
 
     virtual status_t startAudioSource(const struct audio_port_config *source,
                                       const audio_attributes_t *attributes,
-                                      audio_patch_handle_t *handle) = 0;
-    virtual status_t stopAudioSource(audio_patch_handle_t handle) = 0;
+                                      audio_port_handle_t *portId) = 0;
+    virtual status_t stopAudioSource(audio_port_handle_t portId) = 0;
 
     virtual status_t setMasterMono(bool mono) = 0;
     virtual status_t getMasterMono(bool *mono) = 0;
diff --git a/media/libaudioprocessing/AudioMixer.cpp b/media/libaudioprocessing/AudioMixer.cpp
index f6f817a..f3ea826 100644
--- a/media/libaudioprocessing/AudioMixer.cpp
+++ b/media/libaudioprocessing/AudioMixer.cpp
@@ -875,7 +875,7 @@
                 t->hook = Track::getTrackHook(TRACKTYPE_RESAMPLE, t->mMixerChannelCount,
                         t->mMixerInFormat, t->mMixerFormat);
                 ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
-                        "Track %d needs downmix + resample", i);
+                        "Track %d needs downmix + resample", name);
             } else {
                 if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
                     t->hook = Track::getTrackHook(
@@ -890,7 +890,7 @@
                     t->hook = Track::getTrackHook(TRACKTYPE_NORESAMPLE, t->mMixerChannelCount,
                             t->mMixerInFormat, t->mMixerFormat);
                     ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
-                            "Track %d needs downmix", i);
+                            "Track %d needs downmix", name);
                 }
             }
         }
diff --git a/media/libstagefright/CameraSource.cpp b/media/libstagefright/CameraSource.cpp
index db37021..41f5db0 100644
--- a/media/libstagefright/CameraSource.cpp
+++ b/media/libstagefright/CameraSource.cpp
@@ -217,6 +217,7 @@
       mNumFramesReceived(0),
       mLastFrameTimestampUs(0),
       mStarted(false),
+      mEos(false),
       mNumFramesEncoded(0),
       mTimeBetweenFrameCaptureUs(0),
       mFirstFrameTimeUs(0),
@@ -880,6 +881,7 @@
     {
         Mutex::Autolock autoLock(mLock);
         mStarted = false;
+        mEos = false;
         mStopSystemTimeUs = -1;
         mFrameAvailableCondition.signal();
 
@@ -1075,7 +1077,7 @@
 
     {
         Mutex::Autolock autoLock(mLock);
-        while (mStarted && mFramesReceived.empty()) {
+        while (mStarted && !mEos && mFramesReceived.empty()) {
             if (NO_ERROR !=
                 mFrameAvailableCondition.waitRelative(mLock,
                     mTimeBetweenFrameCaptureUs * 1000LL + CAMERA_SOURCE_TIMEOUT_NS)) {
@@ -1091,6 +1093,9 @@
         if (!mStarted) {
             return OK;
         }
+        if (mFramesReceived.empty()) {
+            return ERROR_END_OF_STREAM;
+        }
         frame = *mFramesReceived.begin();
         mFramesReceived.erase(mFramesReceived.begin());
 
@@ -1129,6 +1134,8 @@
     if (mStopSystemTimeUs != -1 && timestampUs >= mStopSystemTimeUs) {
         ALOGV("Drop Camera frame at %lld  stop time: %lld us",
                 (long long)timestampUs, (long long)mStopSystemTimeUs);
+        mEos = true;
+        mFrameAvailableCondition.signal();
         return true;
     }
 
diff --git a/media/libstagefright/VideoFrameScheduler.cpp b/media/libstagefright/VideoFrameScheduler.cpp
index 6819bba..9020fc1 100644
--- a/media/libstagefright/VideoFrameScheduler.cpp
+++ b/media/libstagefright/VideoFrameScheduler.cpp
@@ -475,7 +475,16 @@
                 nextVsyncTime += mVsyncPeriod;
                 if (vsyncsForLastFrame < ULONG_MAX)
                     ++vsyncsForLastFrame;
+            } else if (mTimeCorrection < -correctionLimit * 2
+                    || mTimeCorrection > correctionLimit * 2) {
+                ALOGW("correction beyond limit: %lld vs %lld (vsyncs for last frame: %zu, min: %zu)"
+                        " restarting. render=%lld",
+                        (long long)mTimeCorrection, (long long)correctionLimit,
+                        vsyncsForLastFrame, minVsyncsPerFrame, (long long)origRenderTime);
+                restart();
+                return origRenderTime;
             }
+
             ATRACE_INT("FRAME_VSYNCS", vsyncsForLastFrame);
         }
         mLastVsyncTime = nextVsyncTime;
diff --git a/media/libstagefright/include/media/stagefright/CameraSource.h b/media/libstagefright/include/media/stagefright/CameraSource.h
index 475976b..3037b72 100644
--- a/media/libstagefright/include/media/stagefright/CameraSource.h
+++ b/media/libstagefright/include/media/stagefright/CameraSource.h
@@ -204,6 +204,7 @@
     int32_t mNumFramesReceived;
     int64_t mLastFrameTimestampUs;
     bool mStarted;
+    bool mEos;
     int32_t mNumFramesEncoded;
 
     // Time between capture of two frames.
diff --git a/services/audioflinger/Configuration.h b/services/audioflinger/Configuration.h
index 64bb426..34cd821 100644
--- a/services/audioflinger/Configuration.h
+++ b/services/audioflinger/Configuration.h
@@ -27,7 +27,7 @@
 //#define AUDIO_WATCHDOG
 
 // uncomment to display CPU load adjusted for CPU frequency
-#define CPU_FREQUENCY_STATISTICS
+//define CPU_FREQUENCY_STATISTICS
 
 // uncomment to enable fast threads to take performance samples for later statistical analysis
 #define FAST_THREAD_STATISTICS
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index ad35264..a42d6b3 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -32,7 +32,7 @@
 #include <utils/Trace.h>
 #include <system/audio.h>
 #ifdef FAST_THREAD_STATISTICS
-#include <cpustats/CentralTendencyStatistics.h>
+#include <audio_utils/Statistics.h>
 #ifdef CPU_FREQUENCY_STATISTICS
 #include <cpustats/ThreadCpuUsage.h>
 #endif
diff --git a/services/audioflinger/FastMixerDumpState.cpp b/services/audioflinger/FastMixerDumpState.cpp
index c0f9f0f..ffdc117 100644
--- a/services/audioflinger/FastMixerDumpState.cpp
+++ b/services/audioflinger/FastMixerDumpState.cpp
@@ -19,7 +19,7 @@
 
 #include "Configuration.h"
 #ifdef FAST_THREAD_STATISTICS
-#include <cpustats/CentralTendencyStatistics.h>
+#include <audio_utils/Statistics.h>
 #ifdef CPU_FREQUENCY_STATISTICS
 #include <cpustats/ThreadCpuUsage.h>
 #endif
@@ -92,9 +92,9 @@
     }
     // statistics for monotonic (wall clock) time, thread raw CPU load in time, CPU clock frequency,
     // and adjusted CPU load in MHz normalized for CPU clock frequency
-    CentralTendencyStatistics wall, loadNs;
+    Statistics<double> wall, loadNs;
 #ifdef CPU_FREQUENCY_STATISTICS
-    CentralTendencyStatistics kHz, loadMHz;
+    Statistics<double> kHz, loadMHz;
     uint32_t previousCpukHz = 0;
 #endif
     // Assuming a normal distribution for cycle times, three standard deviations on either side of
@@ -109,18 +109,18 @@
         if (tail != NULL) {
             tail[j] = wallNs;
         }
-        wall.sample(wallNs);
+        wall.add(wallNs);
         uint32_t sampleLoadNs = mLoadNs[i];
-        loadNs.sample(sampleLoadNs);
+        loadNs.add(sampleLoadNs);
 #ifdef CPU_FREQUENCY_STATISTICS
         uint32_t sampleCpukHz = mCpukHz[i];
         // skip bad kHz samples
         if ((sampleCpukHz & ~0xF) != 0) {
-            kHz.sample(sampleCpukHz >> 4);
+            kHz.add(sampleCpukHz >> 4);
             if (sampleCpukHz == previousCpukHz) {
                 double megacycles = (double) sampleLoadNs * (double) (sampleCpukHz >> 4) * 1e-12;
                 double adjMHz = megacycles / mixPeriodSec;  // _not_ wallNs * 1e9
-                loadMHz.sample(adjMHz);
+                loadMHz.add(adjMHz);
             }
         }
         previousCpukHz = sampleCpukHz;
@@ -128,42 +128,42 @@
     }
     if (n) {
         dprintf(fd, "  Simple moving statistics over last %.1f seconds:\n",
-                    wall.n() * mixPeriodSec);
+                    wall.getN() * mixPeriodSec);
         dprintf(fd, "    wall clock time in ms per mix cycle:\n"
                     "      mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
-                    wall.mean()*1e-6, wall.minimum()*1e-6, wall.maximum()*1e-6,
-                    wall.stddev()*1e-6);
+                    wall.getMean()*1e-6, wall.getMin()*1e-6, wall.getMax()*1e-6,
+                    wall.getStdDev()*1e-6);
         dprintf(fd, "    raw CPU load in us per mix cycle:\n"
                     "      mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
-                    loadNs.mean()*1e-3, loadNs.minimum()*1e-3, loadNs.maximum()*1e-3,
-                    loadNs.stddev()*1e-3);
+                    loadNs.getMean()*1e-3, loadNs.getMin()*1e-3, loadNs.getMax()*1e-3,
+                    loadNs.getStdDev()*1e-3);
     } else {
         dprintf(fd, "  No FastMixer statistics available currently\n");
     }
 #ifdef CPU_FREQUENCY_STATISTICS
     dprintf(fd, "  CPU clock frequency in MHz:\n"
                 "    mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
-                kHz.mean()*1e-3, kHz.minimum()*1e-3, kHz.maximum()*1e-3, kHz.stddev()*1e-3);
+                kHz.getMean()*1e-3, kHz.getMin()*1e-3, kHz.getMax()*1e-3, kHz.getStdDev()*1e-3);
     dprintf(fd, "  adjusted CPU load in MHz (i.e. normalized for CPU clock frequency):\n"
                 "    mean=%.1f min=%.1f max=%.1f stddev=%.1f\n",
-                loadMHz.mean(), loadMHz.minimum(), loadMHz.maximum(), loadMHz.stddev());
+                loadMHz.getMean(), loadMHz.getMin(), loadMHz.getMax(), loadMHz.getStdDev());
 #endif
     if (tail != NULL) {
         qsort(tail, n, sizeof(uint32_t), compare_uint32_t);
         // assume same number of tail samples on each side, left and right
         uint32_t count = n / kTailDenominator;
-        CentralTendencyStatistics left, right;
+        Statistics<double> left, right;
         for (uint32_t i = 0; i < count; ++i) {
-            left.sample(tail[i]);
-            right.sample(tail[n - (i + 1)]);
+            left.add(tail[i]);
+            right.add(tail[n - (i + 1)]);
         }
         dprintf(fd, "  Distribution of mix cycle times in ms for the tails "
                     "(> ~3 stddev outliers):\n"
                     "    left tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n"
                     "    right tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
-                    left.mean()*1e-6, left.minimum()*1e-6, left.maximum()*1e-6, left.stddev()*1e-6,
-                    right.mean()*1e-6, right.minimum()*1e-6, right.maximum()*1e-6,
-                    right.stddev()*1e-6);
+                    left.getMean()*1e-6, left.getMin()*1e-6, left.getMax()*1e-6, left.getStdDev()*1e-6,
+                    right.getMean()*1e-6, right.getMin()*1e-6, right.getMax()*1e-6,
+                    right.getStdDev()*1e-6);
         delete[] tail;
     }
 #endif
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index d55da1b..6c7179e 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -73,7 +73,7 @@
 #endif
 
 #ifdef DEBUG_CPU_USAGE
-#include <cpustats/CentralTendencyStatistics.h>
+#include <audio_utils/Statistics.h>
 #include <cpustats/ThreadCpuUsage.h>
 #endif
 
@@ -335,9 +335,9 @@
 #ifdef DEBUG_CPU_USAGE
 private:
     ThreadCpuUsage mCpuUsage;           // instantaneous thread CPU usage in wall clock ns
-    CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns
+    Statistics<double> mWcStats;        // statistics on thread CPU usage in wall clock ns
 
-    CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles
+    Statistics<double> mHzStats;        // statistics on thread CPU usage in cycles
 
     int mCpuNum;                        // thread's current CPU number
     int mCpukHz;                        // frequency of thread's current CPU in kHz
@@ -363,7 +363,7 @@
 
     // record sample for wall clock statistics
     if (valid) {
-        mWcStats.sample(wcNs);
+        mWcStats.add(wcNs);
     }
 
     // get the current CPU number
@@ -382,26 +382,26 @@
 
     // if no change in CPU number or frequency, then record sample for cycle statistics
     if (valid && mCpukHz > 0) {
-        double cycles = wcNs * cpukHz * 0.000001;
-        mHzStats.sample(cycles);
+        const double cycles = wcNs * cpukHz * 0.000001;
+        mHzStats.add(cycles);
     }
 
-    unsigned n = mWcStats.n();
+    const unsigned n = mWcStats.getN();
     // mCpuUsage.elapsed() is expensive, so don't call it every loop
     if ((n & 127) == 1) {
-        long long elapsed = mCpuUsage.elapsed();
+        const long long elapsed = mCpuUsage.elapsed();
         if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
-            double perLoop = elapsed / (double) n;
-            double perLoop100 = perLoop * 0.01;
-            double perLoop1k = perLoop * 0.001;
-            double mean = mWcStats.mean();
-            double stddev = mWcStats.stddev();
-            double minimum = mWcStats.minimum();
-            double maximum = mWcStats.maximum();
-            double meanCycles = mHzStats.mean();
-            double stddevCycles = mHzStats.stddev();
-            double minCycles = mHzStats.minimum();
-            double maxCycles = mHzStats.maximum();
+            const double perLoop = elapsed / (double) n;
+            const double perLoop100 = perLoop * 0.01;
+            const double perLoop1k = perLoop * 0.001;
+            const double mean = mWcStats.getMean();
+            const double stddev = mWcStats.getStdDev();
+            const double minimum = mWcStats.getMin();
+            const double maximum = mWcStats.getMax();
+            const double meanCycles = mHzStats.getMean();
+            const double stddevCycles = mHzStats.getStdDev();
+            const double minCycles = mHzStats.getMin();
+            const double maxCycles = mHzStats.getMax();
             mCpuUsage.resetElapsed();
             mWcStats.reset();
             mHzStats.reset();
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index c9e99d5..d4c49d9 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -226,9 +226,9 @@
 
     virtual status_t startAudioSource(const struct audio_port_config *source,
                                       const audio_attributes_t *attributes,
-                                      audio_patch_handle_t *handle,
+                                      audio_port_handle_t *portId,
                                       uid_t uid) = 0;
-    virtual status_t stopAudioSource(audio_patch_handle_t handle) = 0;
+    virtual status_t stopAudioSource(audio_port_handle_t portId) = 0;
 
     virtual status_t setMasterMono(bool mono) = 0;
     virtual status_t getMasterMono(bool *mono) = 0;
diff --git a/services/audiopolicy/common/managerdefinitions/Android.mk b/services/audiopolicy/common/managerdefinitions/Android.mk
index 09dbb32..9b8f095 100644
--- a/services/audiopolicy/common/managerdefinitions/Android.mk
+++ b/services/audiopolicy/common/managerdefinitions/Android.mk
@@ -18,7 +18,6 @@
     src/EffectDescriptor.cpp \
     src/SoundTriggerSession.cpp \
     src/SessionRoute.cpp \
-    src/AudioSourceDescriptor.cpp \
     src/VolumeCurve.cpp \
     src/TypeConverter.cpp \
     src/AudioSession.cpp \
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
index e6112bf..ff0201a 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -24,7 +24,6 @@
 #include <RoutingStrategy.h>
 #include "AudioIODescriptorInterface.h"
 #include "AudioPort.h"
-#include "AudioSourceDescriptor.h"
 #include "ClientDescriptor.h"
 
 namespace android {
@@ -159,7 +158,7 @@
 class HwAudioOutputDescriptor: public AudioOutputDescriptor
 {
 public:
-    HwAudioOutputDescriptor(const sp<AudioSourceDescriptor>& source,
+    HwAudioOutputDescriptor(const sp<SourceClientDescriptor>& source,
                             AudioPolicyClientInterface *clientInterface);
     virtual ~HwAudioOutputDescriptor() {}
 
@@ -176,7 +175,7 @@
                            const struct audio_port_config *srcConfig = NULL) const;
     virtual void toAudioPort(struct audio_port *port) const;
 
-    const sp<AudioSourceDescriptor> mSource;
+    const sp<SourceClientDescriptor> mSource;
 
 };
 
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h b/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h
index 31c75a4..a1ee708 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h
@@ -16,6 +16,8 @@
 
 #pragma once
 
+#include <vector>
+
 #include <system/audio.h>
 #include <utils/RefBase.h>
 #include <utils/SortedVector.h>
@@ -139,6 +141,16 @@
                                     audio_port_type_t portType,
                                     audio_port_role_t portRole) const;
     void clearProfiles();
+    // Assuming that this profile vector contains input profiles,
+    // find the best matching config from 'outputProfiles', according to
+    // the given preferences for audio formats and channel masks.
+    // Note: std::vectors are used because specialized containers for formats
+    //       and channels can be sorted and use their own ordering.
+    status_t findBestMatchingOutputConfig(const AudioProfileVector& outputProfiles,
+            const std::vector<audio_format_t>& preferredFormats, // order: most pref -> least pref
+            const std::vector<audio_channel_mask_t>& preferredOutputChannels,
+            bool preferHigherSamplingRates,
+            audio_config_base *bestOutputConfig) const;
 
     sp<AudioProfile> getFirstValidProfile() const;
     sp<AudioProfile> getFirstValidProfileFor(audio_format_t format) const;
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioSourceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioSourceDescriptor.h
deleted file mode 100644
index 0d90f42..0000000
--- a/services/audiopolicy/common/managerdefinitions/include/AudioSourceDescriptor.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- * Copyright (C) 2015 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#pragma once
-
-#include <system/audio.h>
-#include <utils/Errors.h>
-#include <utils/KeyedVector.h>
-#include <utils/RefBase.h>
-#include <RoutingStrategy.h>
-#include <AudioPatch.h>
-
-namespace android {
-
-class SwAudioOutputDescriptor;
-class HwAudioOutputDescriptor;
-class DeviceDescriptor;
-
-class AudioSourceDescriptor: public RefBase
-{
-public:
-    AudioSourceDescriptor(const sp<DeviceDescriptor> device, const audio_attributes_t *attributes,
-                          uid_t uid) :
-        mDevice(device), mAttributes(*attributes), mUid(uid) {}
-    virtual ~AudioSourceDescriptor() {}
-
-    audio_patch_handle_t getHandle() const { return mPatchDesc->mHandle; }
-
-    status_t    dump(int fd);
-
-    const sp<DeviceDescriptor> mDevice;
-    const audio_attributes_t mAttributes;
-    uid_t mUid;
-    sp<AudioPatch> mPatchDesc;
-    wp<SwAudioOutputDescriptor> mSwOutput;
-    wp<HwAudioOutputDescriptor> mHwOutput;
-};
-
-class AudioSourceCollection :
-        public DefaultKeyedVector< audio_patch_handle_t, sp<AudioSourceDescriptor> >
-{
-public:
-    status_t dump(int fd) const;
-};
-
-} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
index 221c2e9..9efe57f 100644
--- a/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
@@ -23,11 +23,17 @@
 
 #include <system/audio.h>
 #include <utils/Errors.h>
+#include <utils/KeyedVector.h>
 #include <utils/RefBase.h>
 #include <utils/String8.h>
+#include "AudioPatch.h"
 
 namespace android {
 
+class DeviceDescriptor;
+class HwAudioOutputDescriptor;
+class SwAudioOutputDescriptor;
+
 class ClientDescriptor: public RefBase
 {
 public:
@@ -58,6 +64,10 @@
     const audio_config_base_t mConfig;
     const audio_port_handle_t mPreferredDeviceId;  // selected input device port ID
           bool mActive;
+
+protected:
+    // FIXME: use until other descriptor classes have a dump to String8 method
+    int mDumpFd;
 };
 
 class TrackClientDescriptor: public ClientDescriptor
@@ -104,6 +114,38 @@
     const audio_input_flags_t mFlags;
 };
 
+class SourceClientDescriptor: public TrackClientDescriptor
+{
+public:
+    SourceClientDescriptor(audio_port_handle_t portId, uid_t uid, audio_attributes_t attributes,
+                           const sp<AudioPatch>& patchDesc, const sp<DeviceDescriptor>& srcDevice,
+                           audio_stream_type_t stream);
+    ~SourceClientDescriptor() override = default;
+
+    sp<AudioPatch> patchDesc() const { return mPatchDesc; }
+    sp<DeviceDescriptor> srcDevice() const { return mSrcDevice; };
+    wp<SwAudioOutputDescriptor> swOutput() const { return mSwOutput; }
+    void setSwOutput(const sp<SwAudioOutputDescriptor>& swOutput);
+    wp<HwAudioOutputDescriptor> hwOutput() const { return mHwOutput; }
+    void setHwOutput(const sp<HwAudioOutputDescriptor>& hwOutput);
+
+    using ClientDescriptor::dump;
+    status_t dump(String8& dst, int spaces, int index) override;
+
+ private:
+    const sp<AudioPatch> mPatchDesc;
+    const sp<DeviceDescriptor> mSrcDevice;
+    wp<SwAudioOutputDescriptor> mSwOutput;
+    wp<HwAudioOutputDescriptor> mHwOutput;
+};
+
+class SourceClientCollection :
+    public DefaultKeyedVector< audio_port_handle_t, sp<SourceClientDescriptor> >
+{
+public:
+    status_t dump(int fd) const;
+};
+
 typedef std::vector< sp<TrackClientDescriptor> > TrackClientVector;
 typedef std::map< audio_port_handle_t, sp<TrackClientDescriptor> > TrackClientMap;
 typedef std::vector< sp<RecordClientDescriptor> > RecordClientVector;
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 3dfbe1b..39fce4d 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -558,9 +558,9 @@
 }
 
 // HwAudioOutputDescriptor implementation
-HwAudioOutputDescriptor::HwAudioOutputDescriptor(const sp<AudioSourceDescriptor>& source,
+HwAudioOutputDescriptor::HwAudioOutputDescriptor(const sp<SourceClientDescriptor>& source,
                                                  AudioPolicyClientInterface *clientInterface)
-    : AudioOutputDescriptor(source->mDevice, clientInterface),
+    : AudioOutputDescriptor(source->srcDevice(), clientInterface),
       mSource(source)
 {
 }
@@ -576,7 +576,7 @@
     snprintf(buffer, SIZE, "Source:\n");
     result.append(buffer);
     write(fd, result.string(), result.size());
-    mSource->dump(fd);
+    mSource->dump(fd, 0, 0);
 
     return NO_ERROR;
 }
@@ -590,13 +590,13 @@
                                                  struct audio_port_config *dstConfig,
                                                  const struct audio_port_config *srcConfig) const
 {
-    mSource->mDevice->toAudioPortConfig(dstConfig, srcConfig);
+    mSource->srcDevice()->toAudioPortConfig(dstConfig, srcConfig);
 }
 
 void HwAudioOutputDescriptor::toAudioPort(
                                                     struct audio_port *port) const
 {
-    mSource->mDevice->toAudioPort(port);
+    mSource->srcDevice()->toAudioPort(port);
 }
 
 
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp
index b76b609..d04beec 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp
@@ -14,6 +14,8 @@
  * limitations under the License.
  */
 
+#include <algorithm>
+#include <set>
 #include <string>
 
 #define LOG_TAG "APM::AudioProfile"
@@ -406,6 +408,76 @@
     }
 }
 
+// Returns an intersection between two possibly unsorted vectors and the contents of 'order'.
+// The result is ordered according to 'order'.
+template<typename T, typename Order>
+std::vector<typename T::value_type> intersectFilterAndOrder(
+        const T& input1, const T& input2, const Order& order)
+{
+    std::set<typename T::value_type> set1{input1.begin(), input1.end()};
+    std::set<typename T::value_type> set2{input2.begin(), input2.end()};
+    std::set<typename T::value_type> common;
+    std::set_intersection(set1.begin(), set1.end(), set2.begin(), set2.end(),
+            std::inserter(common, common.begin()));
+    std::vector<typename T::value_type> result;
+    for (const auto& e : order) {
+        if (common.find(e) != common.end()) result.push_back(e);
+    }
+    return result;
+}
+
+// Intersect two possibly unsorted vectors, return common elements according to 'comp' ordering.
+// 'comp' is a comparator function.
+template<typename T, typename Compare>
+std::vector<typename T::value_type> intersectAndOrder(
+        const T& input1, const T& input2, Compare comp)
+{
+    std::set<typename T::value_type, Compare> set1{input1.begin(), input1.end(), comp};
+    std::set<typename T::value_type, Compare> set2{input2.begin(), input2.end(), comp};
+    std::vector<typename T::value_type> result;
+    std::set_intersection(set1.begin(), set1.end(), set2.begin(), set2.end(),
+            std::back_inserter(result), comp);
+    return result;
+}
+
+status_t AudioProfileVector::findBestMatchingOutputConfig(const AudioProfileVector& outputProfiles,
+            const std::vector<audio_format_t>& preferredFormats,
+            const std::vector<audio_channel_mask_t>& preferredOutputChannels,
+            bool preferHigherSamplingRates,
+            audio_config_base *bestOutputConfig) const
+{
+    auto formats = intersectFilterAndOrder(getSupportedFormats(),
+            outputProfiles.getSupportedFormats(), preferredFormats);
+    // Pick the best compatible profile.
+    for (const auto& f : formats) {
+        sp<AudioProfile> inputProfile = getFirstValidProfileFor(f);
+        sp<AudioProfile> outputProfile = outputProfiles.getFirstValidProfileFor(f);
+        if (inputProfile == nullptr || outputProfile == nullptr) {
+            continue;
+        }
+        auto channels = intersectFilterAndOrder(inputProfile->getChannels().asOutMask(),
+                outputProfile->getChannels(), preferredOutputChannels);
+        if (channels.empty()) {
+            continue;
+        }
+        auto sampleRates = preferHigherSamplingRates ?
+                intersectAndOrder(inputProfile->getSampleRates(), outputProfile->getSampleRates(),
+                        std::greater<typename SampleRateVector::value_type>()) :
+                intersectAndOrder(inputProfile->getSampleRates(), outputProfile->getSampleRates(),
+                        std::less<typename SampleRateVector::value_type>());
+        if (sampleRates.empty()) {
+            continue;
+        }
+        ALOGD("%s() found channel mask %#x and sample rate %d for format %#x.",
+                __func__, *channels.begin(), *sampleRates.begin(), f);
+        bestOutputConfig->format = f;
+        bestOutputConfig->sample_rate = *sampleRates.begin();
+        bestOutputConfig->channel_mask = *channels.begin();
+        return NO_ERROR;
+    }
+    return BAD_VALUE;
+}
+
 sp<AudioProfile> AudioProfileVector::getFirstValidProfile() const
 {
     for (size_t i = 0; i < size(); i++) {
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioSourceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioSourceDescriptor.cpp
deleted file mode 100644
index ba33e57..0000000
--- a/services/audiopolicy/common/managerdefinitions/src/AudioSourceDescriptor.cpp
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- * Copyright (C) 2015 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "APM::AudioSourceDescriptor"
-//#define LOG_NDEBUG 0
-
-#include <utils/Log.h>
-#include <utils/String8.h>
-#include <media/AudioPolicyHelper.h>
-#include <HwModule.h>
-#include <AudioGain.h>
-#include <AudioSourceDescriptor.h>
-#include <DeviceDescriptor.h>
-#include <IOProfile.h>
-#include <AudioOutputDescriptor.h>
-
-namespace android {
-
-status_t AudioSourceDescriptor::dump(int fd)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, "mStream: %d\n", audio_attributes_to_stream_type(&mAttributes));
-    result.append(buffer);
-    snprintf(buffer, SIZE, "mDevice:\n");
-    result.append(buffer);
-    write(fd, result.string(), result.size());
-    mDevice->dump(fd, 2 , 0);
-    return NO_ERROR;
-}
-
-
-status_t AudioSourceCollection::dump(int fd) const
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-
-    snprintf(buffer, SIZE, "\nAudio sources dump:\n");
-    write(fd, buffer, strlen(buffer));
-    for (size_t i = 0; i < size(); i++) {
-        snprintf(buffer, SIZE, "- Source %d dump:\n", keyAt(i));
-        write(fd, buffer, strlen(buffer));
-        valueAt(i)->dump(fd);
-    }
-
-    return NO_ERROR;
-}
-
-}; //namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
index bdc748e..5aca3cc 100644
--- a/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
@@ -19,7 +19,13 @@
 
 #include <utils/Log.h>
 #include <utils/String8.h>
+#include "AudioGain.h"
+#include "AudioOutputDescriptor.h"
+#include "AudioPatch.h"
 #include "ClientDescriptor.h"
+#include "DeviceDescriptor.h"
+#include "HwModule.h"
+#include "IOProfile.h"
 
 namespace android {
 
@@ -27,6 +33,9 @@
 {
     String8 out;
 
+    // FIXME: use until other descriptor classes have a dump to String8 method
+    mDumpFd = fd;
+
     status_t status = dump(out, spaces, index);
     if (status == NO_ERROR) {
         write(fd, out.string(), out.size());
@@ -65,4 +74,50 @@
     return NO_ERROR;
 }
 
+SourceClientDescriptor::SourceClientDescriptor(audio_port_handle_t portId, uid_t uid,
+         audio_attributes_t attributes, const sp<AudioPatch>& patchDesc,
+         const sp<DeviceDescriptor>& srcDevice, audio_stream_type_t stream) :
+    TrackClientDescriptor::TrackClientDescriptor(portId, uid, AUDIO_SESSION_NONE, attributes,
+        AUDIO_CONFIG_BASE_INITIALIZER, AUDIO_PORT_HANDLE_NONE, stream, AUDIO_OUTPUT_FLAG_NONE),
+        mPatchDesc(patchDesc), mSrcDevice(srcDevice)
+{
+}
+
+void SourceClientDescriptor::setSwOutput(const sp<SwAudioOutputDescriptor>& swOutput)
+{
+    mSwOutput = swOutput;
+}
+
+void SourceClientDescriptor::setHwOutput(const sp<HwAudioOutputDescriptor>& hwOutput)
+{
+    mHwOutput = hwOutput;
+}
+
+status_t SourceClientDescriptor::dump(String8& out, int spaces, int index)
+{
+    TrackClientDescriptor::dump(out, spaces, index);
+
+    if (mDumpFd >= 0) {
+        out.appendFormat("%*s- Device:\n", spaces, "");
+        write(mDumpFd, out.string(), out.size());
+
+        mSrcDevice->dump(mDumpFd, 2, 0);
+        mDumpFd = -1;
+    }
+
+    return NO_ERROR;
+}
+
+status_t SourceClientCollection::dump(int fd) const
+{
+    String8 out;
+    out.append("\nAudio sources:\n");
+    write(fd, out.string(), out.size());
+    for (size_t i = 0; i < size(); i++) {
+        valueAt(i)->dump(fd, 2, i);
+    }
+
+    return NO_ERROR;
+}
+
 }; //namespace android
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index f343bec..92f6c79 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -31,6 +31,7 @@
 
 #include <inttypes.h>
 #include <math.h>
+#include <vector>
 
 #include <AudioPolicyManagerInterface.h>
 #include <AudioPolicyEngineInstance.h>
@@ -38,7 +39,6 @@
 #include <utils/Log.h>
 #include <media/AudioParameter.h>
 #include <media/AudioPolicyHelper.h>
-#include <media/PatchBuilder.h>
 #include <private/android_filesystem_config.h>
 #include <soundtrigger/SoundTrigger.h>
 #include <system/audio.h>
@@ -82,6 +82,16 @@
     AUDIO_FORMAT_AAC_XHE,
 };
 
+// Compressed formats for MSD module, ordered from most preferred to least preferred.
+static const std::vector<audio_format_t> compressedFormatsOrder = {{
+        AUDIO_FORMAT_MAT_2_1, AUDIO_FORMAT_MAT_2_0, AUDIO_FORMAT_E_AC3,
+        AUDIO_FORMAT_AC3, AUDIO_FORMAT_PCM_16_BIT }};
+// Channel masks for MSD module, 3D > 2D > 1D ordering (most preferred to least preferred).
+static const std::vector<audio_channel_mask_t> surroundChannelMasksOrder = {{
+        AUDIO_CHANNEL_OUT_3POINT1POINT2, AUDIO_CHANNEL_OUT_3POINT0POINT2,
+        AUDIO_CHANNEL_OUT_2POINT1POINT2, AUDIO_CHANNEL_OUT_2POINT0POINT2,
+        AUDIO_CHANNEL_OUT_5POINT1, AUDIO_CHANNEL_OUT_STEREO }};
+
 // ----------------------------------------------------------------------------
 // AudioPolicyInterface implementation
 // ----------------------------------------------------------------------------
@@ -225,6 +235,7 @@
             audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
             updateCallRouting(newDevice);
         }
+        const audio_devices_t msdOutDevice = getMsdAudioOutDeviceTypes();
         for (size_t i = 0; i < mOutputs.size(); i++) {
             sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
             if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) {
@@ -232,7 +243,8 @@
                 // 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 = !desc->isDuplicated()
+                bool force = (msdOutDevice == AUDIO_DEVICE_NONE || msdOutDevice != desc->device())
+                        && !desc->isDuplicated()
                         && (!device_distinguishes_on_address(device)
                                 // always force when disconnecting (a non-duplicated device)
                                 || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
@@ -525,7 +537,7 @@
 }
 
 sp<DeviceDescriptor> AudioPolicyManager::findDevice(
-        const DeviceVector& devices, audio_devices_t device) {
+        const DeviceVector& devices, audio_devices_t device) const {
     DeviceVector deviceList = devices.getDevicesFromTypeMask(device);
     ALOG_ASSERT(!deviceList.isEmpty(),
             "%s() selected device type %#x is not in devices list", __func__, device);
@@ -777,6 +789,7 @@
     routing_strategy strategy;
     audio_devices_t device;
     audio_port_handle_t requestedDeviceId = *selectedDeviceId;
+    audio_devices_t msdDevice = getMsdAudioOutDeviceTypes();
 
     if (attr != NULL) {
         if (!isValidAttributes(attr)) {
@@ -869,7 +882,20 @@
           "flags %#x",
           device, config->sample_rate, config->format, config->channel_mask, *flags);
 
-    *output = getOutputForDevice(device, session, *stream, config, 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;
+        } else {
+            *output = AUDIO_IO_HANDLE_NONE;
+        }
+    }
+    if (*output == AUDIO_IO_HANDLE_NONE) {
+        *output = getOutputForDevice(device, session, *stream, config, flags);
+    }
     if (*output == AUDIO_IO_HANDLE_NONE) {
         mOutputRoutes.removeRoute(session);
         return INVALID_OPERATION;
@@ -1052,6 +1078,164 @@
     return output;
 }
 
+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;
+}
+
+audio_devices_t AudioPolicyManager::getMsdAudioOutDeviceTypes() const {
+    sp<HwModule> msdModule = mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD);
+    if (msdModule != 0) {
+        return mAvailableOutputDevices.getDeviceTypesFromHwModule(msdModule->getHandle());
+    }
+    return AUDIO_DEVICE_NONE;
+}
+
+const AudioPatchCollection AudioPolicyManager::getMsdPatches() const {
+    AudioPatchCollection msdPatches;
+    audio_module_handle_t msdModuleHandle = mHwModules.getModuleFromName(
+            AUDIO_HARDWARE_MODULE_ID_MSD)->getHandle();
+    if (msdModuleHandle == AUDIO_MODULE_HANDLE_NONE) return msdPatches;
+    for (size_t i = 0; i < mAudioPatches.size(); ++i) {
+        sp<AudioPatch> patch = mAudioPatches.valueAt(i);
+        for (size_t j = 0; j < patch->mPatch.num_sources; ++j) {
+            const struct audio_port_config *source = &patch->mPatch.sources[j];
+            if (source->type == AUDIO_PORT_TYPE_DEVICE &&
+                    source->ext.device.hw_module == msdModuleHandle) {
+                msdPatches.addAudioPatch(patch->mHandle, patch);
+            }
+        }
+    }
+    return msdPatches;
+}
+
+status_t AudioPolicyManager::getBestMsdAudioProfileFor(audio_devices_t outputDevice,
+        bool hwAvSync, audio_port_config *sourceConfig, audio_port_config *sinkConfig) const
+{
+    sp<HwModule> msdModule = mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD);
+    if (msdModule == nullptr) {
+        ALOGE("%s() unable to get MSD module", __func__);
+        return NO_INIT;
+    }
+    sp<HwModule> deviceModule = mHwModules.getModuleForDevice(outputDevice);
+    if (deviceModule == nullptr) {
+        ALOGE("%s() unable to get module for %#x", __func__, outputDevice);
+        return NO_INIT;
+    }
+    const InputProfileCollection &inputProfiles = msdModule->getInputProfiles();
+    if (inputProfiles.isEmpty()) {
+        ALOGE("%s() no input profiles for MSD module", __func__);
+        return NO_INIT;
+    }
+    const OutputProfileCollection &outputProfiles = deviceModule->getOutputProfiles();
+    if (outputProfiles.isEmpty()) {
+        ALOGE("%s() no output profiles for device %#x", __func__, outputDevice);
+        return NO_INIT;
+    }
+    AudioProfileVector msdProfiles;
+    // Each IOProfile represents a MixPort from audio_policy_configuration.xml
+    for (const auto &inProfile : inputProfiles) {
+        if (hwAvSync == ((inProfile->getFlags() & AUDIO_INPUT_FLAG_HW_AV_SYNC) != 0)) {
+            msdProfiles.appendVector(inProfile->getAudioProfiles());
+        }
+    }
+    AudioProfileVector deviceProfiles;
+    for (const auto &outProfile : outputProfiles) {
+        if (hwAvSync == ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0)) {
+            deviceProfiles.appendVector(outProfile->getAudioProfiles());
+        }
+    }
+    struct audio_config_base bestSinkConfig;
+    status_t result = msdProfiles.findBestMatchingOutputConfig(deviceProfiles,
+            compressedFormatsOrder, surroundChannelMasksOrder, true /*preferHigherSamplingRates*/,
+            &bestSinkConfig);
+    if (result != NO_ERROR) {
+        ALOGD("%s() no matching profiles found for device: %#x, hwAvSync: %d",
+                __func__, outputDevice, hwAvSync);
+        return result;
+    }
+    sinkConfig->sample_rate = bestSinkConfig.sample_rate;
+    sinkConfig->channel_mask = bestSinkConfig.channel_mask;
+    sinkConfig->format = bestSinkConfig.format;
+    // For encoded streams force direct flag to prevent downstream mixing.
+    sinkConfig->flags.output = static_cast<audio_output_flags_t>(
+            sinkConfig->flags.output | AUDIO_OUTPUT_FLAG_DIRECT);
+    sourceConfig->sample_rate = bestSinkConfig.sample_rate;
+    // Specify exact channel mask to prevent guessing by bit count in PatchPanel.
+    sourceConfig->channel_mask = audio_channel_mask_out_to_in(bestSinkConfig.channel_mask);
+    sourceConfig->format = bestSinkConfig.format;
+    // Copy input stream directly without any processing (e.g. resampling).
+    sourceConfig->flags.input = static_cast<audio_input_flags_t>(
+            sourceConfig->flags.input | AUDIO_INPUT_FLAG_DIRECT);
+    if (hwAvSync) {
+        sinkConfig->flags.output = static_cast<audio_output_flags_t>(
+                sinkConfig->flags.output | AUDIO_OUTPUT_FLAG_HW_AV_SYNC);
+        sourceConfig->flags.input = static_cast<audio_input_flags_t>(
+                sourceConfig->flags.input | AUDIO_INPUT_FLAG_HW_AV_SYNC);
+    }
+    const unsigned int config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE |
+            AUDIO_PORT_CONFIG_CHANNEL_MASK | AUDIO_PORT_CONFIG_FORMAT | AUDIO_PORT_CONFIG_FLAGS;
+    sinkConfig->config_mask |= config_mask;
+    sourceConfig->config_mask |= config_mask;
+    return NO_ERROR;
+}
+
+PatchBuilder AudioPolicyManager::buildMsdPatch(audio_devices_t outputDevice) const
+{
+    PatchBuilder patchBuilder;
+    patchBuilder.addSource(getMsdAudioInDevice()).
+            addSink(findDevice(mAvailableOutputDevices, 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.
+    // For now, we just forcefully try with HwAvSync first.
+    status_t res = getBestMsdAudioProfileFor(outputDevice, true /*hwAvSync*/,
+            &sourceConfig, &sinkConfig) == NO_ERROR ? NO_ERROR :
+            getBestMsdAudioProfileFor(
+                    outputDevice, false /*hwAvSync*/, &sourceConfig, &sinkConfig);
+    if (res == NO_ERROR) {
+        // Found a matching profile for encoded audio. Re-create PatchBuilder with this config.
+        return (PatchBuilder()).addSource(sourceConfig).addSink(sinkConfig);
+    }
+    ALOGV("%s() no matching profile found. Fall through to default PCM patch"
+            " supporting PCM format conversion.", __func__);
+    return patchBuilder;
+}
+
+status_t AudioPolicyManager::setMsdPatch(audio_devices_t outputDevice) {
+    ALOGV("%s() for outputDevice %#x", __func__, outputDevice);
+    if (outputDevice == AUDIO_DEVICE_NONE) {
+        // 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*/);
+    }
+    PatchBuilder patchBuilder = buildMsdPatch(outputDevice);
+    const struct audio_patch* patch = patchBuilder.patch();
+    const AudioPatchCollection msdPatches = getMsdPatches();
+    if (!msdPatches.isEmpty()) {
+        LOG_ALWAYS_FATAL_IF(msdPatches.size() > 1,
+                "The current MSD prototype only supports one output patch");
+        sp<AudioPatch> currentPatch = msdPatches.valueAt(0);
+        if (audio_patches_are_equal(&currentPatch->mPatch, patch)) {
+            return NO_ERROR;
+        }
+        releaseAudioPatch(currentPatch->mHandle, mUidCached);
+    }
+    status_t status = installPatch(__func__, -1 /*index*/, nullptr /*patchHandle*/,
+            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);
+    return status;
+}
+
 audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector<audio_io_handle_t>& outputs,
                                                        audio_output_flags_t flags,
                                                        audio_format_t format)
@@ -2610,6 +2794,7 @@
     mEffects.dump(fd);
     mAudioPatches.dump(fd);
     mPolicyMixes.dump(fd);
+    mAudioSources.dump(fd);
 
     return NO_ERROR;
 }
@@ -3252,8 +3437,8 @@
 void AudioPolicyManager::clearAudioSources(uid_t uid)
 {
     for (ssize_t i = (ssize_t)mAudioSources.size() - 1; i >= 0; i--)  {
-        sp<AudioSourceDescriptor> sourceDesc = mAudioSources.valueAt(i);
-        if (sourceDesc->mUid == uid) {
+        sp<SourceClientDescriptor> sourceDesc = mAudioSources.valueAt(i);
+        if (sourceDesc->uid() == uid) {
             stopAudioSource(mAudioSources.keyAt(i));
         }
     }
@@ -3271,20 +3456,23 @@
 }
 
 status_t AudioPolicyManager::startAudioSource(const struct audio_port_config *source,
-                                  const audio_attributes_t *attributes,
-                                  audio_patch_handle_t *handle,
-                                  uid_t uid)
+                                              const audio_attributes_t *attributes,
+                                              audio_port_handle_t *portId,
+                                              uid_t uid)
 {
-    ALOGV("%s source %p attributes %p handle %p", __FUNCTION__, source, attributes, handle);
-    if (source == NULL || attributes == NULL || handle == NULL) {
+    ALOGV("%s", __FUNCTION__);
+    *portId = AUDIO_PORT_HANDLE_NONE;
+
+    if (source == NULL || attributes == NULL || portId == NULL) {
+        ALOGW("%s invalid argument: source %p attributes %p handle %p",
+              __FUNCTION__, source, attributes, portId);
         return BAD_VALUE;
     }
 
-    *handle = AUDIO_PATCH_HANDLE_NONE;
-
     if (source->role != AUDIO_PORT_ROLE_SOURCE ||
             source->type != AUDIO_PORT_TYPE_DEVICE) {
-        ALOGV("%s INVALID_OPERATION source->role %d source->type %d", __FUNCTION__, source->role, source->type);
+        ALOGW("%s INVALID_OPERATION source->role %d source->type %d",
+              __FUNCTION__, source->role, source->type);
         return INVALID_OPERATION;
     }
 
@@ -3292,34 +3480,37 @@
             mAvailableInputDevices.getDevice(source->ext.device.type,
                                               String8(source->ext.device.address));
     if (srcDeviceDesc == 0) {
-        ALOGV("%s source->ext.device.type %08x not found", __FUNCTION__, source->ext.device.type);
+        ALOGW("%s source->ext.device.type %08x not found", __FUNCTION__, source->ext.device.type);
         return BAD_VALUE;
     }
-    sp<AudioSourceDescriptor> sourceDesc =
-            new AudioSourceDescriptor(srcDeviceDesc, attributes, uid);
+
+    *portId = AudioPort::getNextUniqueId();
 
     struct audio_patch dummyPatch = {};
     sp<AudioPatch> patchDesc = new AudioPatch(&dummyPatch, uid);
-    sourceDesc->mPatchDesc = patchDesc;
+
+    sp<SourceClientDescriptor> sourceDesc =
+        new SourceClientDescriptor(*portId, uid, *attributes, patchDesc, srcDeviceDesc,
+                                   streamTypefromAttributesInt(attributes));
 
     status_t status = connectAudioSource(sourceDesc);
     if (status == NO_ERROR) {
-        mAudioSources.add(sourceDesc->getHandle(), sourceDesc);
-        *handle = sourceDesc->getHandle();
+        mAudioSources.add(*portId, sourceDesc);
     }
     return status;
 }
 
-status_t AudioPolicyManager::connectAudioSource(const sp<AudioSourceDescriptor>& sourceDesc)
+status_t AudioPolicyManager::connectAudioSource(const sp<SourceClientDescriptor>& sourceDesc)
 {
-    ALOGV("%s handle %d", __FUNCTION__, sourceDesc->getHandle());
+    ALOGV("%s handle %d", __FUNCTION__, sourceDesc->portId());
 
     // make sure we only have one patch per source.
     disconnectAudioSource(sourceDesc);
 
-    routing_strategy strategy = (routing_strategy) getStrategyForAttr(&sourceDesc->mAttributes);
-    audio_stream_type_t stream = streamTypefromAttributesInt(&sourceDesc->mAttributes);
-    sp<DeviceDescriptor> srcDeviceDesc = sourceDesc->mDevice;
+    audio_attributes_t attributes = sourceDesc->attributes();
+    routing_strategy strategy = (routing_strategy) getStrategyForAttr(&attributes);
+    audio_stream_type_t stream = sourceDesc->stream();
+    sp<DeviceDescriptor> srcDeviceDesc = sourceDesc->srcDevice();
 
     audio_devices_t sinkDevice = getDeviceForStrategy(strategy, true);
     sp<DeviceDescriptor> sinkDeviceDesc =
@@ -3364,7 +3555,7 @@
                                                               0);
         ALOGV("%s patch panel returned %d patchHandle %d", __FUNCTION__,
                                                               status, afPatchHandle);
-        sourceDesc->mPatchDesc->mPatch = *patchBuilder.patch();
+        sourceDesc->patchDesc()->mPatch = *patchBuilder.patch();
         if (status != NO_ERROR) {
             ALOGW("%s patch panel could not connect device patch, error %d",
                   __FUNCTION__, status);
@@ -3374,32 +3565,32 @@
         status = startSource(outputDesc, stream, sinkDevice, NULL, &delayMs);
 
         if (status != NO_ERROR) {
-            mpClientInterface->releaseAudioPatch(sourceDesc->mPatchDesc->mAfPatchHandle, 0);
+            mpClientInterface->releaseAudioPatch(sourceDesc->patchDesc()->mAfPatchHandle, 0);
             return status;
         }
-        sourceDesc->mSwOutput = outputDesc;
+        sourceDesc->setSwOutput(outputDesc);
         if (delayMs != 0) {
             usleep(delayMs * 1000);
         }
     }
 
-    sourceDesc->mPatchDesc->mAfPatchHandle = afPatchHandle;
-    addAudioPatch(sourceDesc->mPatchDesc->mHandle, sourceDesc->mPatchDesc);
+    sourceDesc->patchDesc()->mAfPatchHandle = afPatchHandle;
+    addAudioPatch(sourceDesc->patchDesc()->mHandle, sourceDesc->patchDesc());
 
     return NO_ERROR;
 }
 
-status_t AudioPolicyManager::stopAudioSource(audio_patch_handle_t handle)
+status_t AudioPolicyManager::stopAudioSource(audio_port_handle_t portId)
 {
-    sp<AudioSourceDescriptor> sourceDesc = mAudioSources.valueFor(handle);
-    ALOGV("%s handle %d", __FUNCTION__, handle);
+    sp<SourceClientDescriptor> sourceDesc = mAudioSources.valueFor(portId);
+    ALOGV("%s port ID %d", __FUNCTION__, portId);
     if (sourceDesc == 0) {
-        ALOGW("%s unknown source for handle %d", __FUNCTION__, handle);
+        ALOGW("%s unknown source for port ID %d", __FUNCTION__, portId);
         return BAD_VALUE;
     }
     status_t status = disconnectAudioSource(sourceDesc);
 
-    mAudioSources.removeItem(handle);
+    mAudioSources.removeItem(portId);
     return status;
 }
 
@@ -3733,20 +3924,20 @@
     }
 }
 
-status_t AudioPolicyManager::disconnectAudioSource(const sp<AudioSourceDescriptor>& sourceDesc)
+status_t AudioPolicyManager::disconnectAudioSource(const sp<SourceClientDescriptor>& sourceDesc)
 {
-    ALOGV("%s handle %d", __FUNCTION__, sourceDesc->getHandle());
+    ALOGV("%s port Id %d", __FUNCTION__, sourceDesc->portId());
 
-    sp<AudioPatch> patchDesc = mAudioPatches.valueFor(sourceDesc->mPatchDesc->mHandle);
+    sp<AudioPatch> patchDesc = mAudioPatches.valueFor(sourceDesc->patchDesc()->mHandle);
     if (patchDesc == 0) {
         ALOGW("%s source has no patch with handle %d", __FUNCTION__,
-              sourceDesc->mPatchDesc->mHandle);
+              sourceDesc->patchDesc()->mHandle);
         return BAD_VALUE;
     }
-    removeAudioPatch(sourceDesc->mPatchDesc->mHandle);
+    removeAudioPatch(sourceDesc->patchDesc()->mHandle);
 
-    audio_stream_type_t stream = streamTypefromAttributesInt(&sourceDesc->mAttributes);
-    sp<SwAudioOutputDescriptor> swOutputDesc = sourceDesc->mSwOutput.promote();
+    audio_stream_type_t stream = sourceDesc->stream();
+    sp<SwAudioOutputDescriptor> swOutputDesc = sourceDesc->swOutput().promote();
     if (swOutputDesc != 0) {
         status_t status = stopSource(swOutputDesc, stream, false);
         if (status == NO_ERROR) {
@@ -3754,7 +3945,7 @@
         }
         mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
     } else {
-        sp<HwAudioOutputDescriptor> hwOutputDesc = sourceDesc->mHwOutput.promote();
+        sp<HwAudioOutputDescriptor> hwOutputDesc = sourceDesc->hwOutput().promote();
         if (hwOutputDesc != 0) {
           //   release patch between src device and output device
           //   close Hwoutput and remove from mHwOutputs
@@ -3765,15 +3956,16 @@
     return NO_ERROR;
 }
 
-sp<AudioSourceDescriptor> AudioPolicyManager::getSourceForStrategyOnOutput(
+sp<SourceClientDescriptor> AudioPolicyManager::getSourceForStrategyOnOutput(
         audio_io_handle_t output, routing_strategy strategy)
 {
-    sp<AudioSourceDescriptor> source;
+    sp<SourceClientDescriptor> source;
     for (size_t i = 0; i < mAudioSources.size(); i++)  {
-        sp<AudioSourceDescriptor> sourceDesc = mAudioSources.valueAt(i);
+        sp<SourceClientDescriptor> sourceDesc = mAudioSources.valueAt(i);
+        audio_attributes_t attributes = sourceDesc->attributes();
         routing_strategy sourceStrategy =
-                (routing_strategy) getStrategyForAttr(&sourceDesc->mAttributes);
-        sp<SwAudioOutputDescriptor> outputDesc = sourceDesc->mSwOutput.promote();
+                (routing_strategy) getStrategyForAttr(&attributes);
+        sp<SwAudioOutputDescriptor> outputDesc = sourceDesc->swOutput().promote();
         if (sourceStrategy == strategy && outputDesc != 0 && outputDesc->mIoHandle == output) {
             source = sourceDesc;
             break;
@@ -4600,19 +4792,17 @@
     return outputs;
 }
 
-void AudioPolicyManager::checkForDeviceAndOutputChanges()
-{
-    checkForDeviceAndOutputChanges([](){ return false; });
-}
-
 void AudioPolicyManager::checkForDeviceAndOutputChanges(std::function<bool()> onOutputsChecked)
 {
     // checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP
     // output is suspended before any tracks are moved to it
     checkA2dpSuspend();
     checkOutputForAllStrategies();
-    if (onOutputsChecked()) checkA2dpSuspend();
+    if (onOutputsChecked != nullptr && onOutputsChecked()) checkA2dpSuspend();
     updateDevicesAndOutputs();
+    if (mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD) != 0) {
+        setMsdPatch();
+    }
 }
 
 void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy)
@@ -4659,7 +4849,7 @@
                 setStrategyMute(strategy, true, desc);
                 setStrategyMute(strategy, false, desc, maxLatency * LATENCY_MUTE_FACTOR, newDevice);
             }
-            sp<AudioSourceDescriptor> source =
+            sp<SourceClientDescriptor> source =
                     getSourceForStrategyOnOutput(srcOut, strategy);
             if (source != 0){
                 connectAudioSource(source);
@@ -5740,10 +5930,10 @@
 void AudioPolicyManager::cleanUpForDevice(const sp<DeviceDescriptor>& deviceDesc)
 {
     for (ssize_t i = (ssize_t)mAudioSources.size() - 1; i >= 0; i--)  {
-        sp<AudioSourceDescriptor> sourceDesc = mAudioSources.valueAt(i);
-        if (sourceDesc->mDevice->equals(deviceDesc)) {
-            ALOGV("%s releasing audio source %d", __FUNCTION__, sourceDesc->getHandle());
-            stopAudioSource(sourceDesc->getHandle());
+        sp<SourceClientDescriptor> sourceDesc = mAudioSources.valueAt(i);
+        if (sourceDesc->srcDevice()->equals(deviceDesc)) {
+            ALOGV("%s releasing audio source %d", __FUNCTION__, sourceDesc->portId());
+            stopAudioSource(sourceDesc->portId());
         }
     }
 
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index e025803..9436767 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -31,6 +31,7 @@
 #include <utils/SortedVector.h>
 #include <media/AudioParameter.h>
 #include <media/AudioPolicy.h>
+#include <media/PatchBuilder.h>
 #include "AudioPolicyInterface.h"
 
 #include <AudioPolicyManagerInterface.h>
@@ -222,9 +223,9 @@
 
         virtual status_t startAudioSource(const struct audio_port_config *source,
                                           const audio_attributes_t *attributes,
-                                          audio_patch_handle_t *handle,
+                                          audio_port_handle_t *portId,
                                           uid_t uid);
-        virtual status_t stopAudioSource(audio_patch_handle_t handle);
+        virtual status_t stopAudioSource(audio_port_handle_t portId);
 
         virtual status_t setMasterMono(bool mono);
         virtual status_t getMasterMono(bool *mono);
@@ -407,8 +408,7 @@
         // if 'onOutputsChecked' callback is provided, it is executed after the outputs
         // check via 'checkOutputForAllStrategies'. If the callback returns 'true',
         // A2DP suspend status is rechecked.
-        void checkForDeviceAndOutputChanges();
-        void checkForDeviceAndOutputChanges(std::function<bool()> onOutputsChecked);
+        void checkForDeviceAndOutputChanges(std::function<bool()> onOutputsChecked = nullptr);
 
         // checks and if necessary changes outputs used for all strategies.
         // must be called every time a condition that affects the output choice for a given strategy
@@ -506,7 +506,7 @@
         uint32_t updateCallRouting(audio_devices_t rxDevice, uint32_t delayMs = 0);
         sp<AudioPatch> createTelephonyPatch(bool isRx, audio_devices_t device, uint32_t delayMs);
         sp<DeviceDescriptor> findDevice(
-                const DeviceVector& devices, audio_devices_t device);
+                const DeviceVector& devices, audio_devices_t device) const;
 
         // if argument "device" is different from AUDIO_DEVICE_NONE,  startSource() will force
         // the re-evaluation of the output device.
@@ -525,10 +525,10 @@
 
         status_t hasPrimaryOutput() const { return mPrimaryOutput != 0; }
 
-        status_t connectAudioSource(const sp<AudioSourceDescriptor>& sourceDesc);
-        status_t disconnectAudioSource(const sp<AudioSourceDescriptor>& sourceDesc);
+        status_t connectAudioSource(const sp<SourceClientDescriptor>& sourceDesc);
+        status_t disconnectAudioSource(const sp<SourceClientDescriptor>& sourceDesc);
 
-        sp<AudioSourceDescriptor> getSourceForStrategyOnOutput(audio_io_handle_t output,
+        sp<SourceClientDescriptor> getSourceForStrategyOnOutput(audio_io_handle_t output,
                                                                routing_strategy strategy);
 
         void cleanUpForDevice(const sp<DeviceDescriptor>& deviceDesc);
@@ -587,7 +587,7 @@
         sp<AudioPatch> mCallRxPatch;
 
         HwAudioOutputCollection mHwOutputs;
-        AudioSourceCollection mAudioSources;
+        SourceClientCollection mAudioSources;
 
         // for supporting "beacon" streams, i.e. streams that only play on speaker, and never
         // when something other than STREAM_TTS (a.k.a. "Transmitted Through Speaker") is playing
@@ -620,6 +620,17 @@
 
         status_t getSupportedFormats(audio_io_handle_t ioHandle, FormatVector& formats);
 
+        // Support for Multi-Stream Decoder (MSD) module
+        sp<DeviceDescriptor> getMsdAudioInDevice() const;
+        audio_devices_t getMsdAudioOutDeviceTypes() const;
+        const AudioPatchCollection getMsdPatches() const;
+        status_t getBestMsdAudioProfileFor(audio_devices_t 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);
+
         // If any, resolve any "dynamic" fields of an Audio Profiles collection
         void updateAudioProfiles(audio_devices_t device, audio_io_handle_t ioHandle,
                 AudioProfileVector &profiles);
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index fdfd573..dbfda44 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -990,26 +990,26 @@
 }
 
 status_t AudioPolicyService::startAudioSource(const struct audio_port_config *source,
-                                  const audio_attributes_t *attributes,
-                                  audio_patch_handle_t *handle)
+                                              const audio_attributes_t *attributes,
+                                              audio_port_handle_t *portId)
 {
     Mutex::Autolock _l(mLock);
     if (mAudioPolicyManager == NULL) {
         return NO_INIT;
     }
     AutoCallerClear acc;
-    return mAudioPolicyManager->startAudioSource(source, attributes, handle,
+    return mAudioPolicyManager->startAudioSource(source, attributes, portId,
                                                  IPCThreadState::self()->getCallingUid());
 }
 
-status_t AudioPolicyService::stopAudioSource(audio_patch_handle_t handle)
+status_t AudioPolicyService::stopAudioSource(audio_port_handle_t portId)
 {
     Mutex::Autolock _l(mLock);
     if (mAudioPolicyManager == NULL) {
         return NO_INIT;
     }
     AutoCallerClear acc;
-    return mAudioPolicyManager->stopAudioSource(handle);
+    return mAudioPolicyManager->stopAudioSource(portId);
 }
 
 status_t AudioPolicyService::setMasterMono(bool mono)
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index d41069e..6a25668 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -184,8 +184,8 @@
 
     virtual status_t startAudioSource(const struct audio_port_config *source,
                                       const audio_attributes_t *attributes,
-                                      audio_patch_handle_t *handle);
-    virtual status_t stopAudioSource(audio_patch_handle_t handle);
+                                      audio_port_handle_t *portId);
+    virtual status_t stopAudioSource(audio_port_handle_t portId);
 
     virtual status_t setMasterMono(bool mono);
     virtual status_t getMasterMono(bool *mono);
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index d4f78e0..491ed72 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -2173,8 +2173,14 @@
 
         res = stream->finishConfiguration();
         if (res != OK) {
-            SET_ERR_L("Can't finish configuring output stream %d: %s (%d)",
-                   stream->getId(), strerror(-res), res);
+            // If finishConfiguration fails due to abandoned surface, do not set
+            // device to error state.
+            bool isSurfaceAbandoned =
+                    (res == NO_INIT || res == DEAD_OBJECT) && stream->isAbandoned();
+            if (!isSurfaceAbandoned) {
+                SET_ERR_L("Can't finish configuring output stream %d: %s (%d)",
+                        stream->getId(), strerror(-res), res);
+            }
             return res;
         }
     }
@@ -2391,9 +2397,16 @@
             //present streams end up with outstanding buffers that will
             //not get drained.
             internalUpdateStatusLocked(STATUS_ACTIVE);
+        } else if (rc == DEAD_OBJECT) {
+            // DEAD_OBJECT can be returned if either the consumer surface is
+            // abandoned, or the HAL has died.
+            // - If the HAL has died, configureStreamsLocked call will set
+            // device to error state,
+            // - If surface is abandoned, we should not set device to error
+            // state.
+            ALOGE("Failed to re-configure camera due to abandoned surface");
         } else {
-            setErrorStateLocked("%s: Failed to re-configure camera: %d",
-                    __FUNCTION__, rc);
+            SET_ERR_L("Failed to re-configure camera: %d", rc);
         }
     } else {
         ALOGE("%s: Failed to pause streaming: %d", __FUNCTION__, rc);
@@ -2527,6 +2540,9 @@
             CLOGE("Can't finish configuring input stream %d: %s (%d)",
                     mInputStream->getId(), strerror(-res), res);
             cancelStreamsConfigurationLocked();
+            if ((res == NO_INIT || res == DEAD_OBJECT) && mInputStream->isAbandoned()) {
+                return DEAD_OBJECT;
+            }
             return BAD_VALUE;
         }
     }
@@ -2540,6 +2556,9 @@
                 CLOGE("Can't finish configuring output stream %d: %s (%d)",
                         outputStream->getId(), strerror(-res), res);
                 cancelStreamsConfigurationLocked();
+                if ((res == NO_INIT || res == DEAD_OBJECT) && outputStream->isAbandoned()) {
+                    return DEAD_OBJECT;
+                }
                 return BAD_VALUE;
             }
         }
diff --git a/services/camera/libcameraservice/device3/Camera3Stream.cpp b/services/camera/libcameraservice/device3/Camera3Stream.cpp
index 1105b75..6030d15 100644
--- a/services/camera/libcameraservice/device3/Camera3Stream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Stream.cpp
@@ -331,7 +331,14 @@
 
     status_t res;
     res = configureQueueLocked();
-    if (res != OK) {
+    // configureQueueLocked could return error in case of abandoned surface.
+    // Treat as non-fatal error.
+    if (res == NO_INIT || res == DEAD_OBJECT) {
+        ALOGE("%s: Unable to configure stream %d queue (non-fatal): %s (%d)",
+                __FUNCTION__, mId, strerror(-res), res);
+        mState = STATE_ABANDONED;
+        return res;
+    } else if (res != OK) {
         ALOGE("%s: Unable to configure stream %d queue: %s (%d)",
                 __FUNCTION__, mId, strerror(-res), res);
         mState = STATE_ERROR;
diff --git a/services/camera/libcameraservice/device3/Camera3Stream.h b/services/camera/libcameraservice/device3/Camera3Stream.h
index a60cb56..4ddcf1a 100644
--- a/services/camera/libcameraservice/device3/Camera3Stream.h
+++ b/services/camera/libcameraservice/device3/Camera3Stream.h
@@ -482,6 +482,7 @@
     // after the HAL has provided usage and max_buffers values. After this call,
     // the stream must be ready to produce all buffers for registration with
     // HAL.
+    // Returns NO_INIT or DEAD_OBJECT if the queue has been abandoned.
     virtual status_t configureQueueLocked() = 0;
 
     // Get the total number of buffers in the queue